* [alsa-devel] Applied "ASoC: SOF: Intel: Fix build break" to the asoc tree
@ 2019-11-18 17:48 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2019-11-18 17:48 UTC (permalink / raw)
To: Mark Brown; +Cc: Stephen Rothwell, alsa-devel
The patch
ASoC: SOF: Intel: Fix build break
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 302ee055af0201e61be7670f957c1622d6ce176e Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@kernel.org>
Date: Mon, 18 Nov 2019 15:52:47 +0000
Subject: [PATCH] ASoC: SOF: Intel: Fix build break
Commit 130d3e9077 (Fix CFL and CML FW nocodec binary names.) broke the
build in some configurations as it depends on changes in the development
branch, revert it.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/sof-pci-dev.c | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c
index 2ef927371b23..d66412a77873 100644
--- a/sound/soc/sof/sof-pci-dev.c
+++ b/sound/soc/sof/sof-pci-dev.c
@@ -113,7 +113,7 @@ static const struct sof_dev_desc cnl_desc = {
#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE)
static const struct sof_dev_desc cfl_desc = {
- .machines = snd_soc_acpi_intel_cfl_machines,
+ .machines = snd_soc_acpi_intel_cnl_machines,
.resindex_lpe_base = 0,
.resindex_pcicfg_base = -1,
.resindex_imr_base = -1,
@@ -122,7 +122,7 @@ static const struct sof_dev_desc cfl_desc = {
.chip_info = &cnl_chip_info,
.default_fw_path = "intel/sof",
.default_tplg_path = "intel/sof-tplg",
- .nocodec_fw_filename = "sof-cfl.ri",
+ .nocodec_fw_filename = "sof-cnl.ri",
.nocodec_tplg_filename = "sof-cnl-nocodec.tplg",
.ops = &sof_cnl_ops,
.arch_ops = &sof_xtensa_arch_ops
@@ -133,7 +133,7 @@ static const struct sof_dev_desc cfl_desc = {
IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H)
static const struct sof_dev_desc cml_desc = {
- .machines = snd_soc_acpi_intel_cml_machines,
+ .machines = snd_soc_acpi_intel_cnl_machines,
.resindex_lpe_base = 0,
.resindex_pcicfg_base = -1,
.resindex_imr_base = -1,
@@ -142,7 +142,7 @@ static const struct sof_dev_desc cml_desc = {
.chip_info = &cnl_chip_info,
.default_fw_path = "intel/sof",
.default_tplg_path = "intel/sof-tplg",
- .nocodec_fw_filename = "sof-cml.ri",
+ .nocodec_fw_filename = "sof-cnl.ri",
.nocodec_tplg_filename = "sof-cnl-nocodec.tplg",
.ops = &sof_cnl_ops,
.arch_ops = &sof_xtensa_arch_ops
--
2.20.1
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^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: fsl_audmix: add missed pm_runtime_disable" to the asoc tree
@ 2019-12-25 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2019-12-25 0:09 UTC (permalink / raw)
To: Chuhong Yuan; +Cc: Nicolin Chen, alsa-devel, Mark Brown
The patch
ASoC: fsl_audmix: add missed pm_runtime_disable
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 77fffa742285f2b587648d6c72b5c705633f146f Mon Sep 17 00:00:00 2001
From: Chuhong Yuan <hslester96@gmail.com>
Date: Tue, 3 Dec 2019 19:13:03 +0800
Subject: [PATCH] ASoC: fsl_audmix: add missed pm_runtime_disable
The driver forgets to call pm_runtime_disable in probe failure
and remove.
Add the missed calls to fix it.
Signed-off-by: Chuhong Yuan <hslester96@gmail.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20191203111303.12933-1-hslester96@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/fsl_audmix.c | 9 ++++++++-
1 file changed, 8 insertions(+), 1 deletion(-)
diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c
index a1db1bce330f..5faecbeb5497 100644
--- a/sound/soc/fsl/fsl_audmix.c
+++ b/sound/soc/fsl/fsl_audmix.c
@@ -505,15 +505,20 @@ static int fsl_audmix_probe(struct platform_device *pdev)
ARRAY_SIZE(fsl_audmix_dai));
if (ret) {
dev_err(dev, "failed to register ASoC DAI\n");
- return ret;
+ goto err_disable_pm;
}
priv->pdev = platform_device_register_data(dev, mdrv, 0, NULL, 0);
if (IS_ERR(priv->pdev)) {
ret = PTR_ERR(priv->pdev);
dev_err(dev, "failed to register platform %s: %d\n", mdrv, ret);
+ goto err_disable_pm;
}
+ return 0;
+
+err_disable_pm:
+ pm_runtime_disable(dev);
return ret;
}
@@ -521,6 +526,8 @@ static int fsl_audmix_remove(struct platform_device *pdev)
{
struct fsl_audmix *priv = dev_get_drvdata(&pdev->dev);
+ pm_runtime_disable(&pdev->dev);
+
if (priv->pdev)
platform_device_unregister(priv->pdev);
--
2.20.1
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^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga" to the asoc tree
@ 2019-12-25 18:21 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2019-12-25 18:21 UTC (permalink / raw)
To: Srinivas Kandagatla; +Cc: alsa-devel, Mark Brown
The patch
ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 5caf64c633a3564f70e734868254281b25932fc0 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Thu, 19 Dec 2019 10:31:53 +0000
Subject: [PATCH] ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga
This patch adds support to Lenovo Yoga c630 compatible strings
and related setup to the sound machine driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20191219103153.14875-12-srinivas.kandagatla@linaro.org
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/sdm845.c | 86 ++++++++++++++++++++++++++++++++++++++++-
1 file changed, 85 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 28f3cef696e6..3b5547a27aad 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -24,6 +24,9 @@
#define RIGHT_SPK_TDM_TX_MASK 0xC0
#define SPK_TDM_RX_MASK 0x03
#define NUM_TDM_SLOTS 8
+#define SLIM_MAX_TX_PORTS 16
+#define SLIM_MAX_RX_PORTS 16
+#define WCD934X_DEFAULT_MCLK_RATE 9600000
struct sdm845_snd_data {
struct snd_soc_jack jack;
@@ -36,6 +39,39 @@ struct sdm845_snd_data {
static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
+static int sdm845_slim_snd_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai_link *dai_link = rtd->dai_link;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
+ u32 rx_ch_cnt = 0, tx_ch_cnt = 0;
+ int ret = 0, i;
+
+ for (i = 0 ; i < dai_link->num_codecs; i++) {
+ ret = snd_soc_dai_get_channel_map(rtd->codec_dais[i],
+ &tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch);
+
+ if (ret != 0 && ret != -ENOTSUPP) {
+ pr_err("failed to get codec chan map, err:%d\n", ret);
+ return ret;
+ } else if (ret == -ENOTSUPP) {
+ /* Ignore unsupported */
+ continue;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL,
+ rx_ch_cnt, rx_ch);
+ else
+ ret = snd_soc_dai_set_channel_map(cpu_dai, tx_ch_cnt,
+ tx_ch, 0, NULL);
+ }
+
+ return 0;
+}
+
static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -151,6 +187,11 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream,
case QUATERNARY_TDM_TX_0:
ret = sdm845_tdm_snd_hw_params(substream, params);
break;
+ case SLIMBUS_0_RX...SLIMBUS_6_TX:
+ ret = sdm845_slim_snd_hw_params(substream, params);
+ break;
+ case QUATERNARY_MI2S_RX:
+ break;
default:
pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
break;
@@ -173,7 +214,20 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card);
struct snd_jack *jack;
- int rval;
+ struct snd_soc_dai_link *dai_link = rtd->dai_link;
+ /*
+ * Codec SLIMBUS configuration
+ * RX1, RX2, RX3, RX4, RX5, RX6, RX7, RX8, RX9, RX10, RX11, RX12, RX13
+ * TX1, TX2, TX3, TX4, TX5, TX6, TX7, TX8, TX9, TX10, TX11, TX12, TX13
+ * TX14, TX15, TX16
+ */
+ unsigned int rx_ch[SLIM_MAX_RX_PORTS] = {144, 145, 146, 147, 148, 149,
+ 150, 151, 152, 153, 154, 155, 156};
+ unsigned int tx_ch[SLIM_MAX_TX_PORTS] = {128, 129, 130, 131, 132, 133,
+ 134, 135, 136, 137, 138, 139,
+ 140, 141, 142, 143};
+ int rval, i;
+
if (!pdata->jack_setup) {
rval = snd_soc_card_jack_new(card, "Headset Jack",
@@ -211,6 +265,21 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
return rval;
}
break;
+ case SLIMBUS_0_RX...SLIMBUS_6_TX:
+ for (i = 0 ; i < dai_link->num_codecs; i++) {
+ rval = snd_soc_dai_set_channel_map(rtd->codec_dais[i],
+ ARRAY_SIZE(tx_ch),
+ tx_ch,
+ ARRAY_SIZE(rx_ch),
+ rx_ch);
+ if (rval != 0 && rval != -ENOTSUPP)
+ return rval;
+
+ snd_soc_dai_set_sysclk(rtd->codec_dais[i], 0,
+ WCD934X_DEFAULT_MCLK_RATE,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ }
+ break;
default:
break;
}
@@ -256,6 +325,14 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream)
}
snd_soc_dai_set_fmt(cpu_dai, fmt);
snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt);
+ break;
+ case QUATERNARY_MI2S_RX:
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT,
+ MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+ snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS);
+
+
break;
case QUATERNARY_TDM_RX_0:
@@ -294,6 +371,8 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream)
}
}
break;
+ case SLIMBUS_0_RX...SLIMBUS_6_TX:
+ break;
default:
pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
@@ -338,6 +417,9 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream)
0, SNDRV_PCM_STREAM_PLAYBACK);
}
break;
+ case SLIMBUS_0_RX...SLIMBUS_6_TX:
+ case QUATERNARY_MI2S_RX:
+ break;
default:
pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
@@ -451,6 +533,8 @@ static int sdm845_snd_platform_remove(struct platform_device *pdev)
static const struct of_device_id sdm845_snd_device_id[] = {
{ .compatible = "qcom,sdm845-sndcard" },
+ { .compatible = "qcom,db845c-sndcard" },
+ { .compatible = "lenovo,yoga-c630-sndcard" },
{},
};
MODULE_DEVICE_TABLE(of, sdm845_snd_device_id);
--
2.20.1
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^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: qcom: dt-bindings: Add compatible for DB845c and Lenovo Yoga" to the asoc tree
@ 2019-12-25 18:21 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2019-12-25 18:21 UTC (permalink / raw)
To: Srinivas Kandagatla; +Cc: Rob Herring, alsa-devel, Mark Brown
The patch
ASoC: qcom: dt-bindings: Add compatible for DB845c and Lenovo Yoga
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 834d899794f4ec3bbf9836c85e8f76b7de6e8a59 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Thu, 19 Dec 2019 10:31:52 +0000
Subject: [PATCH] ASoC: qcom: dt-bindings: Add compatible for DB845c and Lenovo
Yoga
This patch adds compatible strings for DB845c and Lenovo Yoga C630
soundcard. Based on this compatible strings common machine driver
will be in better position to setup board specific configuration.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20191219103153.14875-11-srinivas.kandagatla@linaro.org
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/qcom,sdm845.txt | 5 ++++-
1 file changed, 4 insertions(+), 1 deletion(-)
diff --git a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
index 408c4837e6d5..ca8c89e88bfa 100644
--- a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
+++ b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
@@ -5,7 +5,10 @@ This binding describes the SDM845 sound card, which uses qdsp for audio.
- compatible:
Usage: required
Value type: <stringlist>
- Definition: must be "qcom,sdm845-sndcard"
+ Definition: must be one of this
+ "qcom,sdm845-sndcard"
+ "qcom,db845c-sndcard"
+ "lenovo,yoga-c630-sndcard"
- audio-routing:
Usage: Optional
--
2.20.1
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^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: wcd934x: add audio routings" to the asoc tree
@ 2019-12-25 18:21 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2019-12-25 18:21 UTC (permalink / raw)
To: Srinivas Kandagatla; +Cc: alsa-devel, Mark Brown
The patch
ASoC: wcd934x: add audio routings
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From da3e83f8bb866a91945ef5c47bdb25de189a381e Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Thu, 19 Dec 2019 10:31:49 +0000
Subject: [PATCH] ASoC: wcd934x: add audio routings
This patch adds audio routing for both playback and capture.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20191219103153.14875-8-srinivas.kandagatla@linaro.org
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wcd934x.c | 290 +++++++++++++++++++++++++++++++++++++
1 file changed, 290 insertions(+)
diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c
index 24f78a0b7f5a..158e878abd6c 100644
--- a/sound/soc/codecs/wcd934x.c
+++ b/sound/soc/codecs/wcd934x.c
@@ -135,6 +135,162 @@
} \
}
+#define WCD934X_INTERPOLATOR_PATH(id) \
+ {"RX INT" #id "_1 MIX1 INP0", "RX0", "SLIM RX0"}, \
+ {"RX INT" #id "_1 MIX1 INP0", "RX1", "SLIM RX1"}, \
+ {"RX INT" #id "_1 MIX1 INP0", "RX2", "SLIM RX2"}, \
+ {"RX INT" #id "_1 MIX1 INP0", "RX3", "SLIM RX3"}, \
+ {"RX INT" #id "_1 MIX1 INP0", "RX4", "SLIM RX4"}, \
+ {"RX INT" #id "_1 MIX1 INP0", "RX5", "SLIM RX5"}, \
+ {"RX INT" #id "_1 MIX1 INP0", "RX6", "SLIM RX6"}, \
+ {"RX INT" #id "_1 MIX1 INP0", "RX7", "SLIM RX7"}, \
+ {"RX INT" #id "_1 MIX1 INP0", "IIR0", "IIR0"}, \
+ {"RX INT" #id "_1 MIX1 INP0", "IIR1", "IIR1"}, \
+ {"RX INT" #id "_1 MIX1 INP1", "RX0", "SLIM RX0"}, \
+ {"RX INT" #id "_1 MIX1 INP1", "RX1", "SLIM RX1"}, \
+ {"RX INT" #id "_1 MIX1 INP1", "RX2", "SLIM RX2"}, \
+ {"RX INT" #id "_1 MIX1 INP1", "RX3", "SLIM RX3"}, \
+ {"RX INT" #id "_1 MIX1 INP1", "RX4", "SLIM RX4"}, \
+ {"RX INT" #id "_1 MIX1 INP1", "RX5", "SLIM RX5"}, \
+ {"RX INT" #id "_1 MIX1 INP1", "RX6", "SLIM RX6"}, \
+ {"RX INT" #id "_1 MIX1 INP1", "RX7", "SLIM RX7"}, \
+ {"RX INT" #id "_1 MIX1 INP1", "IIR0", "IIR0"}, \
+ {"RX INT" #id "_1 MIX1 INP1", "IIR1", "IIR1"}, \
+ {"RX INT" #id "_1 MIX1 INP2", "RX0", "SLIM RX0"}, \
+ {"RX INT" #id "_1 MIX1 INP2", "RX1", "SLIM RX1"}, \
+ {"RX INT" #id "_1 MIX1 INP2", "RX2", "SLIM RX2"}, \
+ {"RX INT" #id "_1 MIX1 INP2", "RX3", "SLIM RX3"}, \
+ {"RX INT" #id "_1 MIX1 INP2", "RX4", "SLIM RX4"}, \
+ {"RX INT" #id "_1 MIX1 INP2", "RX5", "SLIM RX5"}, \
+ {"RX INT" #id "_1 MIX1 INP2", "RX6", "SLIM RX6"}, \
+ {"RX INT" #id "_1 MIX1 INP2", "RX7", "SLIM RX7"}, \
+ {"RX INT" #id "_1 MIX1 INP2", "IIR0", "IIR0"}, \
+ {"RX INT" #id "_1 MIX1 INP2", "IIR1", "IIR1"}, \
+ {"RX INT" #id "_1 MIX1", NULL, "RX INT" #id "_1 MIX1 INP0"}, \
+ {"RX INT" #id "_1 MIX1", NULL, "RX INT" #id "_1 MIX1 INP1"}, \
+ {"RX INT" #id "_1 MIX1", NULL, "RX INT" #id "_1 MIX1 INP2"}, \
+ {"RX INT" #id "_2 MUX", "RX0", "SLIM RX0"}, \
+ {"RX INT" #id "_2 MUX", "RX1", "SLIM RX1"}, \
+ {"RX INT" #id "_2 MUX", "RX2", "SLIM RX2"}, \
+ {"RX INT" #id "_2 MUX", "RX3", "SLIM RX3"}, \
+ {"RX INT" #id "_2 MUX", "RX4", "SLIM RX4"}, \
+ {"RX INT" #id "_2 MUX", "RX5", "SLIM RX5"}, \
+ {"RX INT" #id "_2 MUX", "RX6", "SLIM RX6"}, \
+ {"RX INT" #id "_2 MUX", "RX7", "SLIM RX7"}, \
+ {"RX INT" #id "_2 MUX", NULL, "INT" #id "_CLK"}, \
+ {"RX INT" #id "_2 MUX", NULL, "DSMDEM" #id "_CLK"}, \
+ {"RX INT" #id "_2 INTERP", NULL, "RX INT" #id "_2 MUX"}, \
+ {"RX INT" #id " SEC MIX", NULL, "RX INT" #id "_2 INTERP"}, \
+ {"RX INT" #id "_1 INTERP", NULL, "RX INT" #id "_1 MIX1"}, \
+ {"RX INT" #id "_1 INTERP", NULL, "INT" #id "_CLK"}, \
+ {"RX INT" #id "_1 INTERP", NULL, "DSMDEM" #id "_CLK"}, \
+ {"RX INT" #id " SEC MIX", NULL, "RX INT" #id "_1 INTERP"}
+
+#define WCD934X_INTERPOLATOR_MIX2(id) \
+ {"RX INT" #id " MIX2", NULL, "RX INT" #id " SEC MIX"}, \
+ {"RX INT" #id " MIX2", NULL, "RX INT" #id " MIX2 INP"}
+
+#define WCD934X_SLIM_RX_AIF_PATH(id) \
+ {"SLIM RX"#id" MUX", "AIF1_PB", "AIF1 PB"}, \
+ {"SLIM RX"#id" MUX", "AIF2_PB", "AIF2 PB"}, \
+ {"SLIM RX"#id" MUX", "AIF3_PB", "AIF3 PB"}, \
+ {"SLIM RX"#id" MUX", "AIF4_PB", "AIF4 PB"}, \
+ {"SLIM RX"#id, NULL, "SLIM RX"#id" MUX"}
+
+#define WCD934X_ADC_MUX(id) \
+ {"ADC MUX" #id, "DMIC", "DMIC MUX" #id }, \
+ {"ADC MUX" #id, "AMIC", "AMIC MUX" #id }, \
+ {"DMIC MUX" #id, "DMIC0", "DMIC0"}, \
+ {"DMIC MUX" #id, "DMIC1", "DMIC1"}, \
+ {"DMIC MUX" #id, "DMIC2", "DMIC2"}, \
+ {"DMIC MUX" #id, "DMIC3", "DMIC3"}, \
+ {"DMIC MUX" #id, "DMIC4", "DMIC4"}, \
+ {"DMIC MUX" #id, "DMIC5", "DMIC5"}, \
+ {"AMIC MUX" #id, "ADC1", "ADC1"}, \
+ {"AMIC MUX" #id, "ADC2", "ADC2"}, \
+ {"AMIC MUX" #id, "ADC3", "ADC3"}, \
+ {"AMIC MUX" #id, "ADC4", "ADC4"}
+
+#define WCD934X_IIR_INP_MUX(id) \
+ {"IIR" #id, NULL, "IIR" #id " INP0 MUX"}, \
+ {"IIR" #id " INP0 MUX", "DEC0", "ADC MUX0"}, \
+ {"IIR" #id " INP0 MUX", "DEC1", "ADC MUX1"}, \
+ {"IIR" #id " INP0 MUX", "DEC2", "ADC MUX2"}, \
+ {"IIR" #id " INP0 MUX", "DEC3", "ADC MUX3"}, \
+ {"IIR" #id " INP0 MUX", "DEC4", "ADC MUX4"}, \
+ {"IIR" #id " INP0 MUX", "DEC5", "ADC MUX5"}, \
+ {"IIR" #id " INP0 MUX", "DEC6", "ADC MUX6"}, \
+ {"IIR" #id " INP0 MUX", "DEC7", "ADC MUX7"}, \
+ {"IIR" #id " INP0 MUX", "DEC8", "ADC MUX8"}, \
+ {"IIR" #id " INP0 MUX", "RX0", "SLIM RX0"}, \
+ {"IIR" #id " INP0 MUX", "RX1", "SLIM RX1"}, \
+ {"IIR" #id " INP0 MUX", "RX2", "SLIM RX2"}, \
+ {"IIR" #id " INP0 MUX", "RX3", "SLIM RX3"}, \
+ {"IIR" #id " INP0 MUX", "RX4", "SLIM RX4"}, \
+ {"IIR" #id " INP0 MUX", "RX5", "SLIM RX5"}, \
+ {"IIR" #id " INP0 MUX", "RX6", "SLIM RX6"}, \
+ {"IIR" #id " INP0 MUX", "RX7", "SLIM RX7"}, \
+ {"IIR" #id, NULL, "IIR" #id " INP1 MUX"}, \
+ {"IIR" #id " INP1 MUX", "DEC0", "ADC MUX0"}, \
+ {"IIR" #id " INP1 MUX", "DEC1", "ADC MUX1"}, \
+ {"IIR" #id " INP1 MUX", "DEC2", "ADC MUX2"}, \
+ {"IIR" #id " INP1 MUX", "DEC3", "ADC MUX3"}, \
+ {"IIR" #id " INP1 MUX", "DEC4", "ADC MUX4"}, \
+ {"IIR" #id " INP1 MUX", "DEC5", "ADC MUX5"}, \
+ {"IIR" #id " INP1 MUX", "DEC6", "ADC MUX6"}, \
+ {"IIR" #id " INP1 MUX", "DEC7", "ADC MUX7"}, \
+ {"IIR" #id " INP1 MUX", "DEC8", "ADC MUX8"}, \
+ {"IIR" #id " INP1 MUX", "RX0", "SLIM RX0"}, \
+ {"IIR" #id " INP1 MUX", "RX1", "SLIM RX1"}, \
+ {"IIR" #id " INP1 MUX", "RX2", "SLIM RX2"}, \
+ {"IIR" #id " INP1 MUX", "RX3", "SLIM RX3"}, \
+ {"IIR" #id " INP1 MUX", "RX4", "SLIM RX4"}, \
+ {"IIR" #id " INP1 MUX", "RX5", "SLIM RX5"}, \
+ {"IIR" #id " INP1 MUX", "RX6", "SLIM RX6"}, \
+ {"IIR" #id " INP1 MUX", "RX7", "SLIM RX7"}, \
+ {"IIR" #id, NULL, "IIR" #id " INP2 MUX"}, \
+ {"IIR" #id " INP2 MUX", "DEC0", "ADC MUX0"}, \
+ {"IIR" #id " INP2 MUX", "DEC1", "ADC MUX1"}, \
+ {"IIR" #id " INP2 MUX", "DEC2", "ADC MUX2"}, \
+ {"IIR" #id " INP2 MUX", "DEC3", "ADC MUX3"}, \
+ {"IIR" #id " INP2 MUX", "DEC4", "ADC MUX4"}, \
+ {"IIR" #id " INP2 MUX", "DEC5", "ADC MUX5"}, \
+ {"IIR" #id " INP2 MUX", "DEC6", "ADC MUX6"}, \
+ {"IIR" #id " INP2 MUX", "DEC7", "ADC MUX7"}, \
+ {"IIR" #id " INP2 MUX", "DEC8", "ADC MUX8"}, \
+ {"IIR" #id " INP2 MUX", "RX0", "SLIM RX0"}, \
+ {"IIR" #id " INP2 MUX", "RX1", "SLIM RX1"}, \
+ {"IIR" #id " INP2 MUX", "RX2", "SLIM RX2"}, \
+ {"IIR" #id " INP2 MUX", "RX3", "SLIM RX3"}, \
+ {"IIR" #id " INP2 MUX", "RX4", "SLIM RX4"}, \
+ {"IIR" #id " INP2 MUX", "RX5", "SLIM RX5"}, \
+ {"IIR" #id " INP2 MUX", "RX6", "SLIM RX6"}, \
+ {"IIR" #id " INP2 MUX", "RX7", "SLIM RX7"}, \
+ {"IIR" #id, NULL, "IIR" #id " INP3 MUX"}, \
+ {"IIR" #id " INP3 MUX", "DEC0", "ADC MUX0"}, \
+ {"IIR" #id " INP3 MUX", "DEC1", "ADC MUX1"}, \
+ {"IIR" #id " INP3 MUX", "DEC2", "ADC MUX2"}, \
+ {"IIR" #id " INP3 MUX", "DEC3", "ADC MUX3"}, \
+ {"IIR" #id " INP3 MUX", "DEC4", "ADC MUX4"}, \
+ {"IIR" #id " INP3 MUX", "DEC5", "ADC MUX5"}, \
+ {"IIR" #id " INP3 MUX", "DEC6", "ADC MUX6"}, \
+ {"IIR" #id " INP3 MUX", "DEC7", "ADC MUX7"}, \
+ {"IIR" #id " INP3 MUX", "DEC8", "ADC MUX8"}, \
+ {"IIR" #id " INP3 MUX", "RX0", "SLIM RX0"}, \
+ {"IIR" #id " INP3 MUX", "RX1", "SLIM RX1"}, \
+ {"IIR" #id " INP3 MUX", "RX2", "SLIM RX2"}, \
+ {"IIR" #id " INP3 MUX", "RX3", "SLIM RX3"}, \
+ {"IIR" #id " INP3 MUX", "RX4", "SLIM RX4"}, \
+ {"IIR" #id " INP3 MUX", "RX5", "SLIM RX5"}, \
+ {"IIR" #id " INP3 MUX", "RX6", "SLIM RX6"}, \
+ {"IIR" #id " INP3 MUX", "RX7", "SLIM RX7"}
+
+#define WCD934X_SLIM_TX_AIF_PATH(id) \
+ {"AIF1_CAP Mixer", "SLIM TX" #id, "SLIM TX" #id }, \
+ {"AIF2_CAP Mixer", "SLIM TX" #id, "SLIM TX" #id }, \
+ {"AIF3_CAP Mixer", "SLIM TX" #id, "SLIM TX" #id }, \
+ {"SLIM TX" #id, NULL, "CDC_IF TX" #id " MUX"}
+
enum {
MIC_BIAS_1 = 1,
MIC_BIAS_2,
@@ -4678,6 +4834,138 @@ static const struct snd_soc_dapm_widget wcd934x_dapm_widgets[] = {
ARRAY_SIZE(aif3_slim_cap_mixer)),
};
+static const struct snd_soc_dapm_route wcd934x_audio_map[] = {
+ /* RX0-RX7 */
+ WCD934X_SLIM_RX_AIF_PATH(0),
+ WCD934X_SLIM_RX_AIF_PATH(1),
+ WCD934X_SLIM_RX_AIF_PATH(2),
+ WCD934X_SLIM_RX_AIF_PATH(3),
+ WCD934X_SLIM_RX_AIF_PATH(4),
+ WCD934X_SLIM_RX_AIF_PATH(5),
+ WCD934X_SLIM_RX_AIF_PATH(6),
+ WCD934X_SLIM_RX_AIF_PATH(7),
+
+ /* RX0 Ear out */
+ WCD934X_INTERPOLATOR_PATH(0),
+ WCD934X_INTERPOLATOR_MIX2(0),
+ {"RX INT0 DEM MUX", "CLSH_DSM_OUT", "RX INT0 MIX2"},
+ {"RX INT0 DAC", NULL, "RX INT0 DEM MUX"},
+ {"RX INT0 DAC", NULL, "RX_BIAS"},
+ {"EAR PA", NULL, "RX INT0 DAC"},
+ {"EAR", NULL, "EAR PA"},
+
+ /* RX1 Headphone left */
+ WCD934X_INTERPOLATOR_PATH(1),
+ WCD934X_INTERPOLATOR_MIX2(1),
+ {"RX INT1 MIX3", NULL, "RX INT1 MIX2"},
+ {"RX INT1 DEM MUX", "CLSH_DSM_OUT", "RX INT1 MIX3"},
+ {"RX INT1 DAC", NULL, "RX INT1 DEM MUX"},
+ {"RX INT1 DAC", NULL, "RX_BIAS"},
+ {"HPHL PA", NULL, "RX INT1 DAC"},
+ {"HPHL", NULL, "HPHL PA"},
+
+ /* RX2 Headphone right */
+ WCD934X_INTERPOLATOR_PATH(2),
+ WCD934X_INTERPOLATOR_MIX2(2),
+ {"RX INT2 MIX3", NULL, "RX INT2 MIX2"},
+ {"RX INT2 DEM MUX", "CLSH_DSM_OUT", "RX INT2 MIX3"},
+ {"RX INT2 DAC", NULL, "RX INT2 DEM MUX"},
+ {"RX INT2 DAC", NULL, "RX_BIAS"},
+ {"HPHR PA", NULL, "RX INT2 DAC"},
+ {"HPHR", NULL, "HPHR PA"},
+
+ /* RX3 HIFi LineOut1 */
+ WCD934X_INTERPOLATOR_PATH(3),
+ WCD934X_INTERPOLATOR_MIX2(3),
+ {"RX INT3 MIX3", NULL, "RX INT3 MIX2"},
+ {"RX INT3 DAC", NULL, "RX INT3 MIX3"},
+ {"RX INT3 DAC", NULL, "RX_BIAS"},
+ {"LINEOUT1 PA", NULL, "RX INT3 DAC"},
+ {"LINEOUT1", NULL, "LINEOUT1 PA"},
+
+ /* RX4 HIFi LineOut2 */
+ WCD934X_INTERPOLATOR_PATH(4),
+ WCD934X_INTERPOLATOR_MIX2(4),
+ {"RX INT4 MIX3", NULL, "RX INT4 MIX2"},
+ {"RX INT4 DAC", NULL, "RX INT4 MIX3"},
+ {"RX INT4 DAC", NULL, "RX_BIAS"},
+ {"LINEOUT2 PA", NULL, "RX INT4 DAC"},
+ {"LINEOUT2", NULL, "LINEOUT2 PA"},
+
+ /* RX7 Speaker Left Out PA */
+ WCD934X_INTERPOLATOR_PATH(7),
+ WCD934X_INTERPOLATOR_MIX2(7),
+ {"RX INT7 CHAIN", NULL, "RX INT7 MIX2"},
+ {"RX INT7 CHAIN", NULL, "RX_BIAS"},
+ {"RX INT7 CHAIN", NULL, "SBOOST0"},
+ {"RX INT7 CHAIN", NULL, "SBOOST0_CLK"},
+ {"SPK1 OUT", NULL, "RX INT7 CHAIN"},
+
+ /* RX8 Speaker Right Out PA */
+ WCD934X_INTERPOLATOR_PATH(8),
+ {"RX INT8 CHAIN", NULL, "RX INT8 SEC MIX"},
+ {"RX INT8 CHAIN", NULL, "RX_BIAS"},
+ {"RX INT8 CHAIN", NULL, "SBOOST1"},
+ {"RX INT8 CHAIN", NULL, "SBOOST1_CLK"},
+ {"SPK2 OUT", NULL, "RX INT8 CHAIN"},
+
+ /* Tx */
+ {"AIF1 CAP", NULL, "AIF1_CAP Mixer"},
+ {"AIF2 CAP", NULL, "AIF2_CAP Mixer"},
+ {"AIF3 CAP", NULL, "AIF3_CAP Mixer"},
+
+ WCD934X_SLIM_TX_AIF_PATH(0),
+ WCD934X_SLIM_TX_AIF_PATH(1),
+ WCD934X_SLIM_TX_AIF_PATH(2),
+ WCD934X_SLIM_TX_AIF_PATH(3),
+ WCD934X_SLIM_TX_AIF_PATH(4),
+ WCD934X_SLIM_TX_AIF_PATH(5),
+ WCD934X_SLIM_TX_AIF_PATH(6),
+ WCD934X_SLIM_TX_AIF_PATH(7),
+ WCD934X_SLIM_TX_AIF_PATH(8),
+
+ WCD934X_ADC_MUX(0),
+ WCD934X_ADC_MUX(1),
+ WCD934X_ADC_MUX(2),
+ WCD934X_ADC_MUX(3),
+ WCD934X_ADC_MUX(4),
+ WCD934X_ADC_MUX(5),
+ WCD934X_ADC_MUX(6),
+ WCD934X_ADC_MUX(7),
+ WCD934X_ADC_MUX(8),
+
+ {"CDC_IF TX0 MUX", "DEC0", "ADC MUX0"},
+ {"CDC_IF TX1 MUX", "DEC1", "ADC MUX1"},
+ {"CDC_IF TX2 MUX", "DEC2", "ADC MUX2"},
+ {"CDC_IF TX3 MUX", "DEC3", "ADC MUX3"},
+ {"CDC_IF TX4 MUX", "DEC4", "ADC MUX4"},
+ {"CDC_IF TX5 MUX", "DEC5", "ADC MUX5"},
+ {"CDC_IF TX6 MUX", "DEC6", "ADC MUX6"},
+ {"CDC_IF TX7 MUX", "DEC7", "ADC MUX7"},
+ {"CDC_IF TX8 MUX", "DEC8", "ADC MUX8"},
+
+ {"AMIC4_5 SEL", "AMIC4", "AMIC4"},
+ {"AMIC4_5 SEL", "AMIC5", "AMIC5"},
+
+ { "DMIC0", NULL, "DMIC0 Pin" },
+ { "DMIC1", NULL, "DMIC1 Pin" },
+ { "DMIC2", NULL, "DMIC2 Pin" },
+ { "DMIC3", NULL, "DMIC3 Pin" },
+ { "DMIC4", NULL, "DMIC4 Pin" },
+ { "DMIC5", NULL, "DMIC5 Pin" },
+
+ {"ADC1", NULL, "AMIC1"},
+ {"ADC2", NULL, "AMIC2"},
+ {"ADC3", NULL, "AMIC3"},
+ {"ADC4", NULL, "AMIC4_5 SEL"},
+
+ WCD934X_IIR_INP_MUX(0),
+ WCD934X_IIR_INP_MUX(1),
+
+ {"SRC0", NULL, "IIR0"},
+ {"SRC1", NULL, "IIR1"},
+};
+
static const struct snd_soc_component_driver wcd934x_component_drv = {
.probe = wcd934x_comp_probe,
.remove = wcd934x_comp_remove,
@@ -4686,6 +4974,8 @@ static const struct snd_soc_component_driver wcd934x_component_drv = {
.num_controls = ARRAY_SIZE(wcd934x_snd_controls),
.dapm_widgets = wcd934x_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wcd934x_dapm_widgets),
+ .dapm_routes = wcd934x_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(wcd934x_audio_map),
};
static int wcd934x_codec_parse_data(struct wcd934x_codec *wcd)
--
2.20.1
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^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: wcd934x: add capture dapm widgets" to the asoc tree
@ 2019-12-25 18:21 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2019-12-25 18:21 UTC (permalink / raw)
To: Srinivas Kandagatla; +Cc: alsa-devel, Mark Brown
The patch
ASoC: wcd934x: add capture dapm widgets
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From a70d9245759a48e57bb1dc9f63213dcf3017db32 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Thu, 19 Dec 2019 10:31:48 +0000
Subject: [PATCH] ASoC: wcd934x: add capture dapm widgets
This patch adds required dapm widgets for capture path.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20191219103153.14875-7-srinivas.kandagatla@linaro.org
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wcd934x.c | 1151 +++++++++++++++++++++++++++++++++++-
1 file changed, 1149 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c
index 556c051f7f99..24f78a0b7f5a 100644
--- a/sound/soc/codecs/wcd934x.c
+++ b/sound/soc/codecs/wcd934x.c
@@ -135,6 +135,13 @@
} \
}
+enum {
+ MIC_BIAS_1 = 1,
+ MIC_BIAS_2,
+ MIC_BIAS_3,
+ MIC_BIAS_4
+};
+
enum {
SIDO_SOURCE_INTERNAL,
SIDO_SOURCE_RCO_BG,
@@ -500,6 +507,83 @@ static const char * const rx_int8_2_interp_mux_text[] = {
"ZERO", "RX INT8_2 MUX",
};
+static const char * const dmic_mux_text[] = {
+ "ZERO", "DMIC0", "DMIC1", "DMIC2", "DMIC3", "DMIC4", "DMIC5"
+};
+
+static const char * const amic_mux_text[] = {
+ "ZERO", "ADC1", "ADC2", "ADC3", "ADC4"
+};
+
+static const char * const amic4_5_sel_text[] = {
+ "AMIC4", "AMIC5"
+};
+
+static const char * const adc_mux_text[] = {
+ "DMIC", "AMIC", "ANC_FB_TUNE1", "ANC_FB_TUNE2"
+};
+
+static const char * const cdc_if_tx0_mux_text[] = {
+ "ZERO", "RX_MIX_TX0", "DEC0", "DEC0_192"
+};
+
+static const char * const cdc_if_tx1_mux_text[] = {
+ "ZERO", "RX_MIX_TX1", "DEC1", "DEC1_192"
+};
+
+static const char * const cdc_if_tx2_mux_text[] = {
+ "ZERO", "RX_MIX_TX2", "DEC2", "DEC2_192"
+};
+
+static const char * const cdc_if_tx3_mux_text[] = {
+ "ZERO", "RX_MIX_TX3", "DEC3", "DEC3_192"
+};
+
+static const char * const cdc_if_tx4_mux_text[] = {
+ "ZERO", "RX_MIX_TX4", "DEC4", "DEC4_192"
+};
+
+static const char * const cdc_if_tx5_mux_text[] = {
+ "ZERO", "RX_MIX_TX5", "DEC5", "DEC5_192"
+};
+
+static const char * const cdc_if_tx6_mux_text[] = {
+ "ZERO", "RX_MIX_TX6", "DEC6", "DEC6_192"
+};
+
+static const char * const cdc_if_tx7_mux_text[] = {
+ "ZERO", "RX_MIX_TX7", "DEC7", "DEC7_192"
+};
+
+static const char * const cdc_if_tx8_mux_text[] = {
+ "ZERO", "RX_MIX_TX8", "DEC8", "DEC8_192"
+};
+
+static const char * const cdc_if_tx9_mux_text[] = {
+ "ZERO", "DEC7", "DEC7_192"
+};
+
+static const char * const cdc_if_tx10_mux_text[] = {
+ "ZERO", "DEC6", "DEC6_192"
+};
+
+static const char * const cdc_if_tx11_mux_text[] = {
+ "DEC_0_5", "DEC_9_12", "MAD_AUDIO", "MAD_BRDCST"
+};
+
+static const char * const cdc_if_tx11_inp1_mux_text[] = {
+ "ZERO", "DEC0", "DEC1", "DEC2", "DEC3", "DEC4",
+ "DEC5", "RX_MIX_TX5", "DEC9_10", "DEC11_12"
+};
+
+static const char * const cdc_if_tx13_mux_text[] = {
+ "CDC_DEC_5", "MAD_BRDCST"
+};
+
+static const char * const cdc_if_tx13_inp1_mux_text[] = {
+ "ZERO", "DEC5", "DEC5_192"
+};
+
static const struct soc_enum cf_dec0_enum =
SOC_ENUM_SINGLE(WCD934X_CDC_TX0_TX_PATH_CFG0, 5, 3, cf_text);
@@ -758,6 +842,35 @@ static const struct soc_enum rx_int2_dem_inp_mux_enum =
SOC_ENUM_SINGLE(WCD934X_CDC_RX2_RX_PATH_SEC0, 0,
ARRAY_SIZE(rx_int_dem_inp_mux_text),
rx_int_dem_inp_mux_text);
+
+static const struct soc_enum tx_adc_mux0_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1, 0,
+ ARRAY_SIZE(adc_mux_text), adc_mux_text);
+static const struct soc_enum tx_adc_mux1_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX1_CFG1, 0,
+ ARRAY_SIZE(adc_mux_text), adc_mux_text);
+static const struct soc_enum tx_adc_mux2_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX2_CFG1, 0,
+ ARRAY_SIZE(adc_mux_text), adc_mux_text);
+static const struct soc_enum tx_adc_mux3_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG1, 0,
+ ARRAY_SIZE(adc_mux_text), adc_mux_text);
+static const struct soc_enum tx_adc_mux4_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1, 2,
+ ARRAY_SIZE(adc_mux_text), adc_mux_text);
+static const struct soc_enum tx_adc_mux5_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX1_CFG1, 2,
+ ARRAY_SIZE(adc_mux_text), adc_mux_text);
+static const struct soc_enum tx_adc_mux6_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX2_CFG1, 2,
+ ARRAY_SIZE(adc_mux_text), adc_mux_text);
+static const struct soc_enum tx_adc_mux7_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG1, 2,
+ ARRAY_SIZE(adc_mux_text), adc_mux_text);
+static const struct soc_enum tx_adc_mux8_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX1_CFG1, 4,
+ ARRAY_SIZE(adc_mux_text), adc_mux_text);
+
static const struct soc_enum rx_int0_1_interp_mux_enum =
SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2,
rx_int0_1_interp_mux_text);
@@ -803,8 +916,122 @@ static const struct soc_enum rx_int7_2_interp_mux_enum =
static const struct soc_enum rx_int8_2_interp_mux_enum =
SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int8_2_interp_mux_text);
-static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd,
- int sido_src)
+static const struct soc_enum tx_dmic_mux0_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG0, 3, 7,
+ dmic_mux_text);
+
+static const struct soc_enum tx_dmic_mux1_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX1_CFG0, 3, 7,
+ dmic_mux_text);
+
+static const struct soc_enum tx_dmic_mux2_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX2_CFG0, 3, 7,
+ dmic_mux_text);
+
+static const struct soc_enum tx_dmic_mux3_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG0, 3, 7,
+ dmic_mux_text);
+
+static const struct soc_enum tx_dmic_mux4_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0, 3, 7,
+ dmic_mux_text);
+
+static const struct soc_enum tx_dmic_mux5_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX5_CFG0, 3, 7,
+ dmic_mux_text);
+
+static const struct soc_enum tx_dmic_mux6_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX6_CFG0, 3, 7,
+ dmic_mux_text);
+
+static const struct soc_enum tx_dmic_mux7_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX7_CFG0, 3, 7,
+ dmic_mux_text);
+
+static const struct soc_enum tx_dmic_mux8_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX8_CFG0, 3, 7,
+ dmic_mux_text);
+
+static const struct soc_enum tx_amic_mux0_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG0, 0, 5,
+ amic_mux_text);
+static const struct soc_enum tx_amic_mux1_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX1_CFG0, 0, 5,
+ amic_mux_text);
+static const struct soc_enum tx_amic_mux2_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX2_CFG0, 0, 5,
+ amic_mux_text);
+static const struct soc_enum tx_amic_mux3_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG0, 0, 5,
+ amic_mux_text);
+static const struct soc_enum tx_amic_mux4_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0, 0, 5,
+ amic_mux_text);
+static const struct soc_enum tx_amic_mux5_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX5_CFG0, 0, 5,
+ amic_mux_text);
+static const struct soc_enum tx_amic_mux6_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX6_CFG0, 0, 5,
+ amic_mux_text);
+static const struct soc_enum tx_amic_mux7_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX7_CFG0, 0, 5,
+ amic_mux_text);
+static const struct soc_enum tx_amic_mux8_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX_INP_MUX_ADC_MUX8_CFG0, 0, 5,
+ amic_mux_text);
+
+static const struct soc_enum tx_amic4_5_enum =
+ SOC_ENUM_SINGLE(WCD934X_TX_NEW_AMIC_4_5_SEL, 7, 2, amic4_5_sel_text);
+
+static const struct soc_enum cdc_if_tx0_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG0, 0,
+ ARRAY_SIZE(cdc_if_tx0_mux_text), cdc_if_tx0_mux_text);
+static const struct soc_enum cdc_if_tx1_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG0, 2,
+ ARRAY_SIZE(cdc_if_tx1_mux_text), cdc_if_tx1_mux_text);
+static const struct soc_enum cdc_if_tx2_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG0, 4,
+ ARRAY_SIZE(cdc_if_tx2_mux_text), cdc_if_tx2_mux_text);
+static const struct soc_enum cdc_if_tx3_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG0, 6,
+ ARRAY_SIZE(cdc_if_tx3_mux_text), cdc_if_tx3_mux_text);
+static const struct soc_enum cdc_if_tx4_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG1, 0,
+ ARRAY_SIZE(cdc_if_tx4_mux_text), cdc_if_tx4_mux_text);
+static const struct soc_enum cdc_if_tx5_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG1, 2,
+ ARRAY_SIZE(cdc_if_tx5_mux_text), cdc_if_tx5_mux_text);
+static const struct soc_enum cdc_if_tx6_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG1, 4,
+ ARRAY_SIZE(cdc_if_tx6_mux_text), cdc_if_tx6_mux_text);
+static const struct soc_enum cdc_if_tx7_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG1, 6,
+ ARRAY_SIZE(cdc_if_tx7_mux_text), cdc_if_tx7_mux_text);
+static const struct soc_enum cdc_if_tx8_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG2, 0,
+ ARRAY_SIZE(cdc_if_tx8_mux_text), cdc_if_tx8_mux_text);
+static const struct soc_enum cdc_if_tx9_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG2, 2,
+ ARRAY_SIZE(cdc_if_tx9_mux_text), cdc_if_tx9_mux_text);
+static const struct soc_enum cdc_if_tx10_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG2, 4,
+ ARRAY_SIZE(cdc_if_tx10_mux_text), cdc_if_tx10_mux_text);
+static const struct soc_enum cdc_if_tx11_inp1_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG3, 0,
+ ARRAY_SIZE(cdc_if_tx11_inp1_mux_text),
+ cdc_if_tx11_inp1_mux_text);
+static const struct soc_enum cdc_if_tx11_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_DATA_HUB_SB_TX11_INP_CFG, 0,
+ ARRAY_SIZE(cdc_if_tx11_mux_text), cdc_if_tx11_mux_text);
+static const struct soc_enum cdc_if_tx13_inp1_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_IF_ROUTER_TX_MUX_CFG3, 4,
+ ARRAY_SIZE(cdc_if_tx13_inp1_mux_text),
+ cdc_if_tx13_inp1_mux_text);
+static const struct soc_enum cdc_if_tx13_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_DATA_HUB_SB_TX13_INP_CFG, 0,
+ ARRAY_SIZE(cdc_if_tx13_mux_text), cdc_if_tx13_mux_text);
+
+static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd, int sido_src)
{
if (sido_src == wcd->sido_input_src)
return 0;
@@ -2241,6 +2468,63 @@ static int wcd934x_int_dem_inp_mux_put(struct snd_kcontrol *kc,
return ret;
}
+static int wcd934x_dec_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp;
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int val;
+ u16 mic_sel_reg = 0;
+ u8 mic_sel;
+
+ comp = snd_soc_dapm_kcontrol_component(kcontrol);
+
+ val = ucontrol->value.enumerated.item[0];
+ if (val > e->items - 1)
+ return -EINVAL;
+
+ switch (e->reg) {
+ case WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1:
+ if (e->shift_l == 0)
+ mic_sel_reg = WCD934X_CDC_TX0_TX_PATH_CFG0;
+ else if (e->shift_l == 2)
+ mic_sel_reg = WCD934X_CDC_TX4_TX_PATH_CFG0;
+ else if (e->shift_l == 4)
+ mic_sel_reg = WCD934X_CDC_TX8_TX_PATH_CFG0;
+ break;
+ case WCD934X_CDC_TX_INP_MUX_ADC_MUX1_CFG1:
+ if (e->shift_l == 0)
+ mic_sel_reg = WCD934X_CDC_TX1_TX_PATH_CFG0;
+ else if (e->shift_l == 2)
+ mic_sel_reg = WCD934X_CDC_TX5_TX_PATH_CFG0;
+ break;
+ case WCD934X_CDC_TX_INP_MUX_ADC_MUX2_CFG1:
+ if (e->shift_l == 0)
+ mic_sel_reg = WCD934X_CDC_TX2_TX_PATH_CFG0;
+ else if (e->shift_l == 2)
+ mic_sel_reg = WCD934X_CDC_TX6_TX_PATH_CFG0;
+ break;
+ case WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG1:
+ if (e->shift_l == 0)
+ mic_sel_reg = WCD934X_CDC_TX3_TX_PATH_CFG0;
+ else if (e->shift_l == 2)
+ mic_sel_reg = WCD934X_CDC_TX7_TX_PATH_CFG0;
+ break;
+ default:
+ dev_err(comp->dev, "%s: e->reg: 0x%x not expected\n",
+ __func__, e->reg);
+ return -EINVAL;
+ }
+
+ /* ADC: 0, DMIC: 1 */
+ mic_sel = val ? 0x0 : 0x1;
+ if (mic_sel_reg)
+ snd_soc_component_update_bits(comp, mic_sel_reg, BIT(7),
+ mic_sel << 7);
+
+ return snd_soc_dapm_put_enum_double(kcontrol, ucontrol);
+}
+
static const struct snd_kcontrol_new rx_int0_2_mux =
SOC_DAPM_ENUM("RX INT0_2 MUX Mux", rx_int0_2_mux_chain_enum);
@@ -2453,6 +2737,252 @@ static const struct snd_kcontrol_new rx_int7_2_interp_mux =
static const struct snd_kcontrol_new rx_int8_2_interp_mux =
SOC_DAPM_ENUM("RX INT8_2 INTERP Mux", rx_int8_2_interp_mux_enum);
+static const struct snd_kcontrol_new tx_dmic_mux0 =
+ SOC_DAPM_ENUM("DMIC MUX0 Mux", tx_dmic_mux0_enum);
+
+static const struct snd_kcontrol_new tx_dmic_mux1 =
+ SOC_DAPM_ENUM("DMIC MUX1 Mux", tx_dmic_mux1_enum);
+
+static const struct snd_kcontrol_new tx_dmic_mux2 =
+ SOC_DAPM_ENUM("DMIC MUX2 Mux", tx_dmic_mux2_enum);
+
+static const struct snd_kcontrol_new tx_dmic_mux3 =
+ SOC_DAPM_ENUM("DMIC MUX3 Mux", tx_dmic_mux3_enum);
+
+static const struct snd_kcontrol_new tx_dmic_mux4 =
+ SOC_DAPM_ENUM("DMIC MUX4 Mux", tx_dmic_mux4_enum);
+
+static const struct snd_kcontrol_new tx_dmic_mux5 =
+ SOC_DAPM_ENUM("DMIC MUX5 Mux", tx_dmic_mux5_enum);
+
+static const struct snd_kcontrol_new tx_dmic_mux6 =
+ SOC_DAPM_ENUM("DMIC MUX6 Mux", tx_dmic_mux6_enum);
+
+static const struct snd_kcontrol_new tx_dmic_mux7 =
+ SOC_DAPM_ENUM("DMIC MUX7 Mux", tx_dmic_mux7_enum);
+
+static const struct snd_kcontrol_new tx_dmic_mux8 =
+ SOC_DAPM_ENUM("DMIC MUX8 Mux", tx_dmic_mux8_enum);
+
+static const struct snd_kcontrol_new tx_amic_mux0 =
+ SOC_DAPM_ENUM("AMIC MUX0 Mux", tx_amic_mux0_enum);
+
+static const struct snd_kcontrol_new tx_amic_mux1 =
+ SOC_DAPM_ENUM("AMIC MUX1 Mux", tx_amic_mux1_enum);
+
+static const struct snd_kcontrol_new tx_amic_mux2 =
+ SOC_DAPM_ENUM("AMIC MUX2 Mux", tx_amic_mux2_enum);
+
+static const struct snd_kcontrol_new tx_amic_mux3 =
+ SOC_DAPM_ENUM("AMIC MUX3 Mux", tx_amic_mux3_enum);
+
+static const struct snd_kcontrol_new tx_amic_mux4 =
+ SOC_DAPM_ENUM("AMIC MUX4 Mux", tx_amic_mux4_enum);
+
+static const struct snd_kcontrol_new tx_amic_mux5 =
+ SOC_DAPM_ENUM("AMIC MUX5 Mux", tx_amic_mux5_enum);
+
+static const struct snd_kcontrol_new tx_amic_mux6 =
+ SOC_DAPM_ENUM("AMIC MUX6 Mux", tx_amic_mux6_enum);
+
+static const struct snd_kcontrol_new tx_amic_mux7 =
+ SOC_DAPM_ENUM("AMIC MUX7 Mux", tx_amic_mux7_enum);
+
+static const struct snd_kcontrol_new tx_amic_mux8 =
+ SOC_DAPM_ENUM("AMIC MUX8 Mux", tx_amic_mux8_enum);
+
+static const struct snd_kcontrol_new tx_amic4_5 =
+ SOC_DAPM_ENUM("AMIC4_5 SEL Mux", tx_amic4_5_enum);
+
+static const struct snd_kcontrol_new tx_adc_mux0_mux =
+ SOC_DAPM_ENUM_EXT("ADC MUX0 Mux", tx_adc_mux0_enum,
+ snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put);
+static const struct snd_kcontrol_new tx_adc_mux1_mux =
+ SOC_DAPM_ENUM_EXT("ADC MUX1 Mux", tx_adc_mux1_enum,
+ snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put);
+static const struct snd_kcontrol_new tx_adc_mux2_mux =
+ SOC_DAPM_ENUM_EXT("ADC MUX2 Mux", tx_adc_mux2_enum,
+ snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put);
+static const struct snd_kcontrol_new tx_adc_mux3_mux =
+ SOC_DAPM_ENUM_EXT("ADC MUX3 Mux", tx_adc_mux3_enum,
+ snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put);
+static const struct snd_kcontrol_new tx_adc_mux4_mux =
+ SOC_DAPM_ENUM_EXT("ADC MUX4 Mux", tx_adc_mux4_enum,
+ snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put);
+static const struct snd_kcontrol_new tx_adc_mux5_mux =
+ SOC_DAPM_ENUM_EXT("ADC MUX5 Mux", tx_adc_mux5_enum,
+ snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put);
+static const struct snd_kcontrol_new tx_adc_mux6_mux =
+ SOC_DAPM_ENUM_EXT("ADC MUX6 Mux", tx_adc_mux6_enum,
+ snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put);
+static const struct snd_kcontrol_new tx_adc_mux7_mux =
+ SOC_DAPM_ENUM_EXT("ADC MUX7 Mux", tx_adc_mux7_enum,
+ snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put);
+static const struct snd_kcontrol_new tx_adc_mux8_mux =
+ SOC_DAPM_ENUM_EXT("ADC MUX8 Mux", tx_adc_mux8_enum,
+ snd_soc_dapm_get_enum_double, wcd934x_dec_enum_put);
+
+static const struct snd_kcontrol_new cdc_if_tx0_mux =
+ SOC_DAPM_ENUM("CDC_IF TX0 MUX Mux", cdc_if_tx0_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx1_mux =
+ SOC_DAPM_ENUM("CDC_IF TX1 MUX Mux", cdc_if_tx1_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx2_mux =
+ SOC_DAPM_ENUM("CDC_IF TX2 MUX Mux", cdc_if_tx2_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx3_mux =
+ SOC_DAPM_ENUM("CDC_IF TX3 MUX Mux", cdc_if_tx3_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx4_mux =
+ SOC_DAPM_ENUM("CDC_IF TX4 MUX Mux", cdc_if_tx4_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx5_mux =
+ SOC_DAPM_ENUM("CDC_IF TX5 MUX Mux", cdc_if_tx5_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx6_mux =
+ SOC_DAPM_ENUM("CDC_IF TX6 MUX Mux", cdc_if_tx6_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx7_mux =
+ SOC_DAPM_ENUM("CDC_IF TX7 MUX Mux", cdc_if_tx7_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx8_mux =
+ SOC_DAPM_ENUM("CDC_IF TX8 MUX Mux", cdc_if_tx8_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx9_mux =
+ SOC_DAPM_ENUM("CDC_IF TX9 MUX Mux", cdc_if_tx9_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx10_mux =
+ SOC_DAPM_ENUM("CDC_IF TX10 MUX Mux", cdc_if_tx10_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx11_mux =
+ SOC_DAPM_ENUM("CDC_IF TX11 MUX Mux", cdc_if_tx11_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx11_inp1_mux =
+ SOC_DAPM_ENUM("CDC_IF TX11 INP1 MUX Mux", cdc_if_tx11_inp1_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx13_mux =
+ SOC_DAPM_ENUM("CDC_IF TX13 MUX Mux", cdc_if_tx13_mux_enum);
+static const struct snd_kcontrol_new cdc_if_tx13_inp1_mux =
+ SOC_DAPM_ENUM("CDC_IF TX13 INP1 MUX Mux", cdc_if_tx13_inp1_mux_enum);
+
+static int slim_tx_mixer_get(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc);
+ struct wcd934x_codec *wcd = dev_get_drvdata(dapm->dev);
+ struct soc_mixer_control *mixer =
+ (struct soc_mixer_control *)kc->private_value;
+ int port_id = mixer->shift;
+
+ ucontrol->value.integer.value[0] = wcd->tx_port_value[port_id];
+
+ return 0;
+}
+
+static int slim_tx_mixer_put(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_soc_dapm_kcontrol_widget(kc);
+ struct wcd934x_codec *wcd = dev_get_drvdata(widget->dapm->dev);
+ struct snd_soc_dapm_update *update = NULL;
+ struct soc_mixer_control *mixer =
+ (struct soc_mixer_control *)kc->private_value;
+ int enable = ucontrol->value.integer.value[0];
+ int dai_id = widget->shift;
+ int port_id = mixer->shift;
+
+ /* only add to the list if value not set */
+ if (enable == wcd->tx_port_value[port_id])
+ return 0;
+
+ wcd->tx_port_value[port_id] = enable;
+
+ if (enable)
+ list_add_tail(&wcd->tx_chs[port_id].list,
+ &wcd->dai[dai_id].slim_ch_list);
+ else
+ list_del_init(&wcd->tx_chs[port_id].list);
+
+ snd_soc_dapm_mixer_update_power(widget->dapm, kc, enable, update);
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new aif1_slim_cap_mixer[] = {
+ SOC_SINGLE_EXT("SLIM TX0", SND_SOC_NOPM, WCD934X_TX0, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX1", SND_SOC_NOPM, WCD934X_TX1, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX2", SND_SOC_NOPM, WCD934X_TX2, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX3", SND_SOC_NOPM, WCD934X_TX3, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX4", SND_SOC_NOPM, WCD934X_TX4, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX5", SND_SOC_NOPM, WCD934X_TX5, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX6", SND_SOC_NOPM, WCD934X_TX6, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX7", SND_SOC_NOPM, WCD934X_TX7, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX8", SND_SOC_NOPM, WCD934X_TX8, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX9", SND_SOC_NOPM, WCD934X_TX9, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX10", SND_SOC_NOPM, WCD934X_TX10, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX11", SND_SOC_NOPM, WCD934X_TX11, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX13", SND_SOC_NOPM, WCD934X_TX13, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+};
+
+static const struct snd_kcontrol_new aif2_slim_cap_mixer[] = {
+ SOC_SINGLE_EXT("SLIM TX0", SND_SOC_NOPM, WCD934X_TX0, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX1", SND_SOC_NOPM, WCD934X_TX1, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX2", SND_SOC_NOPM, WCD934X_TX2, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX3", SND_SOC_NOPM, WCD934X_TX3, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX4", SND_SOC_NOPM, WCD934X_TX4, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX5", SND_SOC_NOPM, WCD934X_TX5, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX6", SND_SOC_NOPM, WCD934X_TX6, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX7", SND_SOC_NOPM, WCD934X_TX7, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX8", SND_SOC_NOPM, WCD934X_TX8, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX9", SND_SOC_NOPM, WCD934X_TX9, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX10", SND_SOC_NOPM, WCD934X_TX10, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX11", SND_SOC_NOPM, WCD934X_TX11, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX13", SND_SOC_NOPM, WCD934X_TX13, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+};
+
+static const struct snd_kcontrol_new aif3_slim_cap_mixer[] = {
+ SOC_SINGLE_EXT("SLIM TX0", SND_SOC_NOPM, WCD934X_TX0, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX1", SND_SOC_NOPM, WCD934X_TX1, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX2", SND_SOC_NOPM, WCD934X_TX2, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX3", SND_SOC_NOPM, WCD934X_TX3, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX4", SND_SOC_NOPM, WCD934X_TX4, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX5", SND_SOC_NOPM, WCD934X_TX5, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX6", SND_SOC_NOPM, WCD934X_TX6, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX7", SND_SOC_NOPM, WCD934X_TX7, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX8", SND_SOC_NOPM, WCD934X_TX8, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX9", SND_SOC_NOPM, WCD934X_TX9, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX10", SND_SOC_NOPM, WCD934X_TX10, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX11", SND_SOC_NOPM, WCD934X_TX11, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+ SOC_SINGLE_EXT("SLIM TX13", SND_SOC_NOPM, WCD934X_TX13, 1, 0,
+ slim_tx_mixer_get, slim_tx_mixer_put),
+};
+
static const struct snd_kcontrol_new wcd934x_snd_controls[] = {
/* Gain Controls */
SOC_SINGLE_TLV("EAR PA Volume", WCD934X_ANA_EAR, 4, 4, 1, ear_pa_gain),
@@ -3193,6 +3723,455 @@ static int wcd934x_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w,
return 0;
}
+static u32 wcd934x_get_dmic_sample_rate(struct snd_soc_component *comp,
+ unsigned int dmic,
+ struct wcd934x_codec *wcd)
+{
+ u8 tx_stream_fs;
+ u8 adc_mux_index = 0, adc_mux_sel = 0;
+ bool dec_found = false;
+ u16 adc_mux_ctl_reg, tx_fs_reg;
+ u32 dmic_fs;
+
+ while (dec_found == 0 && adc_mux_index < WCD934X_MAX_VALID_ADC_MUX) {
+ if (adc_mux_index < 4) {
+ adc_mux_ctl_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG0 +
+ (adc_mux_index * 2);
+ } else if (adc_mux_index < WCD934X_INVALID_ADC_MUX) {
+ adc_mux_ctl_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0 +
+ adc_mux_index - 4;
+ } else if (adc_mux_index == WCD934X_INVALID_ADC_MUX) {
+ ++adc_mux_index;
+ continue;
+ }
+ adc_mux_sel = ((snd_soc_component_read32(comp, adc_mux_ctl_reg)
+ & 0xF8) >> 3) - 1;
+
+ if (adc_mux_sel == dmic) {
+ dec_found = true;
+ break;
+ }
+
+ ++adc_mux_index;
+ }
+
+ if (dec_found && adc_mux_index <= 8) {
+ tx_fs_reg = WCD934X_CDC_TX0_TX_PATH_CTL + (16 * adc_mux_index);
+ tx_stream_fs = snd_soc_component_read32(comp, tx_fs_reg) & 0x0F;
+ if (tx_stream_fs <= 4) {
+ if (wcd->dmic_sample_rate <=
+ WCD9XXX_DMIC_SAMPLE_RATE_2P4MHZ)
+ dmic_fs = wcd->dmic_sample_rate;
+ else
+ dmic_fs = WCD9XXX_DMIC_SAMPLE_RATE_2P4MHZ;
+ } else
+ dmic_fs = WCD9XXX_DMIC_SAMPLE_RATE_4P8MHZ;
+ } else {
+ dmic_fs = wcd->dmic_sample_rate;
+ }
+
+ return dmic_fs;
+}
+
+static u8 wcd934x_get_dmic_clk_val(struct snd_soc_component *comp,
+ u32 mclk_rate, u32 dmic_clk_rate)
+{
+ u32 div_factor;
+ u8 dmic_ctl_val;
+
+ /* Default value to return in case of error */
+ if (mclk_rate == WCD934X_MCLK_CLK_9P6MHZ)
+ dmic_ctl_val = WCD934X_DMIC_CLK_DIV_2;
+ else
+ dmic_ctl_val = WCD934X_DMIC_CLK_DIV_3;
+
+ if (dmic_clk_rate == 0) {
+ dev_err(comp->dev,
+ "%s: dmic_sample_rate cannot be 0\n",
+ __func__);
+ goto done;
+ }
+
+ div_factor = mclk_rate / dmic_clk_rate;
+ switch (div_factor) {
+ case 2:
+ dmic_ctl_val = WCD934X_DMIC_CLK_DIV_2;
+ break;
+ case 3:
+ dmic_ctl_val = WCD934X_DMIC_CLK_DIV_3;
+ break;
+ case 4:
+ dmic_ctl_val = WCD934X_DMIC_CLK_DIV_4;
+ break;
+ case 6:
+ dmic_ctl_val = WCD934X_DMIC_CLK_DIV_6;
+ break;
+ case 8:
+ dmic_ctl_val = WCD934X_DMIC_CLK_DIV_8;
+ break;
+ case 16:
+ dmic_ctl_val = WCD934X_DMIC_CLK_DIV_16;
+ break;
+ default:
+ dev_err(comp->dev,
+ "%s: Invalid div_factor %u, clk_rate(%u), dmic_rate(%u)\n",
+ __func__, div_factor, mclk_rate, dmic_clk_rate);
+ break;
+ }
+
+done:
+ return dmic_ctl_val;
+}
+
+static int wcd934x_codec_enable_dmic(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev);
+ u8 dmic_clk_en = 0x01;
+ u16 dmic_clk_reg;
+ s32 *dmic_clk_cnt;
+ u8 dmic_rate_val, dmic_rate_shift = 1;
+ unsigned int dmic;
+ u32 dmic_sample_rate;
+ int ret;
+ char *wname;
+
+ wname = strpbrk(w->name, "012345");
+ if (!wname) {
+ dev_err(comp->dev, "%s: widget not found\n", __func__);
+ return -EINVAL;
+ }
+
+ ret = kstrtouint(wname, 10, &dmic);
+ if (ret < 0) {
+ dev_err(comp->dev, "%s: Invalid DMIC line on the codec\n",
+ __func__);
+ return -EINVAL;
+ }
+
+ switch (dmic) {
+ case 0:
+ case 1:
+ dmic_clk_cnt = &wcd->dmic_0_1_clk_cnt;
+ dmic_clk_reg = WCD934X_CPE_SS_DMIC0_CTL;
+ break;
+ case 2:
+ case 3:
+ dmic_clk_cnt = &wcd->dmic_2_3_clk_cnt;
+ dmic_clk_reg = WCD934X_CPE_SS_DMIC1_CTL;
+ break;
+ case 4:
+ case 5:
+ dmic_clk_cnt = &wcd->dmic_4_5_clk_cnt;
+ dmic_clk_reg = WCD934X_CPE_SS_DMIC2_CTL;
+ break;
+ default:
+ dev_err(comp->dev, "%s: Invalid DMIC Selection\n",
+ __func__);
+ return -EINVAL;
+ };
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ dmic_sample_rate = wcd934x_get_dmic_sample_rate(comp, dmic,
+ wcd);
+ dmic_rate_val = wcd934x_get_dmic_clk_val(comp, wcd->rate,
+ dmic_sample_rate);
+ (*dmic_clk_cnt)++;
+ if (*dmic_clk_cnt == 1) {
+ dmic_rate_val = dmic_rate_val << dmic_rate_shift;
+ snd_soc_component_update_bits(comp, dmic_clk_reg,
+ WCD934X_DMIC_RATE_MASK,
+ dmic_rate_val);
+ snd_soc_component_update_bits(comp, dmic_clk_reg,
+ dmic_clk_en, dmic_clk_en);
+ }
+
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ (*dmic_clk_cnt)--;
+ if (*dmic_clk_cnt == 0)
+ snd_soc_component_update_bits(comp, dmic_clk_reg,
+ dmic_clk_en, 0);
+ break;
+ };
+
+ return 0;
+}
+
+static int wcd934x_codec_find_amic_input(struct snd_soc_component *comp,
+ int adc_mux_n)
+{
+ u16 mask, shift, adc_mux_in_reg;
+ u16 amic_mux_sel_reg;
+ bool is_amic;
+
+ if (adc_mux_n < 0 || adc_mux_n > WCD934X_MAX_VALID_ADC_MUX ||
+ adc_mux_n == WCD934X_INVALID_ADC_MUX)
+ return 0;
+
+ if (adc_mux_n < 3) {
+ adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1 +
+ adc_mux_n;
+ mask = 0x03;
+ shift = 0;
+ amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG0 +
+ 2 * adc_mux_n;
+ } else if (adc_mux_n < 4) {
+ adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG1;
+ mask = 0x03;
+ shift = 0;
+ amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG0 +
+ 2 * adc_mux_n;
+ } else if (adc_mux_n < 7) {
+ adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1 +
+ (adc_mux_n - 4);
+ mask = 0x0C;
+ shift = 2;
+ amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0 +
+ adc_mux_n - 4;
+ } else if (adc_mux_n < 8) {
+ adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG1;
+ mask = 0x0C;
+ shift = 2;
+ amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0 +
+ adc_mux_n - 4;
+ } else if (adc_mux_n < 12) {
+ adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1 +
+ ((adc_mux_n == 8) ? (adc_mux_n - 8) :
+ (adc_mux_n - 9));
+ mask = 0x30;
+ shift = 4;
+ amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0 +
+ adc_mux_n - 4;
+ } else if (adc_mux_n < 13) {
+ adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX3_CFG1;
+ mask = 0x30;
+ shift = 4;
+ amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0 +
+ adc_mux_n - 4;
+ } else {
+ adc_mux_in_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX0_CFG1;
+ mask = 0xC0;
+ shift = 6;
+ amic_mux_sel_reg = WCD934X_CDC_TX_INP_MUX_ADC_MUX4_CFG0 +
+ adc_mux_n - 4;
+ }
+
+ is_amic = (((snd_soc_component_read32(comp, adc_mux_in_reg)
+ & mask) >> shift) == 1);
+ if (!is_amic)
+ return 0;
+
+ return snd_soc_component_read32(comp, amic_mux_sel_reg) & 0x07;
+}
+
+static u16 wcd934x_codec_get_amic_pwlvl_reg(struct snd_soc_component *comp,
+ int amic)
+{
+ u16 pwr_level_reg = 0;
+
+ switch (amic) {
+ case 1:
+ case 2:
+ pwr_level_reg = WCD934X_ANA_AMIC1;
+ break;
+
+ case 3:
+ case 4:
+ pwr_level_reg = WCD934X_ANA_AMIC3;
+ break;
+ default:
+ break;
+ }
+
+ return pwr_level_reg;
+}
+
+static int wcd934x_codec_enable_dec(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ unsigned int decimator;
+ char *dec_adc_mux_name = NULL;
+ char *widget_name = NULL;
+ char *wname;
+ int ret = 0, amic_n;
+ u16 tx_vol_ctl_reg, pwr_level_reg = 0, dec_cfg_reg, hpf_gate_reg;
+ u16 tx_gain_ctl_reg;
+ char *dec;
+ u8 hpf_coff_freq;
+
+ widget_name = kstrndup(w->name, 15, GFP_KERNEL);
+ if (!widget_name)
+ return -ENOMEM;
+
+ wname = widget_name;
+ dec_adc_mux_name = strsep(&widget_name, " ");
+ if (!dec_adc_mux_name) {
+ dev_err(comp->dev, "%s: Invalid decimator = %s\n",
+ __func__, w->name);
+ ret = -EINVAL;
+ goto out;
+ }
+ dec_adc_mux_name = widget_name;
+
+ dec = strpbrk(dec_adc_mux_name, "012345678");
+ if (!dec) {
+ dev_err(comp->dev, "%s: decimator index not found\n",
+ __func__);
+ ret = -EINVAL;
+ goto out;
+ }
+
+ ret = kstrtouint(dec, 10, &decimator);
+ if (ret < 0) {
+ dev_err(comp->dev, "%s: Invalid decimator = %s\n",
+ __func__, wname);
+ ret = -EINVAL;
+ goto out;
+ }
+
+ tx_vol_ctl_reg = WCD934X_CDC_TX0_TX_PATH_CTL + 16 * decimator;
+ hpf_gate_reg = WCD934X_CDC_TX0_TX_PATH_SEC2 + 16 * decimator;
+ dec_cfg_reg = WCD934X_CDC_TX0_TX_PATH_CFG0 + 16 * decimator;
+ tx_gain_ctl_reg = WCD934X_CDC_TX0_TX_VOL_CTL + 16 * decimator;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ amic_n = wcd934x_codec_find_amic_input(comp, decimator);
+ if (amic_n)
+ pwr_level_reg = wcd934x_codec_get_amic_pwlvl_reg(comp,
+ amic_n);
+
+ if (!pwr_level_reg)
+ break;
+
+ switch ((snd_soc_component_read32(comp, pwr_level_reg) &
+ WCD934X_AMIC_PWR_LVL_MASK) >>
+ WCD934X_AMIC_PWR_LVL_SHIFT) {
+ case WCD934X_AMIC_PWR_LEVEL_LP:
+ snd_soc_component_update_bits(comp, dec_cfg_reg,
+ WCD934X_DEC_PWR_LVL_MASK,
+ WCD934X_DEC_PWR_LVL_LP);
+ break;
+ case WCD934X_AMIC_PWR_LEVEL_HP:
+ snd_soc_component_update_bits(comp, dec_cfg_reg,
+ WCD934X_DEC_PWR_LVL_MASK,
+ WCD934X_DEC_PWR_LVL_HP);
+ break;
+ case WCD934X_AMIC_PWR_LEVEL_DEFAULT:
+ case WCD934X_AMIC_PWR_LEVEL_HYBRID:
+ default:
+ snd_soc_component_update_bits(comp, dec_cfg_reg,
+ WCD934X_DEC_PWR_LVL_MASK,
+ WCD934X_DEC_PWR_LVL_DF);
+ break;
+ }
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ hpf_coff_freq = (snd_soc_component_read32(comp, dec_cfg_reg) &
+ TX_HPF_CUT_OFF_FREQ_MASK) >> 5;
+ if (hpf_coff_freq != CF_MIN_3DB_150HZ) {
+ snd_soc_component_update_bits(comp, dec_cfg_reg,
+ TX_HPF_CUT_OFF_FREQ_MASK,
+ CF_MIN_3DB_150HZ << 5);
+ snd_soc_component_update_bits(comp, hpf_gate_reg,
+ WCD934X_HPH_CUTOFF_FREQ_CHANGE_REQ_MASK,
+ WCD934X_HPH_CUTOFF_FREQ_CHANGE_REQ);
+ /*
+ * Minimum 1 clk cycle delay is required as per
+ * HW spec.
+ */
+ usleep_range(1000, 1010);
+ snd_soc_component_update_bits(comp, hpf_gate_reg,
+ WCD934X_HPH_CUTOFF_FREQ_CHANGE_REQ_MASK,
+ 0);
+ }
+ /* apply gain after decimator is enabled */
+ snd_soc_component_write(comp, tx_gain_ctl_reg,
+ snd_soc_component_read32(comp,
+ tx_gain_ctl_reg));
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ hpf_coff_freq = (snd_soc_component_read32(comp, dec_cfg_reg) &
+ TX_HPF_CUT_OFF_FREQ_MASK) >> 5;
+
+ if (hpf_coff_freq != CF_MIN_3DB_150HZ) {
+ snd_soc_component_update_bits(comp, dec_cfg_reg,
+ TX_HPF_CUT_OFF_FREQ_MASK,
+ hpf_coff_freq << 5);
+ snd_soc_component_update_bits(comp, hpf_gate_reg,
+ WCD934X_HPH_CUTOFF_FREQ_CHANGE_REQ_MASK,
+ WCD934X_HPH_CUTOFF_FREQ_CHANGE_REQ);
+ /*
+ * Minimum 1 clk cycle delay is required as per
+ * HW spec.
+ */
+ usleep_range(1000, 1010);
+ snd_soc_component_update_bits(comp, hpf_gate_reg,
+ WCD934X_HPH_CUTOFF_FREQ_CHANGE_REQ_MASK,
+ 0);
+ }
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_component_update_bits(comp, tx_vol_ctl_reg,
+ 0x10, 0x00);
+ snd_soc_component_update_bits(comp, dec_cfg_reg,
+ WCD934X_DEC_PWR_LVL_MASK,
+ WCD934X_DEC_PWR_LVL_DF);
+ break;
+ };
+out:
+ kfree(wname);
+ return ret;
+}
+
+static void wcd934x_codec_set_tx_hold(struct snd_soc_component *comp,
+ u16 amic_reg, bool set)
+{
+ u8 mask = 0x20;
+ u8 val;
+
+ if (amic_reg == WCD934X_ANA_AMIC1 ||
+ amic_reg == WCD934X_ANA_AMIC3)
+ mask = 0x40;
+
+ val = set ? mask : 0x00;
+
+ switch (amic_reg) {
+ case WCD934X_ANA_AMIC1:
+ case WCD934X_ANA_AMIC2:
+ snd_soc_component_update_bits(comp, WCD934X_ANA_AMIC2,
+ mask, val);
+ break;
+ case WCD934X_ANA_AMIC3:
+ case WCD934X_ANA_AMIC4:
+ snd_soc_component_update_bits(comp, WCD934X_ANA_AMIC4,
+ mask, val);
+ break;
+ default:
+ break;
+ }
+}
+
+static int wcd934x_codec_enable_adc(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ wcd934x_codec_set_tx_hold(comp, w->reg, true);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
static const struct snd_soc_dapm_widget wcd934x_dapm_widgets[] = {
/* Analog Outputs */
SND_SOC_DAPM_OUTPUT("EAR"),
@@ -3529,6 +4508,174 @@ static const struct snd_soc_dapm_widget wcd934x_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0,
wcd934x_codec_enable_mclk,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ /* TX */
+ SND_SOC_DAPM_INPUT("AMIC1"),
+ SND_SOC_DAPM_INPUT("AMIC2"),
+ SND_SOC_DAPM_INPUT("AMIC3"),
+ SND_SOC_DAPM_INPUT("AMIC4"),
+ SND_SOC_DAPM_INPUT("AMIC5"),
+ SND_SOC_DAPM_INPUT("DMIC0 Pin"),
+ SND_SOC_DAPM_INPUT("DMIC1 Pin"),
+ SND_SOC_DAPM_INPUT("DMIC2 Pin"),
+ SND_SOC_DAPM_INPUT("DMIC3 Pin"),
+ SND_SOC_DAPM_INPUT("DMIC4 Pin"),
+ SND_SOC_DAPM_INPUT("DMIC5 Pin"),
+
+ SND_SOC_DAPM_AIF_OUT_E("AIF1 CAP", "AIF1 Capture", 0, SND_SOC_NOPM,
+ AIF1_CAP, 0, wcd934x_codec_enable_slim,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_OUT_E("AIF2 CAP", "AIF2 Capture", 0, SND_SOC_NOPM,
+ AIF2_CAP, 0, wcd934x_codec_enable_slim,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_OUT_E("AIF3 CAP", "AIF3 Capture", 0, SND_SOC_NOPM,
+ AIF3_CAP, 0, wcd934x_codec_enable_slim,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MIXER("SLIM TX0", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM TX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM TX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM TX3", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM TX4", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM TX5", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM TX6", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM TX7", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM TX8", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM TX9", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM TX10", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM TX11", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM TX13", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Digital Mic Inputs */
+ SND_SOC_DAPM_ADC_E("DMIC0", NULL, SND_SOC_NOPM, 0, 0,
+ wcd934x_codec_enable_dmic,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("DMIC1", NULL, SND_SOC_NOPM, 0, 0,
+ wcd934x_codec_enable_dmic,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("DMIC2", NULL, SND_SOC_NOPM, 0, 0,
+ wcd934x_codec_enable_dmic,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("DMIC3", NULL, SND_SOC_NOPM, 0, 0,
+ wcd934x_codec_enable_dmic,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("DMIC4", NULL, SND_SOC_NOPM, 0, 0,
+ wcd934x_codec_enable_dmic,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("DMIC5", NULL, SND_SOC_NOPM, 0, 0,
+ wcd934x_codec_enable_dmic,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX("DMIC MUX0", SND_SOC_NOPM, 0, 0, &tx_dmic_mux0),
+ SND_SOC_DAPM_MUX("DMIC MUX1", SND_SOC_NOPM, 0, 0, &tx_dmic_mux1),
+ SND_SOC_DAPM_MUX("DMIC MUX2", SND_SOC_NOPM, 0, 0, &tx_dmic_mux2),
+ SND_SOC_DAPM_MUX("DMIC MUX3", SND_SOC_NOPM, 0, 0, &tx_dmic_mux3),
+ SND_SOC_DAPM_MUX("DMIC MUX4", SND_SOC_NOPM, 0, 0, &tx_dmic_mux4),
+ SND_SOC_DAPM_MUX("DMIC MUX5", SND_SOC_NOPM, 0, 0, &tx_dmic_mux5),
+ SND_SOC_DAPM_MUX("DMIC MUX6", SND_SOC_NOPM, 0, 0, &tx_dmic_mux6),
+ SND_SOC_DAPM_MUX("DMIC MUX7", SND_SOC_NOPM, 0, 0, &tx_dmic_mux7),
+ SND_SOC_DAPM_MUX("DMIC MUX8", SND_SOC_NOPM, 0, 0, &tx_dmic_mux8),
+ SND_SOC_DAPM_MUX("AMIC MUX0", SND_SOC_NOPM, 0, 0, &tx_amic_mux0),
+ SND_SOC_DAPM_MUX("AMIC MUX1", SND_SOC_NOPM, 0, 0, &tx_amic_mux1),
+ SND_SOC_DAPM_MUX("AMIC MUX2", SND_SOC_NOPM, 0, 0, &tx_amic_mux2),
+ SND_SOC_DAPM_MUX("AMIC MUX3", SND_SOC_NOPM, 0, 0, &tx_amic_mux3),
+ SND_SOC_DAPM_MUX("AMIC MUX4", SND_SOC_NOPM, 0, 0, &tx_amic_mux4),
+ SND_SOC_DAPM_MUX("AMIC MUX5", SND_SOC_NOPM, 0, 0, &tx_amic_mux5),
+ SND_SOC_DAPM_MUX("AMIC MUX6", SND_SOC_NOPM, 0, 0, &tx_amic_mux6),
+ SND_SOC_DAPM_MUX("AMIC MUX7", SND_SOC_NOPM, 0, 0, &tx_amic_mux7),
+ SND_SOC_DAPM_MUX("AMIC MUX8", SND_SOC_NOPM, 0, 0, &tx_amic_mux8),
+ SND_SOC_DAPM_MUX_E("ADC MUX0", WCD934X_CDC_TX0_TX_PATH_CTL, 5, 0,
+ &tx_adc_mux0_mux, wcd934x_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("ADC MUX1", WCD934X_CDC_TX1_TX_PATH_CTL, 5, 0,
+ &tx_adc_mux1_mux, wcd934x_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("ADC MUX2", WCD934X_CDC_TX2_TX_PATH_CTL, 5, 0,
+ &tx_adc_mux2_mux, wcd934x_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("ADC MUX3", WCD934X_CDC_TX3_TX_PATH_CTL, 5, 0,
+ &tx_adc_mux3_mux, wcd934x_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("ADC MUX4", WCD934X_CDC_TX4_TX_PATH_CTL, 5, 0,
+ &tx_adc_mux4_mux, wcd934x_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("ADC MUX5", WCD934X_CDC_TX5_TX_PATH_CTL, 5, 0,
+ &tx_adc_mux5_mux, wcd934x_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("ADC MUX6", WCD934X_CDC_TX6_TX_PATH_CTL, 5, 0,
+ &tx_adc_mux6_mux, wcd934x_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("ADC MUX7", WCD934X_CDC_TX7_TX_PATH_CTL, 5, 0,
+ &tx_adc_mux7_mux, wcd934x_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("ADC MUX8", WCD934X_CDC_TX8_TX_PATH_CTL, 5, 0,
+ &tx_adc_mux8_mux, wcd934x_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("ADC1", NULL, WCD934X_ANA_AMIC1, 7, 0,
+ wcd934x_codec_enable_adc, SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_ADC_E("ADC2", NULL, WCD934X_ANA_AMIC2, 7, 0,
+ wcd934x_codec_enable_adc, SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_ADC_E("ADC3", NULL, WCD934X_ANA_AMIC3, 7, 0,
+ wcd934x_codec_enable_adc, SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_ADC_E("ADC4", NULL, WCD934X_ANA_AMIC4, 7, 0,
+ wcd934x_codec_enable_adc, SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_SUPPLY("MIC BIAS1", WCD934X_ANA_MICB1, 6, 0, NULL,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("MIC BIAS2", WCD934X_ANA_MICB2, 6, 0, NULL,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("MIC BIAS3", WCD934X_ANA_MICB3, 6, 0, NULL,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("MIC BIAS4", WCD934X_ANA_MICB4, 6, 0, NULL,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("AMIC4_5 SEL", SND_SOC_NOPM, 0, 0, &tx_amic4_5),
+ SND_SOC_DAPM_MUX("CDC_IF TX0 MUX", SND_SOC_NOPM, WCD934X_TX0, 0,
+ &cdc_if_tx0_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX1 MUX", SND_SOC_NOPM, WCD934X_TX1, 0,
+ &cdc_if_tx1_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX2 MUX", SND_SOC_NOPM, WCD934X_TX2, 0,
+ &cdc_if_tx2_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX3 MUX", SND_SOC_NOPM, WCD934X_TX3, 0,
+ &cdc_if_tx3_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX4 MUX", SND_SOC_NOPM, WCD934X_TX4, 0,
+ &cdc_if_tx4_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX5 MUX", SND_SOC_NOPM, WCD934X_TX5, 0,
+ &cdc_if_tx5_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX6 MUX", SND_SOC_NOPM, WCD934X_TX6, 0,
+ &cdc_if_tx6_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX7 MUX", SND_SOC_NOPM, WCD934X_TX7, 0,
+ &cdc_if_tx7_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX8 MUX", SND_SOC_NOPM, WCD934X_TX8, 0,
+ &cdc_if_tx8_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX9 MUX", SND_SOC_NOPM, WCD934X_TX9, 0,
+ &cdc_if_tx9_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX10 MUX", SND_SOC_NOPM, WCD934X_TX10, 0,
+ &cdc_if_tx10_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX11 MUX", SND_SOC_NOPM, WCD934X_TX11, 0,
+ &cdc_if_tx11_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX11 INP1 MUX", SND_SOC_NOPM, WCD934X_TX11, 0,
+ &cdc_if_tx11_inp1_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX13 MUX", SND_SOC_NOPM, WCD934X_TX13, 0,
+ &cdc_if_tx13_mux),
+ SND_SOC_DAPM_MUX("CDC_IF TX13 INP1 MUX", SND_SOC_NOPM, WCD934X_TX13, 0,
+ &cdc_if_tx13_inp1_mux),
+ SND_SOC_DAPM_MIXER("AIF1_CAP Mixer", SND_SOC_NOPM, AIF1_CAP, 0,
+ aif1_slim_cap_mixer,
+ ARRAY_SIZE(aif1_slim_cap_mixer)),
+ SND_SOC_DAPM_MIXER("AIF2_CAP Mixer", SND_SOC_NOPM, AIF2_CAP, 0,
+ aif2_slim_cap_mixer,
+ ARRAY_SIZE(aif2_slim_cap_mixer)),
+ SND_SOC_DAPM_MIXER("AIF3_CAP Mixer", SND_SOC_NOPM, AIF3_CAP, 0,
+ aif3_slim_cap_mixer,
+ ARRAY_SIZE(aif3_slim_cap_mixer)),
};
static const struct snd_soc_component_driver wcd934x_component_drv = {
--
2.20.1
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^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: wcd934x: add playback dapm widgets" to the asoc tree
@ 2019-12-25 18:21 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2019-12-25 18:21 UTC (permalink / raw)
To: Srinivas Kandagatla; +Cc: alsa-devel, Mark Brown
The patch
ASoC: wcd934x: add playback dapm widgets
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From dd9eb19b567303e4b92747dcfb5deedb182af111 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Thu, 19 Dec 2019 10:31:47 +0000
Subject: [PATCH] ASoC: wcd934x: add playback dapm widgets
This patch adds required dapm widgets for playback.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20191219103153.14875-6-srinivas.kandagatla@linaro.org
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wcd934x.c | 1550 ++++++++++++++++++++++++++++++++++++
1 file changed, 1550 insertions(+)
diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c
index 27a864f2ff77..556c051f7f99 100644
--- a/sound/soc/codecs/wcd934x.c
+++ b/sound/soc/codecs/wcd934x.c
@@ -412,6 +412,94 @@ static const char * const rx_hph_mode_mux_text[] = {
"Class-H Hi-Fi Low Power"
};
+static const char *const slim_rx_mux_text[] = {
+ "ZERO", "AIF1_PB", "AIF2_PB", "AIF3_PB", "AIF4_PB",
+};
+
+static const char * const rx_int0_7_mix_mux_text[] = {
+ "ZERO", "RX0", "RX1", "RX2", "RX3", "RX4", "RX5",
+ "RX6", "RX7", "PROXIMITY"
+};
+
+static const char * const rx_int_mix_mux_text[] = {
+ "ZERO", "RX0", "RX1", "RX2", "RX3", "RX4", "RX5",
+ "RX6", "RX7"
+};
+
+static const char * const rx_prim_mix_text[] = {
+ "ZERO", "DEC0", "DEC1", "IIR0", "IIR1", "RX0", "RX1", "RX2",
+ "RX3", "RX4", "RX5", "RX6", "RX7"
+};
+
+static const char * const rx_sidetone_mix_text[] = {
+ "ZERO", "SRC0", "SRC1", "SRC_SUM"
+};
+
+static const char * const iir_inp_mux_text[] = {
+ "ZERO", "DEC0", "DEC1", "DEC2", "DEC3", "DEC4", "DEC5", "DEC6",
+ "DEC7", "DEC8", "RX0", "RX1", "RX2", "RX3", "RX4", "RX5", "RX6", "RX7"
+};
+
+static const char * const rx_int_dem_inp_mux_text[] = {
+ "NORMAL_DSM_OUT", "CLSH_DSM_OUT",
+};
+
+static const char * const rx_int0_1_interp_mux_text[] = {
+ "ZERO", "RX INT0_1 MIX1",
+};
+
+static const char * const rx_int1_1_interp_mux_text[] = {
+ "ZERO", "RX INT1_1 MIX1",
+};
+
+static const char * const rx_int2_1_interp_mux_text[] = {
+ "ZERO", "RX INT2_1 MIX1",
+};
+
+static const char * const rx_int3_1_interp_mux_text[] = {
+ "ZERO", "RX INT3_1 MIX1",
+};
+
+static const char * const rx_int4_1_interp_mux_text[] = {
+ "ZERO", "RX INT4_1 MIX1",
+};
+
+static const char * const rx_int7_1_interp_mux_text[] = {
+ "ZERO", "RX INT7_1 MIX1",
+};
+
+static const char * const rx_int8_1_interp_mux_text[] = {
+ "ZERO", "RX INT8_1 MIX1",
+};
+
+static const char * const rx_int0_2_interp_mux_text[] = {
+ "ZERO", "RX INT0_2 MUX",
+};
+
+static const char * const rx_int1_2_interp_mux_text[] = {
+ "ZERO", "RX INT1_2 MUX",
+};
+
+static const char * const rx_int2_2_interp_mux_text[] = {
+ "ZERO", "RX INT2_2 MUX",
+};
+
+static const char * const rx_int3_2_interp_mux_text[] = {
+ "ZERO", "RX INT3_2 MUX",
+};
+
+static const char * const rx_int4_2_interp_mux_text[] = {
+ "ZERO", "RX INT4_2 MUX",
+};
+
+static const char * const rx_int7_2_interp_mux_text[] = {
+ "ZERO", "RX INT7_2 MUX",
+};
+
+static const char * const rx_int8_2_interp_mux_text[] = {
+ "ZERO", "RX INT8_2 MUX",
+};
+
static const struct soc_enum cf_dec0_enum =
SOC_ENUM_SINGLE(WCD934X_CDC_TX0_TX_PATH_CFG0, 5, 3, cf_text);
@@ -485,6 +573,236 @@ static const struct soc_enum rx_hph_mode_mux_enum =
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(rx_hph_mode_mux_text),
rx_hph_mode_mux_text);
+static const struct soc_enum slim_rx_mux_enum =
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(slim_rx_mux_text), slim_rx_mux_text);
+
+static const struct soc_enum rx_int0_2_mux_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT0_CFG1, 0, 10,
+ rx_int0_7_mix_mux_text);
+
+static const struct soc_enum rx_int1_2_mux_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT1_CFG1, 0, 9,
+ rx_int_mix_mux_text);
+
+static const struct soc_enum rx_int2_2_mux_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT2_CFG1, 0, 9,
+ rx_int_mix_mux_text);
+
+static const struct soc_enum rx_int3_2_mux_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT3_CFG1, 0, 9,
+ rx_int_mix_mux_text);
+
+static const struct soc_enum rx_int4_2_mux_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT4_CFG1, 0, 9,
+ rx_int_mix_mux_text);
+
+static const struct soc_enum rx_int7_2_mux_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT7_CFG1, 0, 10,
+ rx_int0_7_mix_mux_text);
+
+static const struct soc_enum rx_int8_2_mux_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT8_CFG1, 0, 9,
+ rx_int_mix_mux_text);
+
+static const struct soc_enum rx_int0_1_mix_inp0_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT0_CFG0, 0, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int0_1_mix_inp1_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT0_CFG0, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int0_1_mix_inp2_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT0_CFG1, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int1_1_mix_inp0_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT1_CFG0, 0, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int1_1_mix_inp1_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT1_CFG0, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int1_1_mix_inp2_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT1_CFG1, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int2_1_mix_inp0_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT2_CFG0, 0, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int2_1_mix_inp1_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT2_CFG0, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int2_1_mix_inp2_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT2_CFG1, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int3_1_mix_inp0_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT3_CFG0, 0, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int3_1_mix_inp1_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT3_CFG0, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int3_1_mix_inp2_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT3_CFG1, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int4_1_mix_inp0_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT4_CFG0, 0, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int4_1_mix_inp1_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT4_CFG0, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int4_1_mix_inp2_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT4_CFG1, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int7_1_mix_inp0_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT7_CFG0, 0, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int7_1_mix_inp1_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT7_CFG0, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int7_1_mix_inp2_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT7_CFG1, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int8_1_mix_inp0_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT8_CFG0, 0, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int8_1_mix_inp1_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT8_CFG0, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int8_1_mix_inp2_chain_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_RX_INT8_CFG1, 4, 13,
+ rx_prim_mix_text);
+
+static const struct soc_enum rx_int0_mix2_inp_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_SIDETONE_SRC_CFG0, 0, 4,
+ rx_sidetone_mix_text);
+
+static const struct soc_enum rx_int1_mix2_inp_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_SIDETONE_SRC_CFG0, 2, 4,
+ rx_sidetone_mix_text);
+
+static const struct soc_enum rx_int2_mix2_inp_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_SIDETONE_SRC_CFG0, 4, 4,
+ rx_sidetone_mix_text);
+
+static const struct soc_enum rx_int3_mix2_inp_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_SIDETONE_SRC_CFG0, 6, 4,
+ rx_sidetone_mix_text);
+
+static const struct soc_enum rx_int4_mix2_inp_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_SIDETONE_SRC_CFG1, 0, 4,
+ rx_sidetone_mix_text);
+
+static const struct soc_enum rx_int7_mix2_inp_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX_INP_MUX_SIDETONE_SRC_CFG1, 2, 4,
+ rx_sidetone_mix_text);
+
+static const struct soc_enum iir0_inp0_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR0_MIX_CFG0,
+ 0, 18, iir_inp_mux_text);
+
+static const struct soc_enum iir0_inp1_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR0_MIX_CFG1,
+ 0, 18, iir_inp_mux_text);
+
+static const struct soc_enum iir0_inp2_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR0_MIX_CFG2,
+ 0, 18, iir_inp_mux_text);
+
+static const struct soc_enum iir0_inp3_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR0_MIX_CFG3,
+ 0, 18, iir_inp_mux_text);
+
+static const struct soc_enum iir1_inp0_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR1_MIX_CFG0,
+ 0, 18, iir_inp_mux_text);
+
+static const struct soc_enum iir1_inp1_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR1_MIX_CFG1,
+ 0, 18, iir_inp_mux_text);
+
+static const struct soc_enum iir1_inp2_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR1_MIX_CFG2,
+ 0, 18, iir_inp_mux_text);
+
+static const struct soc_enum iir1_inp3_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_SIDETONE_IIR_INP_MUX_IIR1_MIX_CFG3,
+ 0, 18, iir_inp_mux_text);
+
+static const struct soc_enum rx_int0_dem_inp_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX0_RX_PATH_SEC0, 0,
+ ARRAY_SIZE(rx_int_dem_inp_mux_text),
+ rx_int_dem_inp_mux_text);
+
+static const struct soc_enum rx_int1_dem_inp_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX1_RX_PATH_SEC0, 0,
+ ARRAY_SIZE(rx_int_dem_inp_mux_text),
+ rx_int_dem_inp_mux_text);
+
+static const struct soc_enum rx_int2_dem_inp_mux_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX2_RX_PATH_SEC0, 0,
+ ARRAY_SIZE(rx_int_dem_inp_mux_text),
+ rx_int_dem_inp_mux_text);
+static const struct soc_enum rx_int0_1_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2,
+ rx_int0_1_interp_mux_text);
+
+static const struct soc_enum rx_int1_1_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2,
+ rx_int1_1_interp_mux_text);
+
+static const struct soc_enum rx_int2_1_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2,
+ rx_int2_1_interp_mux_text);
+
+static const struct soc_enum rx_int3_1_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int3_1_interp_mux_text);
+
+static const struct soc_enum rx_int4_1_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int4_1_interp_mux_text);
+
+static const struct soc_enum rx_int7_1_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int7_1_interp_mux_text);
+
+static const struct soc_enum rx_int8_1_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int8_1_interp_mux_text);
+
+static const struct soc_enum rx_int0_2_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int0_2_interp_mux_text);
+
+static const struct soc_enum rx_int1_2_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int1_2_interp_mux_text);
+
+static const struct soc_enum rx_int2_2_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int2_2_interp_mux_text);
+
+static const struct soc_enum rx_int3_2_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int3_2_interp_mux_text);
+
+static const struct soc_enum rx_int4_2_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int4_2_interp_mux_text);
+
+static const struct soc_enum rx_int7_2_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int7_2_interp_mux_text);
+
+static const struct soc_enum rx_int8_2_interp_mux_enum =
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, 2, rx_int8_2_interp_mux_text);
+
static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd,
int sido_src)
{
@@ -639,6 +957,22 @@ static int __wcd934x_cdc_mclk_enable(struct wcd934x_codec *wcd, bool enable)
return ret;
}
+static int wcd934x_codec_enable_mclk(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kc, int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ return __wcd934x_cdc_mclk_enable(wcd, true);
+ case SND_SOC_DAPM_POST_PMD:
+ return __wcd934x_cdc_mclk_enable(wcd, false);
+ }
+
+ return 0;
+}
+
static int wcd934x_get_version(struct wcd934x_codec *wcd)
{
int val1, val2, ver, ret;
@@ -1814,6 +2148,311 @@ static int wcd934x_rx_hph_mode_put(struct snd_kcontrol *kc,
return 0;
}
+static int slim_rx_mux_get(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc);
+ struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kc);
+ struct wcd934x_codec *wcd = dev_get_drvdata(dapm->dev);
+
+ ucontrol->value.enumerated.item[0] = wcd->rx_port_value[w->shift];
+
+ return 0;
+}
+
+static int slim_rx_mux_put(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kc);
+ struct wcd934x_codec *wcd = dev_get_drvdata(w->dapm->dev);
+ struct soc_enum *e = (struct soc_enum *)kc->private_value;
+ struct snd_soc_dapm_update *update = NULL;
+ u32 port_id = w->shift;
+
+ if (wcd->rx_port_value[port_id] == ucontrol->value.enumerated.item[0])
+ return 0;
+
+ wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0];
+
+ switch (wcd->rx_port_value[port_id]) {
+ case 0:
+ list_del_init(&wcd->rx_chs[port_id].list);
+ break;
+ case 1:
+ list_add_tail(&wcd->rx_chs[port_id].list,
+ &wcd->dai[AIF1_PB].slim_ch_list);
+ break;
+ case 2:
+ list_add_tail(&wcd->rx_chs[port_id].list,
+ &wcd->dai[AIF2_PB].slim_ch_list);
+ break;
+ case 3:
+ list_add_tail(&wcd->rx_chs[port_id].list,
+ &wcd->dai[AIF3_PB].slim_ch_list);
+ break;
+ case 4:
+ list_add_tail(&wcd->rx_chs[port_id].list,
+ &wcd->dai[AIF4_PB].slim_ch_list);
+ break;
+ default:
+ dev_err(wcd->dev, "Unknown AIF %d\n",
+ wcd->rx_port_value[port_id]);
+ goto err;
+ }
+
+ snd_soc_dapm_mux_update_power(w->dapm, kc, wcd->rx_port_value[port_id],
+ e, update);
+
+ return 0;
+err:
+ return -EINVAL;
+}
+
+static int wcd934x_int_dem_inp_mux_put(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_enum *e = (struct soc_enum *)kc->private_value;
+ struct snd_soc_component *component;
+ int reg, val, ret;
+
+ component = snd_soc_dapm_kcontrol_component(kc);
+ val = ucontrol->value.enumerated.item[0];
+ if (e->reg == WCD934X_CDC_RX0_RX_PATH_SEC0)
+ reg = WCD934X_CDC_RX0_RX_PATH_CFG0;
+ else if (e->reg == WCD934X_CDC_RX1_RX_PATH_SEC0)
+ reg = WCD934X_CDC_RX1_RX_PATH_CFG0;
+ else if (e->reg == WCD934X_CDC_RX2_RX_PATH_SEC0)
+ reg = WCD934X_CDC_RX2_RX_PATH_CFG0;
+ else
+ return -EINVAL;
+
+ /* Set Look Ahead Delay */
+ if (val)
+ snd_soc_component_update_bits(component, reg,
+ WCD934X_RX_DLY_ZN_EN_MASK,
+ WCD934X_RX_DLY_ZN_ENABLE);
+ else
+ snd_soc_component_update_bits(component, reg,
+ WCD934X_RX_DLY_ZN_EN_MASK,
+ WCD934X_RX_DLY_ZN_DISABLE);
+
+ ret = snd_soc_dapm_put_enum_double(kc, ucontrol);
+
+ return ret;
+}
+
+static const struct snd_kcontrol_new rx_int0_2_mux =
+ SOC_DAPM_ENUM("RX INT0_2 MUX Mux", rx_int0_2_mux_chain_enum);
+
+static const struct snd_kcontrol_new rx_int1_2_mux =
+ SOC_DAPM_ENUM("RX INT1_2 MUX Mux", rx_int1_2_mux_chain_enum);
+
+static const struct snd_kcontrol_new rx_int2_2_mux =
+ SOC_DAPM_ENUM("RX INT2_2 MUX Mux", rx_int2_2_mux_chain_enum);
+
+static const struct snd_kcontrol_new rx_int3_2_mux =
+ SOC_DAPM_ENUM("RX INT3_2 MUX Mux", rx_int3_2_mux_chain_enum);
+
+static const struct snd_kcontrol_new rx_int4_2_mux =
+ SOC_DAPM_ENUM("RX INT4_2 MUX Mux", rx_int4_2_mux_chain_enum);
+
+static const struct snd_kcontrol_new rx_int7_2_mux =
+ SOC_DAPM_ENUM("RX INT7_2 MUX Mux", rx_int7_2_mux_chain_enum);
+
+static const struct snd_kcontrol_new rx_int8_2_mux =
+ SOC_DAPM_ENUM("RX INT8_2 MUX Mux", rx_int8_2_mux_chain_enum);
+
+static const struct snd_kcontrol_new rx_int0_1_mix_inp0_mux =
+ SOC_DAPM_ENUM("RX INT0_1 MIX1 INP0 Mux", rx_int0_1_mix_inp0_chain_enum);
+
+static const struct snd_kcontrol_new rx_int0_1_mix_inp1_mux =
+ SOC_DAPM_ENUM("RX INT0_1 MIX1 INP1 Mux", rx_int0_1_mix_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx_int0_1_mix_inp2_mux =
+ SOC_DAPM_ENUM("RX INT0_1 MIX1 INP2 Mux", rx_int0_1_mix_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx_int1_1_mix_inp0_mux =
+ SOC_DAPM_ENUM("RX INT1_1 MIX1 INP0 Mux", rx_int1_1_mix_inp0_chain_enum);
+
+static const struct snd_kcontrol_new rx_int1_1_mix_inp1_mux =
+ SOC_DAPM_ENUM("RX INT1_1 MIX1 INP1 Mux", rx_int1_1_mix_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx_int1_1_mix_inp2_mux =
+ SOC_DAPM_ENUM("RX INT1_1 MIX1 INP2 Mux", rx_int1_1_mix_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx_int2_1_mix_inp0_mux =
+ SOC_DAPM_ENUM("RX INT2_1 MIX1 INP0 Mux", rx_int2_1_mix_inp0_chain_enum);
+
+static const struct snd_kcontrol_new rx_int2_1_mix_inp1_mux =
+ SOC_DAPM_ENUM("RX INT2_1 MIX1 INP1 Mux", rx_int2_1_mix_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx_int2_1_mix_inp2_mux =
+ SOC_DAPM_ENUM("RX INT2_1 MIX1 INP2 Mux", rx_int2_1_mix_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx_int3_1_mix_inp0_mux =
+ SOC_DAPM_ENUM("RX INT3_1 MIX1 INP0 Mux", rx_int3_1_mix_inp0_chain_enum);
+
+static const struct snd_kcontrol_new rx_int3_1_mix_inp1_mux =
+ SOC_DAPM_ENUM("RX INT3_1 MIX1 INP1 Mux", rx_int3_1_mix_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx_int3_1_mix_inp2_mux =
+ SOC_DAPM_ENUM("RX INT3_1 MIX1 INP2 Mux", rx_int3_1_mix_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx_int4_1_mix_inp0_mux =
+ SOC_DAPM_ENUM("RX INT4_1 MIX1 INP0 Mux", rx_int4_1_mix_inp0_chain_enum);
+
+static const struct snd_kcontrol_new rx_int4_1_mix_inp1_mux =
+ SOC_DAPM_ENUM("RX INT4_1 MIX1 INP1 Mux", rx_int4_1_mix_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx_int4_1_mix_inp2_mux =
+ SOC_DAPM_ENUM("RX INT4_1 MIX1 INP2 Mux", rx_int4_1_mix_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx_int7_1_mix_inp0_mux =
+ SOC_DAPM_ENUM("RX INT7_1 MIX1 INP0 Mux", rx_int7_1_mix_inp0_chain_enum);
+
+static const struct snd_kcontrol_new rx_int7_1_mix_inp1_mux =
+ SOC_DAPM_ENUM("RX INT7_1 MIX1 INP1 Mux", rx_int7_1_mix_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx_int7_1_mix_inp2_mux =
+ SOC_DAPM_ENUM("RX INT7_1 MIX1 INP2 Mux", rx_int7_1_mix_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx_int8_1_mix_inp0_mux =
+ SOC_DAPM_ENUM("RX INT8_1 MIX1 INP0 Mux", rx_int8_1_mix_inp0_chain_enum);
+
+static const struct snd_kcontrol_new rx_int8_1_mix_inp1_mux =
+ SOC_DAPM_ENUM("RX INT8_1 MIX1 INP1 Mux", rx_int8_1_mix_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx_int8_1_mix_inp2_mux =
+ SOC_DAPM_ENUM("RX INT8_1 MIX1 INP2 Mux", rx_int8_1_mix_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx_int0_mix2_inp_mux =
+ SOC_DAPM_ENUM("RX INT0 MIX2 INP Mux", rx_int0_mix2_inp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int1_mix2_inp_mux =
+ SOC_DAPM_ENUM("RX INT1 MIX2 INP Mux", rx_int1_mix2_inp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int2_mix2_inp_mux =
+ SOC_DAPM_ENUM("RX INT2 MIX2 INP Mux", rx_int2_mix2_inp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int3_mix2_inp_mux =
+ SOC_DAPM_ENUM("RX INT3 MIX2 INP Mux", rx_int3_mix2_inp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int4_mix2_inp_mux =
+ SOC_DAPM_ENUM("RX INT4 MIX2 INP Mux", rx_int4_mix2_inp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int7_mix2_inp_mux =
+ SOC_DAPM_ENUM("RX INT7 MIX2 INP Mux", rx_int7_mix2_inp_mux_enum);
+
+static const struct snd_kcontrol_new iir0_inp0_mux =
+ SOC_DAPM_ENUM("IIR0 INP0 Mux", iir0_inp0_mux_enum);
+static const struct snd_kcontrol_new iir0_inp1_mux =
+ SOC_DAPM_ENUM("IIR0 INP1 Mux", iir0_inp1_mux_enum);
+static const struct snd_kcontrol_new iir0_inp2_mux =
+ SOC_DAPM_ENUM("IIR0 INP2 Mux", iir0_inp2_mux_enum);
+static const struct snd_kcontrol_new iir0_inp3_mux =
+ SOC_DAPM_ENUM("IIR0 INP3 Mux", iir0_inp3_mux_enum);
+
+static const struct snd_kcontrol_new iir1_inp0_mux =
+ SOC_DAPM_ENUM("IIR1 INP0 Mux", iir1_inp0_mux_enum);
+static const struct snd_kcontrol_new iir1_inp1_mux =
+ SOC_DAPM_ENUM("IIR1 INP1 Mux", iir1_inp1_mux_enum);
+static const struct snd_kcontrol_new iir1_inp2_mux =
+ SOC_DAPM_ENUM("IIR1 INP2 Mux", iir1_inp2_mux_enum);
+static const struct snd_kcontrol_new iir1_inp3_mux =
+ SOC_DAPM_ENUM("IIR1 INP3 Mux", iir1_inp3_mux_enum);
+
+static const struct snd_kcontrol_new slim_rx_mux[WCD934X_RX_MAX] = {
+ SOC_DAPM_ENUM_EXT("SLIM RX0 Mux", slim_rx_mux_enum,
+ slim_rx_mux_get, slim_rx_mux_put),
+ SOC_DAPM_ENUM_EXT("SLIM RX1 Mux", slim_rx_mux_enum,
+ slim_rx_mux_get, slim_rx_mux_put),
+ SOC_DAPM_ENUM_EXT("SLIM RX2 Mux", slim_rx_mux_enum,
+ slim_rx_mux_get, slim_rx_mux_put),
+ SOC_DAPM_ENUM_EXT("SLIM RX3 Mux", slim_rx_mux_enum,
+ slim_rx_mux_get, slim_rx_mux_put),
+ SOC_DAPM_ENUM_EXT("SLIM RX4 Mux", slim_rx_mux_enum,
+ slim_rx_mux_get, slim_rx_mux_put),
+ SOC_DAPM_ENUM_EXT("SLIM RX5 Mux", slim_rx_mux_enum,
+ slim_rx_mux_get, slim_rx_mux_put),
+ SOC_DAPM_ENUM_EXT("SLIM RX6 Mux", slim_rx_mux_enum,
+ slim_rx_mux_get, slim_rx_mux_put),
+ SOC_DAPM_ENUM_EXT("SLIM RX7 Mux", slim_rx_mux_enum,
+ slim_rx_mux_get, slim_rx_mux_put),
+};
+
+static const struct snd_kcontrol_new rx_int1_asrc_switch[] = {
+ SOC_DAPM_SINGLE("HPHL Switch", SND_SOC_NOPM, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new rx_int2_asrc_switch[] = {
+ SOC_DAPM_SINGLE("HPHR Switch", SND_SOC_NOPM, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new rx_int3_asrc_switch[] = {
+ SOC_DAPM_SINGLE("LO1 Switch", SND_SOC_NOPM, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new rx_int4_asrc_switch[] = {
+ SOC_DAPM_SINGLE("LO2 Switch", SND_SOC_NOPM, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new rx_int0_dem_inp_mux =
+ SOC_DAPM_ENUM_EXT("RX INT0 DEM MUX Mux", rx_int0_dem_inp_mux_enum,
+ snd_soc_dapm_get_enum_double,
+ wcd934x_int_dem_inp_mux_put);
+
+static const struct snd_kcontrol_new rx_int1_dem_inp_mux =
+ SOC_DAPM_ENUM_EXT("RX INT1 DEM MUX Mux", rx_int1_dem_inp_mux_enum,
+ snd_soc_dapm_get_enum_double,
+ wcd934x_int_dem_inp_mux_put);
+
+static const struct snd_kcontrol_new rx_int2_dem_inp_mux =
+ SOC_DAPM_ENUM_EXT("RX INT2 DEM MUX Mux", rx_int2_dem_inp_mux_enum,
+ snd_soc_dapm_get_enum_double,
+ wcd934x_int_dem_inp_mux_put);
+
+static const struct snd_kcontrol_new rx_int0_1_interp_mux =
+ SOC_DAPM_ENUM("RX INT0_1 INTERP Mux", rx_int0_1_interp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int1_1_interp_mux =
+ SOC_DAPM_ENUM("RX INT1_1 INTERP Mux", rx_int1_1_interp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int2_1_interp_mux =
+ SOC_DAPM_ENUM("RX INT2_1 INTERP Mux", rx_int2_1_interp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int3_1_interp_mux =
+ SOC_DAPM_ENUM("RX INT3_1 INTERP Mux", rx_int3_1_interp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int4_1_interp_mux =
+ SOC_DAPM_ENUM("RX INT4_1 INTERP Mux", rx_int4_1_interp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int7_1_interp_mux =
+ SOC_DAPM_ENUM("RX INT7_1 INTERP Mux", rx_int7_1_interp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int8_1_interp_mux =
+ SOC_DAPM_ENUM("RX INT8_1 INTERP Mux", rx_int8_1_interp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int0_2_interp_mux =
+ SOC_DAPM_ENUM("RX INT0_2 INTERP Mux", rx_int0_2_interp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int1_2_interp_mux =
+ SOC_DAPM_ENUM("RX INT1_2 INTERP Mux", rx_int1_2_interp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int2_2_interp_mux =
+ SOC_DAPM_ENUM("RX INT2_2 INTERP Mux", rx_int2_2_interp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int3_2_interp_mux =
+ SOC_DAPM_ENUM("RX INT3_2 INTERP Mux", rx_int3_2_interp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int4_2_interp_mux =
+ SOC_DAPM_ENUM("RX INT4_2 INTERP Mux", rx_int4_2_interp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int7_2_interp_mux =
+ SOC_DAPM_ENUM("RX INT7_2 INTERP Mux", rx_int7_2_interp_mux_enum);
+
+static const struct snd_kcontrol_new rx_int8_2_interp_mux =
+ SOC_DAPM_ENUM("RX INT8_2 INTERP Mux", rx_int8_2_interp_mux_enum);
+
static const struct snd_kcontrol_new wcd934x_snd_controls[] = {
/* Gain Controls */
SOC_SINGLE_TLV("EAR PA Volume", WCD934X_ANA_EAR, 4, 4, 1, ear_pa_gain),
@@ -1983,12 +2622,923 @@ static const struct snd_kcontrol_new wcd934x_snd_controls[] = {
wcd934x_compander_get, wcd934x_compander_set),
};
+static void wcd934x_codec_enable_int_port(struct wcd_slim_codec_dai_data *dai,
+ struct snd_soc_component *component)
+{
+ int port_num = 0;
+ unsigned short reg = 0;
+ unsigned int val = 0;
+ struct wcd934x_codec *wcd = dev_get_drvdata(component->dev);
+ struct wcd934x_slim_ch *ch;
+
+ list_for_each_entry(ch, &dai->slim_ch_list, list) {
+ if (ch->port >= WCD934X_RX_START) {
+ port_num = ch->port - WCD934X_RX_START;
+ reg = WCD934X_SLIM_PGD_PORT_INT_EN0 + (port_num / 8);
+ } else {
+ port_num = ch->port;
+ reg = WCD934X_SLIM_PGD_PORT_INT_TX_EN0 + (port_num / 8);
+ }
+
+ regmap_read(wcd->if_regmap, reg, &val);
+ if (!(val & BIT(port_num % 8)))
+ regmap_write(wcd->if_regmap, reg,
+ val | BIT(port_num % 8));
+ }
+}
+
+static int wcd934x_codec_enable_slim(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kc, int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ struct wcd934x_codec *wcd = snd_soc_component_get_drvdata(comp);
+ struct wcd_slim_codec_dai_data *dai = &wcd->dai[w->shift];
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ wcd934x_codec_enable_int_port(dai, comp);
+ break;
+ }
+
+ return 0;
+}
+
+static void wcd934x_codec_hd2_control(struct snd_soc_component *component,
+ u16 interp_idx, int event)
+{
+ u16 hd2_scale_reg;
+ u16 hd2_enable_reg = 0;
+
+ switch (interp_idx) {
+ case INTERP_HPHL:
+ hd2_scale_reg = WCD934X_CDC_RX1_RX_PATH_SEC3;
+ hd2_enable_reg = WCD934X_CDC_RX1_RX_PATH_CFG0;
+ break;
+ case INTERP_HPHR:
+ hd2_scale_reg = WCD934X_CDC_RX2_RX_PATH_SEC3;
+ hd2_enable_reg = WCD934X_CDC_RX2_RX_PATH_CFG0;
+ break;
+ default:
+ return;
+ }
+
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ snd_soc_component_update_bits(component, hd2_scale_reg,
+ WCD934X_CDC_RX_PATH_SEC_HD2_ALPHA_MASK,
+ WCD934X_CDC_RX_PATH_SEC_HD2_ALPHA_0P3125);
+ snd_soc_component_update_bits(component, hd2_enable_reg,
+ WCD934X_CDC_RX_PATH_CFG_HD2_EN_MASK,
+ WCD934X_CDC_RX_PATH_CFG_HD2_ENABLE);
+ }
+
+ if (SND_SOC_DAPM_EVENT_OFF(event)) {
+ snd_soc_component_update_bits(component, hd2_enable_reg,
+ WCD934X_CDC_RX_PATH_CFG_HD2_EN_MASK,
+ WCD934X_CDC_RX_PATH_CFG_HD2_DISABLE);
+ snd_soc_component_update_bits(component, hd2_scale_reg,
+ WCD934X_CDC_RX_PATH_SEC_HD2_ALPHA_MASK,
+ WCD934X_CDC_RX_PATH_SEC_HD2_ALPHA_0P0000);
+ }
+}
+
+static void wcd934x_codec_hphdelay_lutbypass(struct snd_soc_component *comp,
+ u16 interp_idx, int event)
+{
+ u8 hph_dly_mask;
+ u16 hph_lut_bypass_reg = 0;
+ u16 hph_comp_ctrl7 = 0;
+
+ switch (interp_idx) {
+ case INTERP_HPHL:
+ hph_dly_mask = 1;
+ hph_lut_bypass_reg = WCD934X_CDC_TOP_HPHL_COMP_LUT;
+ hph_comp_ctrl7 = WCD934X_CDC_COMPANDER1_CTL7;
+ break;
+ case INTERP_HPHR:
+ hph_dly_mask = 2;
+ hph_lut_bypass_reg = WCD934X_CDC_TOP_HPHR_COMP_LUT;
+ hph_comp_ctrl7 = WCD934X_CDC_COMPANDER2_CTL7;
+ break;
+ default:
+ return;
+ }
+
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ snd_soc_component_update_bits(comp, WCD934X_CDC_CLSH_TEST0,
+ hph_dly_mask, 0x0);
+ snd_soc_component_update_bits(comp, hph_lut_bypass_reg,
+ WCD934X_HPH_LUT_BYPASS_MASK,
+ WCD934X_HPH_LUT_BYPASS_ENABLE);
+ }
+
+ if (SND_SOC_DAPM_EVENT_OFF(event)) {
+ snd_soc_component_update_bits(comp, WCD934X_CDC_CLSH_TEST0,
+ hph_dly_mask, hph_dly_mask);
+ snd_soc_component_update_bits(comp, hph_lut_bypass_reg,
+ WCD934X_HPH_LUT_BYPASS_MASK,
+ WCD934X_HPH_LUT_BYPASS_DISABLE);
+ }
+}
+
+static int wcd934x_config_compander(struct snd_soc_component *comp,
+ int interp_n, int event)
+{
+ struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev);
+ int compander;
+ u16 comp_ctl0_reg, rx_path_cfg0_reg;
+
+ /* EAR does not have compander */
+ if (!interp_n)
+ return 0;
+
+ compander = interp_n - 1;
+ if (!wcd->comp_enabled[compander])
+ return 0;
+
+ comp_ctl0_reg = WCD934X_CDC_COMPANDER1_CTL0 + (compander * 8);
+ rx_path_cfg0_reg = WCD934X_CDC_RX1_RX_PATH_CFG0 + (compander * 20);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ /* Enable Compander Clock */
+ snd_soc_component_update_bits(comp, comp_ctl0_reg,
+ WCD934X_COMP_CLK_EN_MASK,
+ WCD934X_COMP_CLK_ENABLE);
+ snd_soc_component_update_bits(comp, comp_ctl0_reg,
+ WCD934X_COMP_SOFT_RST_MASK,
+ WCD934X_COMP_SOFT_RST_ENABLE);
+ snd_soc_component_update_bits(comp, comp_ctl0_reg,
+ WCD934X_COMP_SOFT_RST_MASK,
+ WCD934X_COMP_SOFT_RST_DISABLE);
+ snd_soc_component_update_bits(comp, rx_path_cfg0_reg,
+ WCD934X_HPH_CMP_EN_MASK,
+ WCD934X_HPH_CMP_ENABLE);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_component_update_bits(comp, rx_path_cfg0_reg,
+ WCD934X_HPH_CMP_EN_MASK,
+ WCD934X_HPH_CMP_DISABLE);
+ snd_soc_component_update_bits(comp, comp_ctl0_reg,
+ WCD934X_COMP_HALT_MASK,
+ WCD934X_COMP_HALT);
+ snd_soc_component_update_bits(comp, comp_ctl0_reg,
+ WCD934X_COMP_SOFT_RST_MASK,
+ WCD934X_COMP_SOFT_RST_ENABLE);
+ snd_soc_component_update_bits(comp, comp_ctl0_reg,
+ WCD934X_COMP_SOFT_RST_MASK,
+ WCD934X_COMP_SOFT_RST_DISABLE);
+ snd_soc_component_update_bits(comp, comp_ctl0_reg,
+ WCD934X_COMP_CLK_EN_MASK, 0x0);
+ snd_soc_component_update_bits(comp, comp_ctl0_reg,
+ WCD934X_COMP_SOFT_RST_MASK, 0x0);
+ break;
+ }
+
+ return 0;
+}
+
+static int wcd934x_codec_enable_interp_clk(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kc, int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ int interp_idx = w->shift;
+ u16 main_reg = WCD934X_CDC_RX0_RX_PATH_CTL + (interp_idx * 20);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ /* Clk enable */
+ snd_soc_component_update_bits(comp, main_reg,
+ WCD934X_RX_CLK_EN_MASK,
+ WCD934X_RX_CLK_ENABLE);
+ wcd934x_codec_hd2_control(comp, interp_idx, event);
+ wcd934x_codec_hphdelay_lutbypass(comp, interp_idx, event);
+ wcd934x_config_compander(comp, interp_idx, event);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ wcd934x_config_compander(comp, interp_idx, event);
+ wcd934x_codec_hphdelay_lutbypass(comp, interp_idx, event);
+ wcd934x_codec_hd2_control(comp, interp_idx, event);
+ /* Clk Disable */
+ snd_soc_component_update_bits(comp, main_reg,
+ WCD934X_RX_CLK_EN_MASK, 0);
+ /* Reset enable and disable */
+ snd_soc_component_update_bits(comp, main_reg,
+ WCD934X_RX_RESET_MASK,
+ WCD934X_RX_RESET_ENABLE);
+ snd_soc_component_update_bits(comp, main_reg,
+ WCD934X_RX_RESET_MASK,
+ WCD934X_RX_RESET_DISABLE);
+ /* Reset rate to 48K*/
+ snd_soc_component_update_bits(comp, main_reg,
+ WCD934X_RX_PCM_RATE_MASK,
+ WCD934X_RX_PCM_RATE_F_48K);
+ break;
+ }
+
+ return 0;
+}
+
+static int wcd934x_codec_enable_mix_path(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kc, int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ int offset_val = 0;
+ u16 gain_reg, mix_reg;
+ int val = 0;
+
+ gain_reg = WCD934X_CDC_RX0_RX_VOL_MIX_CTL +
+ (w->shift * WCD934X_RX_PATH_CTL_OFFSET);
+ mix_reg = WCD934X_CDC_RX0_RX_PATH_MIX_CTL +
+ (w->shift * WCD934X_RX_PATH_CTL_OFFSET);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ /* Clk enable */
+ snd_soc_component_update_bits(comp, mix_reg,
+ WCD934X_CDC_RX_MIX_CLK_EN_MASK,
+ WCD934X_CDC_RX_MIX_CLK_ENABLE);
+ break;
+
+ case SND_SOC_DAPM_POST_PMU:
+ val = snd_soc_component_read32(comp, gain_reg);
+ val += offset_val;
+ snd_soc_component_write(comp, gain_reg, val);
+ break;
+ };
+
+ return 0;
+}
+
+static int wcd934x_codec_set_iir_gain(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ int reg = w->reg;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ /* B1 GAIN */
+ snd_soc_component_write(comp, reg,
+ snd_soc_component_read32(comp, reg));
+ /* B2 GAIN */
+ reg++;
+ snd_soc_component_write(comp, reg,
+ snd_soc_component_read32(comp, reg));
+ /* B3 GAIN */
+ reg++;
+ snd_soc_component_write(comp, reg,
+ snd_soc_component_read32(comp, reg));
+ /* B4 GAIN */
+ reg++;
+ snd_soc_component_write(comp, reg,
+ snd_soc_component_read32(comp, reg));
+ /* B5 GAIN */
+ reg++;
+ snd_soc_component_write(comp, reg,
+ snd_soc_component_read32(comp, reg));
+ break;
+ default:
+ break;
+ }
+ return 0;
+}
+
+static int wcd934x_codec_enable_main_path(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ u16 gain_reg;
+
+ gain_reg = WCD934X_CDC_RX0_RX_VOL_CTL + (w->shift *
+ WCD934X_RX_PATH_CTL_OFFSET);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_component_write(comp, gain_reg,
+ snd_soc_component_read32(comp, gain_reg));
+ break;
+ };
+
+ return 0;
+}
+
+static int wcd934x_codec_ear_dac_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kc, int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ /* Disable AutoChop timer during power up */
+ snd_soc_component_update_bits(comp,
+ WCD934X_HPH_NEW_INT_HPH_TIMER1,
+ WCD934X_HPH_AUTOCHOP_TIMER_EN_MASK, 0x0);
+ wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_PRE_DAC,
+ WCD_CLSH_STATE_EAR, CLS_H_NORMAL);
+
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_POST_PA,
+ WCD_CLSH_STATE_EAR, CLS_H_NORMAL);
+ break;
+ };
+
+ return 0;
+}
+
+static int wcd934x_codec_hphl_dac_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev);
+ int hph_mode = wcd->hph_mode;
+ u8 dem_inp;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ /* Read DEM INP Select */
+ dem_inp = snd_soc_component_read32(comp,
+ WCD934X_CDC_RX1_RX_PATH_SEC0) & 0x03;
+
+ if (((hph_mode == CLS_H_HIFI) || (hph_mode == CLS_H_LOHIFI) ||
+ (hph_mode == CLS_H_LP)) && (dem_inp != 0x01)) {
+ return -EINVAL;
+ }
+ if (hph_mode != CLS_H_LP)
+ /* Ripple freq control enable */
+ snd_soc_component_update_bits(comp,
+ WCD934X_SIDO_NEW_VOUT_D_FREQ2,
+ WCD934X_SIDO_RIPPLE_FREQ_EN_MASK,
+ WCD934X_SIDO_RIPPLE_FREQ_ENABLE);
+ /* Disable AutoChop timer during power up */
+ snd_soc_component_update_bits(comp,
+ WCD934X_HPH_NEW_INT_HPH_TIMER1,
+ WCD934X_HPH_AUTOCHOP_TIMER_EN_MASK, 0x0);
+ wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_PRE_DAC,
+ WCD_CLSH_STATE_HPHL, hph_mode);
+
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ /* 1000us required as per HW requirement */
+ usleep_range(1000, 1100);
+ wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_POST_PA,
+ WCD_CLSH_STATE_HPHL, hph_mode);
+ if (hph_mode != CLS_H_LP)
+ /* Ripple freq control disable */
+ snd_soc_component_update_bits(comp,
+ WCD934X_SIDO_NEW_VOUT_D_FREQ2,
+ WCD934X_SIDO_RIPPLE_FREQ_EN_MASK, 0x0);
+
+ break;
+ default:
+ break;
+ };
+
+ return 0;
+}
+
+static int wcd934x_codec_hphr_dac_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev);
+ int hph_mode = wcd->hph_mode;
+ u8 dem_inp;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ dem_inp = snd_soc_component_read32(comp,
+ WCD934X_CDC_RX2_RX_PATH_SEC0) & 0x03;
+ if (((hph_mode == CLS_H_HIFI) || (hph_mode == CLS_H_LOHIFI) ||
+ (hph_mode == CLS_H_LP)) && (dem_inp != 0x01)) {
+ return -EINVAL;
+ }
+ if (hph_mode != CLS_H_LP)
+ /* Ripple freq control enable */
+ snd_soc_component_update_bits(comp,
+ WCD934X_SIDO_NEW_VOUT_D_FREQ2,
+ WCD934X_SIDO_RIPPLE_FREQ_EN_MASK,
+ WCD934X_SIDO_RIPPLE_FREQ_ENABLE);
+ /* Disable AutoChop timer during power up */
+ snd_soc_component_update_bits(comp,
+ WCD934X_HPH_NEW_INT_HPH_TIMER1,
+ WCD934X_HPH_AUTOCHOP_TIMER_EN_MASK, 0x0);
+ wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_PRE_DAC,
+ WCD_CLSH_STATE_HPHR,
+ hph_mode);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ /* 1000us required as per HW requirement */
+ usleep_range(1000, 1100);
+
+ wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_POST_PA,
+ WCD_CLSH_STATE_HPHR, hph_mode);
+ if (hph_mode != CLS_H_LP)
+ /* Ripple freq control disable */
+ snd_soc_component_update_bits(comp,
+ WCD934X_SIDO_NEW_VOUT_D_FREQ2,
+ WCD934X_SIDO_RIPPLE_FREQ_EN_MASK, 0x0);
+ break;
+ default:
+ break;
+ };
+
+ return 0;
+}
+
+static int wcd934x_codec_lineout_dac_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kc, int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ struct wcd934x_codec *wcd = dev_get_drvdata(comp->dev);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_PRE_DAC,
+ WCD_CLSH_STATE_LO, CLS_AB);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_POST_PA,
+ WCD_CLSH_STATE_LO, CLS_AB);
+ break;
+ }
+
+ return 0;
+}
+
+static int wcd934x_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ /*
+ * 7ms sleep is required after PA is enabled as per
+ * HW requirement. If compander is disabled, then
+ * 20ms delay is needed.
+ */
+ usleep_range(20000, 20100);
+
+ snd_soc_component_update_bits(comp, WCD934X_HPH_L_TEST,
+ WCD934X_HPH_OCP_DET_MASK,
+ WCD934X_HPH_OCP_DET_ENABLE);
+ /* Remove Mute on primary path */
+ snd_soc_component_update_bits(comp, WCD934X_CDC_RX1_RX_PATH_CTL,
+ WCD934X_RX_PATH_PGA_MUTE_EN_MASK,
+ 0);
+ /* Enable GM3 boost */
+ snd_soc_component_update_bits(comp, WCD934X_HPH_CNP_WG_CTL,
+ WCD934X_HPH_GM3_BOOST_EN_MASK,
+ WCD934X_HPH_GM3_BOOST_ENABLE);
+ /* Enable AutoChop timer at the end of power up */
+ snd_soc_component_update_bits(comp,
+ WCD934X_HPH_NEW_INT_HPH_TIMER1,
+ WCD934X_HPH_AUTOCHOP_TIMER_EN_MASK,
+ WCD934X_HPH_AUTOCHOP_TIMER_ENABLE);
+ /* Remove mix path mute */
+ snd_soc_component_update_bits(comp,
+ WCD934X_CDC_RX1_RX_PATH_MIX_CTL,
+ WCD934X_CDC_RX_PGA_MUTE_EN_MASK, 0x00);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ /* Enable DSD Mute before PA disable */
+ snd_soc_component_update_bits(comp, WCD934X_HPH_L_TEST,
+ WCD934X_HPH_OCP_DET_MASK,
+ WCD934X_HPH_OCP_DET_DISABLE);
+ snd_soc_component_update_bits(comp, WCD934X_CDC_RX1_RX_PATH_CTL,
+ WCD934X_RX_PATH_PGA_MUTE_EN_MASK,
+ WCD934X_RX_PATH_PGA_MUTE_ENABLE);
+ snd_soc_component_update_bits(comp,
+ WCD934X_CDC_RX1_RX_PATH_MIX_CTL,
+ WCD934X_RX_PATH_PGA_MUTE_EN_MASK,
+ WCD934X_RX_PATH_PGA_MUTE_ENABLE);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ /*
+ * 5ms sleep is required after PA disable. If compander is
+ * disabled, then 20ms delay is needed after PA disable.
+ */
+ usleep_range(20000, 20100);
+ break;
+ };
+
+ return 0;
+}
+
+static int wcd934x_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ /*
+ * 7ms sleep is required after PA is enabled as per
+ * HW requirement. If compander is disabled, then
+ * 20ms delay is needed.
+ */
+ usleep_range(20000, 20100);
+ snd_soc_component_update_bits(comp, WCD934X_HPH_R_TEST,
+ WCD934X_HPH_OCP_DET_MASK,
+ WCD934X_HPH_OCP_DET_ENABLE);
+ /* Remove mute */
+ snd_soc_component_update_bits(comp, WCD934X_CDC_RX2_RX_PATH_CTL,
+ WCD934X_RX_PATH_PGA_MUTE_EN_MASK,
+ 0);
+ /* Enable GM3 boost */
+ snd_soc_component_update_bits(comp, WCD934X_HPH_CNP_WG_CTL,
+ WCD934X_HPH_GM3_BOOST_EN_MASK,
+ WCD934X_HPH_GM3_BOOST_ENABLE);
+ /* Enable AutoChop timer at the end of power up */
+ snd_soc_component_update_bits(comp,
+ WCD934X_HPH_NEW_INT_HPH_TIMER1,
+ WCD934X_HPH_AUTOCHOP_TIMER_EN_MASK,
+ WCD934X_HPH_AUTOCHOP_TIMER_ENABLE);
+ /* Remove mix path mute if it is enabled */
+ if ((snd_soc_component_read32(comp,
+ WCD934X_CDC_RX2_RX_PATH_MIX_CTL)) & 0x10)
+ snd_soc_component_update_bits(comp,
+ WCD934X_CDC_RX2_RX_PATH_MIX_CTL,
+ WCD934X_CDC_RX_PGA_MUTE_EN_MASK,
+ WCD934X_CDC_RX_PGA_MUTE_DISABLE);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_component_update_bits(comp, WCD934X_HPH_R_TEST,
+ WCD934X_HPH_OCP_DET_MASK,
+ WCD934X_HPH_OCP_DET_DISABLE);
+ snd_soc_component_update_bits(comp, WCD934X_CDC_RX2_RX_PATH_CTL,
+ WCD934X_RX_PATH_PGA_MUTE_EN_MASK,
+ WCD934X_RX_PATH_PGA_MUTE_ENABLE);
+ snd_soc_component_update_bits(comp,
+ WCD934X_CDC_RX2_RX_PATH_MIX_CTL,
+ WCD934X_CDC_RX_PGA_MUTE_EN_MASK,
+ WCD934X_CDC_RX_PGA_MUTE_ENABLE);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ /*
+ * 5ms sleep is required after PA disable. If compander is
+ * disabled, then 20ms delay is needed after PA disable.
+ */
+ usleep_range(20000, 20100);
+ break;
+ };
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget wcd934x_dapm_widgets[] = {
+ /* Analog Outputs */
+ SND_SOC_DAPM_OUTPUT("EAR"),
+ SND_SOC_DAPM_OUTPUT("HPHL"),
+ SND_SOC_DAPM_OUTPUT("HPHR"),
+ SND_SOC_DAPM_OUTPUT("LINEOUT1"),
+ SND_SOC_DAPM_OUTPUT("LINEOUT2"),
+ SND_SOC_DAPM_OUTPUT("SPK1 OUT"),
+ SND_SOC_DAPM_OUTPUT("SPK2 OUT"),
+ SND_SOC_DAPM_OUTPUT("ANC EAR"),
+ SND_SOC_DAPM_OUTPUT("ANC HPHL"),
+ SND_SOC_DAPM_OUTPUT("ANC HPHR"),
+ SND_SOC_DAPM_OUTPUT("WDMA3_OUT"),
+ SND_SOC_DAPM_OUTPUT("MAD_CPE_OUT1"),
+ SND_SOC_DAPM_OUTPUT("MAD_CPE_OUT2"),
+ SND_SOC_DAPM_AIF_IN_E("AIF1 PB", "AIF1 Playback", 0, SND_SOC_NOPM,
+ AIF1_PB, 0, wcd934x_codec_enable_slim,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_IN_E("AIF2 PB", "AIF2 Playback", 0, SND_SOC_NOPM,
+ AIF2_PB, 0, wcd934x_codec_enable_slim,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_IN_E("AIF3 PB", "AIF3 Playback", 0, SND_SOC_NOPM,
+ AIF3_PB, 0, wcd934x_codec_enable_slim,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_IN_E("AIF4 PB", "AIF4 Playback", 0, SND_SOC_NOPM,
+ AIF4_PB, 0, wcd934x_codec_enable_slim,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("SLIM RX0 MUX", SND_SOC_NOPM, WCD934X_RX0, 0,
+ &slim_rx_mux[WCD934X_RX0]),
+ SND_SOC_DAPM_MUX("SLIM RX1 MUX", SND_SOC_NOPM, WCD934X_RX1, 0,
+ &slim_rx_mux[WCD934X_RX1]),
+ SND_SOC_DAPM_MUX("SLIM RX2 MUX", SND_SOC_NOPM, WCD934X_RX2, 0,
+ &slim_rx_mux[WCD934X_RX2]),
+ SND_SOC_DAPM_MUX("SLIM RX3 MUX", SND_SOC_NOPM, WCD934X_RX3, 0,
+ &slim_rx_mux[WCD934X_RX3]),
+ SND_SOC_DAPM_MUX("SLIM RX4 MUX", SND_SOC_NOPM, WCD934X_RX4, 0,
+ &slim_rx_mux[WCD934X_RX4]),
+ SND_SOC_DAPM_MUX("SLIM RX5 MUX", SND_SOC_NOPM, WCD934X_RX5, 0,
+ &slim_rx_mux[WCD934X_RX5]),
+ SND_SOC_DAPM_MUX("SLIM RX6 MUX", SND_SOC_NOPM, WCD934X_RX6, 0,
+ &slim_rx_mux[WCD934X_RX6]),
+ SND_SOC_DAPM_MUX("SLIM RX7 MUX", SND_SOC_NOPM, WCD934X_RX7, 0,
+ &slim_rx_mux[WCD934X_RX7]),
+
+ SND_SOC_DAPM_MIXER("SLIM RX0", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM RX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM RX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM RX3", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM RX4", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM RX5", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM RX6", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SLIM RX7", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_MUX_E("RX INT0_2 MUX", SND_SOC_NOPM, INTERP_EAR, 0,
+ &rx_int0_2_mux, wcd934x_codec_enable_mix_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT1_2 MUX", SND_SOC_NOPM, INTERP_HPHL, 0,
+ &rx_int1_2_mux, wcd934x_codec_enable_mix_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT2_2 MUX", SND_SOC_NOPM, INTERP_HPHR, 0,
+ &rx_int2_2_mux, wcd934x_codec_enable_mix_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT3_2 MUX", SND_SOC_NOPM, INTERP_LO1, 0,
+ &rx_int3_2_mux, wcd934x_codec_enable_mix_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT4_2 MUX", SND_SOC_NOPM, INTERP_LO2, 0,
+ &rx_int4_2_mux, wcd934x_codec_enable_mix_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT7_2 MUX", SND_SOC_NOPM, INTERP_SPKR1, 0,
+ &rx_int7_2_mux, wcd934x_codec_enable_mix_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT8_2 MUX", SND_SOC_NOPM, INTERP_SPKR2, 0,
+ &rx_int8_2_mux, wcd934x_codec_enable_mix_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("RX INT0_1 MIX1 INP0", SND_SOC_NOPM, 0, 0,
+ &rx_int0_1_mix_inp0_mux),
+ SND_SOC_DAPM_MUX("RX INT0_1 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx_int0_1_mix_inp1_mux),
+ SND_SOC_DAPM_MUX("RX INT0_1 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx_int0_1_mix_inp2_mux),
+ SND_SOC_DAPM_MUX("RX INT1_1 MIX1 INP0", SND_SOC_NOPM, 0, 0,
+ &rx_int1_1_mix_inp0_mux),
+ SND_SOC_DAPM_MUX("RX INT1_1 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx_int1_1_mix_inp1_mux),
+ SND_SOC_DAPM_MUX("RX INT1_1 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx_int1_1_mix_inp2_mux),
+ SND_SOC_DAPM_MUX("RX INT2_1 MIX1 INP0", SND_SOC_NOPM, 0, 0,
+ &rx_int2_1_mix_inp0_mux),
+ SND_SOC_DAPM_MUX("RX INT2_1 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx_int2_1_mix_inp1_mux),
+ SND_SOC_DAPM_MUX("RX INT2_1 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx_int2_1_mix_inp2_mux),
+ SND_SOC_DAPM_MUX("RX INT3_1 MIX1 INP0", SND_SOC_NOPM, 0, 0,
+ &rx_int3_1_mix_inp0_mux),
+ SND_SOC_DAPM_MUX("RX INT3_1 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx_int3_1_mix_inp1_mux),
+ SND_SOC_DAPM_MUX("RX INT3_1 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx_int3_1_mix_inp2_mux),
+ SND_SOC_DAPM_MUX("RX INT4_1 MIX1 INP0", SND_SOC_NOPM, 0, 0,
+ &rx_int4_1_mix_inp0_mux),
+ SND_SOC_DAPM_MUX("RX INT4_1 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx_int4_1_mix_inp1_mux),
+ SND_SOC_DAPM_MUX("RX INT4_1 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx_int4_1_mix_inp2_mux),
+ SND_SOC_DAPM_MUX("RX INT7_1 MIX1 INP0", SND_SOC_NOPM, 0, 0,
+ &rx_int7_1_mix_inp0_mux),
+ SND_SOC_DAPM_MUX("RX INT7_1 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx_int7_1_mix_inp1_mux),
+ SND_SOC_DAPM_MUX("RX INT7_1 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx_int7_1_mix_inp2_mux),
+ SND_SOC_DAPM_MUX("RX INT8_1 MIX1 INP0", SND_SOC_NOPM, 0, 0,
+ &rx_int8_1_mix_inp0_mux),
+ SND_SOC_DAPM_MUX("RX INT8_1 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx_int8_1_mix_inp1_mux),
+ SND_SOC_DAPM_MUX("RX INT8_1 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx_int8_1_mix_inp2_mux),
+ SND_SOC_DAPM_MIXER("RX INT0_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT0 SEC MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT1_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT1 SEC MIX", SND_SOC_NOPM, 0, 0,
+ rx_int1_asrc_switch,
+ ARRAY_SIZE(rx_int1_asrc_switch)),
+ SND_SOC_DAPM_MIXER("RX INT2_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT2 SEC MIX", SND_SOC_NOPM, 0, 0,
+ rx_int2_asrc_switch,
+ ARRAY_SIZE(rx_int2_asrc_switch)),
+ SND_SOC_DAPM_MIXER("RX INT3_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT3 SEC MIX", SND_SOC_NOPM, 0, 0,
+ rx_int3_asrc_switch,
+ ARRAY_SIZE(rx_int3_asrc_switch)),
+ SND_SOC_DAPM_MIXER("RX INT4_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT4 SEC MIX", SND_SOC_NOPM, 0, 0,
+ rx_int4_asrc_switch,
+ ARRAY_SIZE(rx_int4_asrc_switch)),
+ SND_SOC_DAPM_MIXER("RX INT7_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT7 SEC MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT8_1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT8 SEC MIX", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT0 MIX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT1 MIX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT1 MIX3", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT2 MIX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT2 MIX3", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT3 MIX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT3 MIX3", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT4 MIX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX INT4 MIX3", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_MIXER("RX INT7 MIX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER_E("RX INT7 CHAIN", SND_SOC_NOPM, 0, 0,
+ NULL, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER_E("RX INT8 CHAIN", SND_SOC_NOPM, 0, 0,
+ NULL, 0, NULL, 0),
+ SND_SOC_DAPM_MUX_E("RX INT0 MIX2 INP", WCD934X_CDC_RX0_RX_PATH_CFG0, 4,
+ 0, &rx_int0_mix2_inp_mux, NULL,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT1 MIX2 INP", WCD934X_CDC_RX1_RX_PATH_CFG0, 4,
+ 0, &rx_int1_mix2_inp_mux, NULL,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT2 MIX2 INP", WCD934X_CDC_RX2_RX_PATH_CFG0, 4,
+ 0, &rx_int2_mix2_inp_mux, NULL,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT3 MIX2 INP", WCD934X_CDC_RX3_RX_PATH_CFG0, 4,
+ 0, &rx_int3_mix2_inp_mux, NULL,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT4 MIX2 INP", WCD934X_CDC_RX4_RX_PATH_CFG0, 4,
+ 0, &rx_int4_mix2_inp_mux, NULL,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT7 MIX2 INP", WCD934X_CDC_RX7_RX_PATH_CFG0, 4,
+ 0, &rx_int7_mix2_inp_mux, NULL,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("IIR0 INP0 MUX", SND_SOC_NOPM, 0, 0, &iir0_inp0_mux),
+ SND_SOC_DAPM_MUX("IIR0 INP1 MUX", SND_SOC_NOPM, 0, 0, &iir0_inp1_mux),
+ SND_SOC_DAPM_MUX("IIR0 INP2 MUX", SND_SOC_NOPM, 0, 0, &iir0_inp2_mux),
+ SND_SOC_DAPM_MUX("IIR0 INP3 MUX", SND_SOC_NOPM, 0, 0, &iir0_inp3_mux),
+ SND_SOC_DAPM_MUX("IIR1 INP0 MUX", SND_SOC_NOPM, 0, 0, &iir1_inp0_mux),
+ SND_SOC_DAPM_MUX("IIR1 INP1 MUX", SND_SOC_NOPM, 0, 0, &iir1_inp1_mux),
+ SND_SOC_DAPM_MUX("IIR1 INP2 MUX", SND_SOC_NOPM, 0, 0, &iir1_inp2_mux),
+ SND_SOC_DAPM_MUX("IIR1 INP3 MUX", SND_SOC_NOPM, 0, 0, &iir1_inp3_mux),
+
+ SND_SOC_DAPM_PGA_E("IIR0", WCD934X_CDC_SIDETONE_IIR0_IIR_GAIN_B1_CTL,
+ 0, 0, NULL, 0, wcd934x_codec_set_iir_gain,
+ SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PGA_E("IIR1", WCD934X_CDC_SIDETONE_IIR1_IIR_GAIN_B1_CTL,
+ 1, 0, NULL, 0, wcd934x_codec_set_iir_gain,
+ SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_MIXER("SRC0", WCD934X_CDC_SIDETONE_SRC0_ST_SRC_PATH_CTL,
+ 4, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("SRC1", WCD934X_CDC_SIDETONE_SRC1_ST_SRC_PATH_CTL,
+ 4, 0, NULL, 0),
+ SND_SOC_DAPM_MUX("RX INT0 DEM MUX", SND_SOC_NOPM, 0, 0,
+ &rx_int0_dem_inp_mux),
+ SND_SOC_DAPM_MUX("RX INT1 DEM MUX", SND_SOC_NOPM, 0, 0,
+ &rx_int1_dem_inp_mux),
+ SND_SOC_DAPM_MUX("RX INT2 DEM MUX", SND_SOC_NOPM, 0, 0,
+ &rx_int2_dem_inp_mux),
+
+ SND_SOC_DAPM_MUX_E("RX INT0_1 INTERP", SND_SOC_NOPM, INTERP_EAR, 0,
+ &rx_int0_1_interp_mux,
+ wcd934x_codec_enable_main_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT1_1 INTERP", SND_SOC_NOPM, INTERP_HPHL, 0,
+ &rx_int1_1_interp_mux,
+ wcd934x_codec_enable_main_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT2_1 INTERP", SND_SOC_NOPM, INTERP_HPHR, 0,
+ &rx_int2_1_interp_mux,
+ wcd934x_codec_enable_main_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT3_1 INTERP", SND_SOC_NOPM, INTERP_LO1, 0,
+ &rx_int3_1_interp_mux,
+ wcd934x_codec_enable_main_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT4_1 INTERP", SND_SOC_NOPM, INTERP_LO2, 0,
+ &rx_int4_1_interp_mux,
+ wcd934x_codec_enable_main_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT7_1 INTERP", SND_SOC_NOPM, INTERP_SPKR1, 0,
+ &rx_int7_1_interp_mux,
+ wcd934x_codec_enable_main_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX_E("RX INT8_1 INTERP", SND_SOC_NOPM, INTERP_SPKR2, 0,
+ &rx_int8_1_interp_mux,
+ wcd934x_codec_enable_main_path,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("RX INT0_2 INTERP", SND_SOC_NOPM, 0, 0,
+ &rx_int0_2_interp_mux),
+ SND_SOC_DAPM_MUX("RX INT1_2 INTERP", SND_SOC_NOPM, 0, 0,
+ &rx_int1_2_interp_mux),
+ SND_SOC_DAPM_MUX("RX INT2_2 INTERP", SND_SOC_NOPM, 0, 0,
+ &rx_int2_2_interp_mux),
+ SND_SOC_DAPM_MUX("RX INT3_2 INTERP", SND_SOC_NOPM, 0, 0,
+ &rx_int3_2_interp_mux),
+ SND_SOC_DAPM_MUX("RX INT4_2 INTERP", SND_SOC_NOPM, 0, 0,
+ &rx_int4_2_interp_mux),
+ SND_SOC_DAPM_MUX("RX INT7_2 INTERP", SND_SOC_NOPM, 0, 0,
+ &rx_int7_2_interp_mux),
+ SND_SOC_DAPM_MUX("RX INT8_2 INTERP", SND_SOC_NOPM, 0, 0,
+ &rx_int8_2_interp_mux),
+ SND_SOC_DAPM_DAC_E("RX INT0 DAC", NULL, SND_SOC_NOPM,
+ 0, 0, wcd934x_codec_ear_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_DAC_E("RX INT1 DAC", NULL, WCD934X_ANA_HPH,
+ 5, 0, wcd934x_codec_hphl_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_DAC_E("RX INT2 DAC", NULL, WCD934X_ANA_HPH,
+ 4, 0, wcd934x_codec_hphr_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_DAC_E("RX INT3 DAC", NULL, SND_SOC_NOPM,
+ 0, 0, wcd934x_codec_lineout_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_DAC_E("RX INT4 DAC", NULL, SND_SOC_NOPM,
+ 0, 0, wcd934x_codec_lineout_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("EAR PA", WCD934X_ANA_EAR, 7, 0, NULL, 0, NULL, 0),
+ SND_SOC_DAPM_PGA_E("HPHL PA", WCD934X_ANA_HPH, 7, 0, NULL, 0,
+ wcd934x_codec_enable_hphl_pa,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("HPHR PA", WCD934X_ANA_HPH, 6, 0, NULL, 0,
+ wcd934x_codec_enable_hphr_pa,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("LINEOUT1 PA", WCD934X_ANA_LO_1_2, 7, 0, NULL, 0,
+ NULL, 0),
+ SND_SOC_DAPM_PGA_E("LINEOUT2 PA", WCD934X_ANA_LO_1_2, 6, 0, NULL, 0,
+ NULL, 0),
+ SND_SOC_DAPM_SUPPLY("RX_BIAS", WCD934X_ANA_RX_SUPPLIES, 0, 0, NULL,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("SBOOST0", WCD934X_CDC_RX7_RX_PATH_CFG1,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("SBOOST0_CLK", WCD934X_CDC_BOOST0_BOOST_PATH_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("SBOOST1", WCD934X_CDC_RX8_RX_PATH_CFG1,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("SBOOST1_CLK", WCD934X_CDC_BOOST1_BOOST_PATH_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("INT0_CLK", SND_SOC_NOPM, INTERP_EAR, 0,
+ wcd934x_codec_enable_interp_clk,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("INT1_CLK", SND_SOC_NOPM, INTERP_HPHL, 0,
+ wcd934x_codec_enable_interp_clk,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("INT2_CLK", SND_SOC_NOPM, INTERP_HPHR, 0,
+ wcd934x_codec_enable_interp_clk,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("INT3_CLK", SND_SOC_NOPM, INTERP_LO1, 0,
+ wcd934x_codec_enable_interp_clk,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("INT4_CLK", SND_SOC_NOPM, INTERP_LO2, 0,
+ wcd934x_codec_enable_interp_clk,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("INT7_CLK", SND_SOC_NOPM, INTERP_SPKR1, 0,
+ wcd934x_codec_enable_interp_clk,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("INT8_CLK", SND_SOC_NOPM, INTERP_SPKR2, 0,
+ wcd934x_codec_enable_interp_clk,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("DSMDEM0_CLK", WCD934X_CDC_RX0_RX_PATH_DSMDEM_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DSMDEM1_CLK", WCD934X_CDC_RX1_RX_PATH_DSMDEM_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DSMDEM2_CLK", WCD934X_CDC_RX2_RX_PATH_DSMDEM_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DSMDEM3_CLK", WCD934X_CDC_RX3_RX_PATH_DSMDEM_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DSMDEM4_CLK", WCD934X_CDC_RX4_RX_PATH_DSMDEM_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DSMDEM7_CLK", WCD934X_CDC_RX7_RX_PATH_DSMDEM_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DSMDEM8_CLK", WCD934X_CDC_RX8_RX_PATH_DSMDEM_CTL,
+ 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0,
+ wcd934x_codec_enable_mclk,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+};
+
static const struct snd_soc_component_driver wcd934x_component_drv = {
.probe = wcd934x_comp_probe,
.remove = wcd934x_comp_remove,
.set_sysclk = wcd934x_comp_set_sysclk,
.controls = wcd934x_snd_controls,
.num_controls = ARRAY_SIZE(wcd934x_snd_controls),
+ .dapm_widgets = wcd934x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wcd934x_dapm_widgets),
};
static int wcd934x_codec_parse_data(struct wcd934x_codec *wcd)
--
2.20.1
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^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: wcd934x: add basic controls" to the asoc tree
@ 2019-12-25 18:21 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2019-12-25 18:21 UTC (permalink / raw)
To: Srinivas Kandagatla; +Cc: alsa-devel, Mark Brown
The patch
ASoC: wcd934x: add basic controls
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 1cde8b82233275a72db34a8c1efa4cf6cc8c894f Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Thu, 19 Dec 2019 10:31:46 +0000
Subject: [PATCH] ASoC: wcd934x: add basic controls
This patch adds basic controls found in wcd934x codec.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20191219103153.14875-5-srinivas.kandagatla@linaro.org
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wcd934x.c | 506 +++++++++++++++++++++++++++++++++++++
1 file changed, 506 insertions(+)
diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c
index 825cf057e8b7..27a864f2ff77 100644
--- a/sound/soc/codecs/wcd934x.c
+++ b/sound/soc/codecs/wcd934x.c
@@ -120,6 +120,21 @@
#define WCD934X_DEF_MICBIAS_MV 1800
#define WCD934X_MAX_MICBIAS_MV 2850
+#define WCD_IIR_FILTER_SIZE (sizeof(u32) * BAND_MAX)
+
+#define WCD_IIR_FILTER_CTL(xname, iidx, bidx) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = wcd934x_iir_filter_info, \
+ .get = wcd934x_get_iir_band_audio_mixer, \
+ .put = wcd934x_put_iir_band_audio_mixer, \
+ .private_value = (unsigned long)&(struct wcd_iir_filter_ctl) { \
+ .iir_idx = iidx, \
+ .band_idx = bidx, \
+ .bytes_ext = {.max = WCD_IIR_FILTER_SIZE, }, \
+ } \
+}
+
enum {
SIDO_SOURCE_INTERNAL,
SIDO_SOURCE_RCO_BG,
@@ -217,6 +232,35 @@ static const struct wcd934x_slim_ch wcd934x_rx_chs[WCD934X_RX_MAX] = {
WCD934X_SLIM_RX_CH(12),
};
+/* Codec supports 2 IIR filters */
+enum {
+ IIR0 = 0,
+ IIR1,
+ IIR_MAX,
+};
+
+/* Each IIR has 5 Filter Stages */
+enum {
+ BAND1 = 0,
+ BAND2,
+ BAND3,
+ BAND4,
+ BAND5,
+ BAND_MAX,
+};
+
+enum {
+ COMPANDER_1, /* HPH_L */
+ COMPANDER_2, /* HPH_R */
+ COMPANDER_3, /* LO1_DIFF */
+ COMPANDER_4, /* LO2_DIFF */
+ COMPANDER_5, /* LO3_SE - not used in Tavil */
+ COMPANDER_6, /* LO4_SE - not used in Tavil */
+ COMPANDER_7, /* SWR SPK CH1 */
+ COMPANDER_8, /* SWR SPK CH2 */
+ COMPANDER_MAX,
+};
+
enum {
AIF1_PB = 0,
AIF1_CAP,
@@ -335,12 +379,112 @@ struct wcd934x_codec {
int dmic_2_3_clk_cnt;
int dmic_4_5_clk_cnt;
int dmic_sample_rate;
+ int comp_enabled[COMPANDER_MAX];
int sysclk_users;
struct mutex sysclk_mutex;
};
#define to_wcd934x_codec(_hw) container_of(_hw, struct wcd934x_codec, hw)
+struct wcd_iir_filter_ctl {
+ unsigned int iir_idx;
+ unsigned int band_idx;
+ struct soc_bytes_ext bytes_ext;
+};
+
+static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0);
+static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1);
+static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1);
+static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0);
+
+/* Cutoff frequency for high pass filter */
+static const char * const cf_text[] = {
+ "CF_NEG_3DB_4HZ", "CF_NEG_3DB_75HZ", "CF_NEG_3DB_150HZ"
+};
+
+static const char * const rx_cf_text[] = {
+ "CF_NEG_3DB_4HZ", "CF_NEG_3DB_75HZ", "CF_NEG_3DB_150HZ",
+ "CF_NEG_3DB_0P48HZ"
+};
+
+static const char * const rx_hph_mode_mux_text[] = {
+ "Class H Invalid", "Class-H Hi-Fi", "Class-H Low Power", "Class-AB",
+ "Class-H Hi-Fi Low Power"
+};
+
+static const struct soc_enum cf_dec0_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX0_TX_PATH_CFG0, 5, 3, cf_text);
+
+static const struct soc_enum cf_dec1_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX1_TX_PATH_CFG0, 5, 3, cf_text);
+
+static const struct soc_enum cf_dec2_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX2_TX_PATH_CFG0, 5, 3, cf_text);
+
+static const struct soc_enum cf_dec3_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX3_TX_PATH_CFG0, 5, 3, cf_text);
+
+static const struct soc_enum cf_dec4_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX4_TX_PATH_CFG0, 5, 3, cf_text);
+
+static const struct soc_enum cf_dec5_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX5_TX_PATH_CFG0, 5, 3, cf_text);
+
+static const struct soc_enum cf_dec6_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX6_TX_PATH_CFG0, 5, 3, cf_text);
+
+static const struct soc_enum cf_dec7_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX7_TX_PATH_CFG0, 5, 3, cf_text);
+
+static const struct soc_enum cf_dec8_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_TX8_TX_PATH_CFG0, 5, 3, cf_text);
+
+static const struct soc_enum cf_int0_1_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX0_RX_PATH_CFG2, 0, 4, rx_cf_text);
+
+static SOC_ENUM_SINGLE_DECL(cf_int0_2_enum, WCD934X_CDC_RX0_RX_PATH_MIX_CFG, 2,
+ rx_cf_text);
+
+static const struct soc_enum cf_int1_1_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX1_RX_PATH_CFG2, 0, 4, rx_cf_text);
+
+static SOC_ENUM_SINGLE_DECL(cf_int1_2_enum, WCD934X_CDC_RX1_RX_PATH_MIX_CFG, 2,
+ rx_cf_text);
+
+static const struct soc_enum cf_int2_1_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX2_RX_PATH_CFG2, 0, 4, rx_cf_text);
+
+static SOC_ENUM_SINGLE_DECL(cf_int2_2_enum, WCD934X_CDC_RX2_RX_PATH_MIX_CFG, 2,
+ rx_cf_text);
+
+static const struct soc_enum cf_int3_1_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX3_RX_PATH_CFG2, 0, 4, rx_cf_text);
+
+static SOC_ENUM_SINGLE_DECL(cf_int3_2_enum, WCD934X_CDC_RX3_RX_PATH_MIX_CFG, 2,
+ rx_cf_text);
+
+static const struct soc_enum cf_int4_1_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX4_RX_PATH_CFG2, 0, 4, rx_cf_text);
+
+static SOC_ENUM_SINGLE_DECL(cf_int4_2_enum, WCD934X_CDC_RX4_RX_PATH_MIX_CFG, 2,
+ rx_cf_text);
+
+static const struct soc_enum cf_int7_1_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX7_RX_PATH_CFG2, 0, 4, rx_cf_text);
+
+static SOC_ENUM_SINGLE_DECL(cf_int7_2_enum, WCD934X_CDC_RX7_RX_PATH_MIX_CFG, 2,
+ rx_cf_text);
+
+static const struct soc_enum cf_int8_1_enum =
+ SOC_ENUM_SINGLE(WCD934X_CDC_RX8_RX_PATH_CFG2, 0, 4, rx_cf_text);
+
+static SOC_ENUM_SINGLE_DECL(cf_int8_2_enum, WCD934X_CDC_RX8_RX_PATH_MIX_CFG, 2,
+ rx_cf_text);
+
+static const struct soc_enum rx_hph_mode_mux_enum =
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(rx_hph_mode_mux_text),
+ rx_hph_mode_mux_text);
+
static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd,
int sido_src)
{
@@ -1479,10 +1623,372 @@ static int wcd934x_comp_set_sysclk(struct snd_soc_component *comp,
return clk_set_rate(wcd->extclk, freq);
}
+static uint32_t get_iir_band_coeff(struct snd_soc_component *component,
+ int iir_idx, int band_idx, int coeff_idx)
+{
+ u32 value = 0;
+ int reg, b2_reg;
+
+ /* Address does not automatically update if reading */
+ reg = WCD934X_CDC_SIDETONE_IIR0_IIR_COEF_B1_CTL + 16 * iir_idx;
+ b2_reg = WCD934X_CDC_SIDETONE_IIR0_IIR_COEF_B2_CTL + 16 * iir_idx;
+
+ snd_soc_component_write(component, reg,
+ ((band_idx * BAND_MAX + coeff_idx) *
+ sizeof(uint32_t)) & 0x7F);
+
+ value |= snd_soc_component_read32(component, b2_reg);
+ snd_soc_component_write(component, reg,
+ ((band_idx * BAND_MAX + coeff_idx)
+ * sizeof(uint32_t) + 1) & 0x7F);
+
+ value |= (snd_soc_component_read32(component, b2_reg) << 8);
+ snd_soc_component_write(component, reg,
+ ((band_idx * BAND_MAX + coeff_idx)
+ * sizeof(uint32_t) + 2) & 0x7F);
+
+ value |= (snd_soc_component_read32(component, b2_reg) << 16);
+ snd_soc_component_write(component, reg,
+ ((band_idx * BAND_MAX + coeff_idx)
+ * sizeof(uint32_t) + 3) & 0x7F);
+
+ /* Mask bits top 2 bits since they are reserved */
+ value |= (snd_soc_component_read32(component, b2_reg) << 24);
+ return value;
+}
+
+static void set_iir_band_coeff(struct snd_soc_component *component,
+ int iir_idx, int band_idx, uint32_t value)
+{
+ int reg = WCD934X_CDC_SIDETONE_IIR0_IIR_COEF_B2_CTL + 16 * iir_idx;
+
+ snd_soc_component_write(component, reg, (value & 0xFF));
+ snd_soc_component_write(component, reg, (value >> 8) & 0xFF);
+ snd_soc_component_write(component, reg, (value >> 16) & 0xFF);
+ /* Mask top 2 bits, 7-8 are reserved */
+ snd_soc_component_write(component, reg, (value >> 24) & 0x3F);
+}
+
+static int wcd934x_put_iir_band_audio_mixer(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component =
+ snd_soc_kcontrol_component(kcontrol);
+ struct wcd_iir_filter_ctl *ctl =
+ (struct wcd_iir_filter_ctl *)kcontrol->private_value;
+ struct soc_bytes_ext *params = &ctl->bytes_ext;
+ int iir_idx = ctl->iir_idx;
+ int band_idx = ctl->band_idx;
+ u32 coeff[BAND_MAX];
+ int reg = WCD934X_CDC_SIDETONE_IIR0_IIR_COEF_B1_CTL + 16 * iir_idx;
+
+ memcpy(&coeff[0], ucontrol->value.bytes.data, params->max);
+
+ /* Mask top bit it is reserved */
+ /* Updates addr automatically for each B2 write */
+ snd_soc_component_write(component, reg, (band_idx * BAND_MAX *
+ sizeof(uint32_t)) & 0x7F);
+
+ set_iir_band_coeff(component, iir_idx, band_idx, coeff[0]);
+ set_iir_band_coeff(component, iir_idx, band_idx, coeff[1]);
+ set_iir_band_coeff(component, iir_idx, band_idx, coeff[2]);
+ set_iir_band_coeff(component, iir_idx, band_idx, coeff[3]);
+ set_iir_band_coeff(component, iir_idx, band_idx, coeff[4]);
+
+ return 0;
+}
+
+static int wcd934x_get_iir_band_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component =
+ snd_soc_kcontrol_component(kcontrol);
+ struct wcd_iir_filter_ctl *ctl =
+ (struct wcd_iir_filter_ctl *)kcontrol->private_value;
+ struct soc_bytes_ext *params = &ctl->bytes_ext;
+ int iir_idx = ctl->iir_idx;
+ int band_idx = ctl->band_idx;
+ u32 coeff[BAND_MAX];
+
+ coeff[0] = get_iir_band_coeff(component, iir_idx, band_idx, 0);
+ coeff[1] = get_iir_band_coeff(component, iir_idx, band_idx, 1);
+ coeff[2] = get_iir_band_coeff(component, iir_idx, band_idx, 2);
+ coeff[3] = get_iir_band_coeff(component, iir_idx, band_idx, 3);
+ coeff[4] = get_iir_band_coeff(component, iir_idx, band_idx, 4);
+
+ memcpy(ucontrol->value.bytes.data, &coeff[0], params->max);
+
+ return 0;
+}
+
+static int wcd934x_iir_filter_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *ucontrol)
+{
+ struct wcd_iir_filter_ctl *ctl =
+ (struct wcd_iir_filter_ctl *)kcontrol->private_value;
+ struct soc_bytes_ext *params = &ctl->bytes_ext;
+
+ ucontrol->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ ucontrol->count = params->max;
+
+ return 0;
+}
+
+static int wcd934x_compander_get(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_soc_kcontrol_component(kc);
+ int comp = ((struct soc_mixer_control *)kc->private_value)->shift;
+ struct wcd934x_codec *wcd = dev_get_drvdata(component->dev);
+
+ ucontrol->value.integer.value[0] = wcd->comp_enabled[comp];
+
+ return 0;
+}
+
+static int wcd934x_compander_set(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_soc_kcontrol_component(kc);
+ struct wcd934x_codec *wcd = dev_get_drvdata(component->dev);
+ int comp = ((struct soc_mixer_control *)kc->private_value)->shift;
+ int value = ucontrol->value.integer.value[0];
+ int sel;
+
+ wcd->comp_enabled[comp] = value;
+ sel = value ? WCD934X_HPH_GAIN_SRC_SEL_COMPANDER :
+ WCD934X_HPH_GAIN_SRC_SEL_REGISTER;
+
+ /* Any specific register configuration for compander */
+ switch (comp) {
+ case COMPANDER_1:
+ /* Set Gain Source Select based on compander enable/disable */
+ snd_soc_component_update_bits(component, WCD934X_HPH_L_EN,
+ WCD934X_HPH_GAIN_SRC_SEL_MASK,
+ sel);
+ break;
+ case COMPANDER_2:
+ snd_soc_component_update_bits(component, WCD934X_HPH_R_EN,
+ WCD934X_HPH_GAIN_SRC_SEL_MASK,
+ sel);
+ break;
+ case COMPANDER_3:
+ case COMPANDER_4:
+ case COMPANDER_7:
+ case COMPANDER_8:
+ break;
+ default:
+ break;
+ };
+
+ return 0;
+}
+
+static int wcd934x_rx_hph_mode_get(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_soc_kcontrol_component(kc);
+ struct wcd934x_codec *wcd = dev_get_drvdata(component->dev);
+
+ ucontrol->value.enumerated.item[0] = wcd->hph_mode;
+
+ return 0;
+}
+
+static int wcd934x_rx_hph_mode_put(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *component = snd_soc_kcontrol_component(kc);
+ struct wcd934x_codec *wcd = dev_get_drvdata(component->dev);
+ u32 mode_val;
+
+ mode_val = ucontrol->value.enumerated.item[0];
+
+ if (mode_val == 0) {
+ dev_err(wcd->dev, "Invalid HPH Mode, default to ClSH HiFi\n");
+ mode_val = CLS_H_LOHIFI;
+ }
+ wcd->hph_mode = mode_val;
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new wcd934x_snd_controls[] = {
+ /* Gain Controls */
+ SOC_SINGLE_TLV("EAR PA Volume", WCD934X_ANA_EAR, 4, 4, 1, ear_pa_gain),
+ SOC_SINGLE_TLV("HPHL Volume", WCD934X_HPH_L_EN, 0, 24, 1, line_gain),
+ SOC_SINGLE_TLV("HPHR Volume", WCD934X_HPH_R_EN, 0, 24, 1, line_gain),
+ SOC_SINGLE_TLV("LINEOUT1 Volume", WCD934X_DIFF_LO_LO1_COMPANDER,
+ 3, 16, 1, line_gain),
+ SOC_SINGLE_TLV("LINEOUT2 Volume", WCD934X_DIFF_LO_LO2_COMPANDER,
+ 3, 16, 1, line_gain),
+
+ SOC_SINGLE_TLV("ADC1 Volume", WCD934X_ANA_AMIC1, 0, 20, 0, analog_gain),
+ SOC_SINGLE_TLV("ADC2 Volume", WCD934X_ANA_AMIC2, 0, 20, 0, analog_gain),
+ SOC_SINGLE_TLV("ADC3 Volume", WCD934X_ANA_AMIC3, 0, 20, 0, analog_gain),
+ SOC_SINGLE_TLV("ADC4 Volume", WCD934X_ANA_AMIC4, 0, 20, 0, analog_gain),
+
+ SOC_SINGLE_S8_TLV("RX0 Digital Volume", WCD934X_CDC_RX0_RX_VOL_CTL,
+ -84, 40, digital_gain), /* -84dB min - 40dB max */
+ SOC_SINGLE_S8_TLV("RX1 Digital Volume", WCD934X_CDC_RX1_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX2 Digital Volume", WCD934X_CDC_RX2_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX3 Digital Volume", WCD934X_CDC_RX3_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX4 Digital Volume", WCD934X_CDC_RX4_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX7 Digital Volume", WCD934X_CDC_RX7_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX8 Digital Volume", WCD934X_CDC_RX8_RX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX0 Mix Digital Volume",
+ WCD934X_CDC_RX0_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX1 Mix Digital Volume",
+ WCD934X_CDC_RX1_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX2 Mix Digital Volume",
+ WCD934X_CDC_RX2_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX3 Mix Digital Volume",
+ WCD934X_CDC_RX3_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX4 Mix Digital Volume",
+ WCD934X_CDC_RX4_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX7 Mix Digital Volume",
+ WCD934X_CDC_RX7_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX8 Mix Digital Volume",
+ WCD934X_CDC_RX8_RX_VOL_MIX_CTL,
+ -84, 40, digital_gain),
+
+ SOC_SINGLE_S8_TLV("DEC0 Volume", WCD934X_CDC_TX0_TX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("DEC1 Volume", WCD934X_CDC_TX1_TX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("DEC2 Volume", WCD934X_CDC_TX2_TX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("DEC3 Volume", WCD934X_CDC_TX3_TX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("DEC4 Volume", WCD934X_CDC_TX4_TX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("DEC5 Volume", WCD934X_CDC_TX5_TX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("DEC6 Volume", WCD934X_CDC_TX6_TX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("DEC7 Volume", WCD934X_CDC_TX7_TX_VOL_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("DEC8 Volume", WCD934X_CDC_TX8_TX_VOL_CTL,
+ -84, 40, digital_gain),
+
+ SOC_SINGLE_S8_TLV("IIR0 INP0 Volume",
+ WCD934X_CDC_SIDETONE_IIR0_IIR_GAIN_B1_CTL, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("IIR0 INP1 Volume",
+ WCD934X_CDC_SIDETONE_IIR0_IIR_GAIN_B2_CTL, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("IIR0 INP2 Volume",
+ WCD934X_CDC_SIDETONE_IIR0_IIR_GAIN_B3_CTL, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("IIR0 INP3 Volume",
+ WCD934X_CDC_SIDETONE_IIR0_IIR_GAIN_B4_CTL, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP0 Volume",
+ WCD934X_CDC_SIDETONE_IIR1_IIR_GAIN_B1_CTL, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP1 Volume",
+ WCD934X_CDC_SIDETONE_IIR1_IIR_GAIN_B2_CTL, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP2 Volume",
+ WCD934X_CDC_SIDETONE_IIR1_IIR_GAIN_B3_CTL, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP3 Volume",
+ WCD934X_CDC_SIDETONE_IIR1_IIR_GAIN_B4_CTL, -84, 40,
+ digital_gain),
+
+ SOC_ENUM("TX0 HPF cut off", cf_dec0_enum),
+ SOC_ENUM("TX1 HPF cut off", cf_dec1_enum),
+ SOC_ENUM("TX2 HPF cut off", cf_dec2_enum),
+ SOC_ENUM("TX3 HPF cut off", cf_dec3_enum),
+ SOC_ENUM("TX4 HPF cut off", cf_dec4_enum),
+ SOC_ENUM("TX5 HPF cut off", cf_dec5_enum),
+ SOC_ENUM("TX6 HPF cut off", cf_dec6_enum),
+ SOC_ENUM("TX7 HPF cut off", cf_dec7_enum),
+ SOC_ENUM("TX8 HPF cut off", cf_dec8_enum),
+
+ SOC_ENUM("RX INT0_1 HPF cut off", cf_int0_1_enum),
+ SOC_ENUM("RX INT0_2 HPF cut off", cf_int0_2_enum),
+ SOC_ENUM("RX INT1_1 HPF cut off", cf_int1_1_enum),
+ SOC_ENUM("RX INT1_2 HPF cut off", cf_int1_2_enum),
+ SOC_ENUM("RX INT2_1 HPF cut off", cf_int2_1_enum),
+ SOC_ENUM("RX INT2_2 HPF cut off", cf_int2_2_enum),
+ SOC_ENUM("RX INT3_1 HPF cut off", cf_int3_1_enum),
+ SOC_ENUM("RX INT3_2 HPF cut off", cf_int3_2_enum),
+ SOC_ENUM("RX INT4_1 HPF cut off", cf_int4_1_enum),
+ SOC_ENUM("RX INT4_2 HPF cut off", cf_int4_2_enum),
+ SOC_ENUM("RX INT7_1 HPF cut off", cf_int7_1_enum),
+ SOC_ENUM("RX INT7_2 HPF cut off", cf_int7_2_enum),
+ SOC_ENUM("RX INT8_1 HPF cut off", cf_int8_1_enum),
+ SOC_ENUM("RX INT8_2 HPF cut off", cf_int8_2_enum),
+
+ SOC_ENUM_EXT("RX HPH Mode", rx_hph_mode_mux_enum,
+ wcd934x_rx_hph_mode_get, wcd934x_rx_hph_mode_put),
+
+ SOC_SINGLE("IIR1 Band1 Switch", WCD934X_CDC_SIDETONE_IIR0_IIR_CTL,
+ 0, 1, 0),
+ SOC_SINGLE("IIR1 Band2 Switch", WCD934X_CDC_SIDETONE_IIR0_IIR_CTL,
+ 1, 1, 0),
+ SOC_SINGLE("IIR1 Band3 Switch", WCD934X_CDC_SIDETONE_IIR0_IIR_CTL,
+ 2, 1, 0),
+ SOC_SINGLE("IIR1 Band4 Switch", WCD934X_CDC_SIDETONE_IIR0_IIR_CTL,
+ 3, 1, 0),
+ SOC_SINGLE("IIR1 Band5 Switch", WCD934X_CDC_SIDETONE_IIR0_IIR_CTL,
+ 4, 1, 0),
+ SOC_SINGLE("IIR2 Band1 Switch", WCD934X_CDC_SIDETONE_IIR1_IIR_CTL,
+ 0, 1, 0),
+ SOC_SINGLE("IIR2 Band2 Switch", WCD934X_CDC_SIDETONE_IIR1_IIR_CTL,
+ 1, 1, 0),
+ SOC_SINGLE("IIR2 Band3 Switch", WCD934X_CDC_SIDETONE_IIR1_IIR_CTL,
+ 2, 1, 0),
+ SOC_SINGLE("IIR2 Band4 Switch", WCD934X_CDC_SIDETONE_IIR1_IIR_CTL,
+ 3, 1, 0),
+ SOC_SINGLE("IIR2 Band5 Switch", WCD934X_CDC_SIDETONE_IIR1_IIR_CTL,
+ 4, 1, 0),
+ WCD_IIR_FILTER_CTL("IIR0 Band1", IIR0, BAND1),
+ WCD_IIR_FILTER_CTL("IIR0 Band2", IIR0, BAND2),
+ WCD_IIR_FILTER_CTL("IIR0 Band3", IIR0, BAND3),
+ WCD_IIR_FILTER_CTL("IIR0 Band4", IIR0, BAND4),
+ WCD_IIR_FILTER_CTL("IIR0 Band5", IIR0, BAND5),
+
+ WCD_IIR_FILTER_CTL("IIR1 Band1", IIR1, BAND1),
+ WCD_IIR_FILTER_CTL("IIR1 Band2", IIR1, BAND2),
+ WCD_IIR_FILTER_CTL("IIR1 Band3", IIR1, BAND3),
+ WCD_IIR_FILTER_CTL("IIR1 Band4", IIR1, BAND4),
+ WCD_IIR_FILTER_CTL("IIR1 Band5", IIR1, BAND5),
+
+ SOC_SINGLE_EXT("COMP1 Switch", SND_SOC_NOPM, COMPANDER_1, 1, 0,
+ wcd934x_compander_get, wcd934x_compander_set),
+ SOC_SINGLE_EXT("COMP2 Switch", SND_SOC_NOPM, COMPANDER_2, 1, 0,
+ wcd934x_compander_get, wcd934x_compander_set),
+ SOC_SINGLE_EXT("COMP3 Switch", SND_SOC_NOPM, COMPANDER_3, 1, 0,
+ wcd934x_compander_get, wcd934x_compander_set),
+ SOC_SINGLE_EXT("COMP4 Switch", SND_SOC_NOPM, COMPANDER_4, 1, 0,
+ wcd934x_compander_get, wcd934x_compander_set),
+ SOC_SINGLE_EXT("COMP7 Switch", SND_SOC_NOPM, COMPANDER_7, 1, 0,
+ wcd934x_compander_get, wcd934x_compander_set),
+ SOC_SINGLE_EXT("COMP8 Switch", SND_SOC_NOPM, COMPANDER_8, 1, 0,
+ wcd934x_compander_get, wcd934x_compander_set),
+};
+
static const struct snd_soc_component_driver wcd934x_component_drv = {
.probe = wcd934x_comp_probe,
.remove = wcd934x_comp_remove,
.set_sysclk = wcd934x_comp_set_sysclk,
+ .controls = wcd934x_snd_controls,
+ .num_controls = ARRAY_SIZE(wcd934x_snd_controls),
};
static int wcd934x_codec_parse_data(struct wcd934x_codec *wcd)
--
2.20.1
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^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec" to the asoc tree
@ 2019-12-25 18:21 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2019-12-25 18:21 UTC (permalink / raw)
To: Srinivas Kandagatla; +Cc: alsa-devel, Mark Brown
The patch
ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From a6b748713205242d1483a78662bf147b1aa7d90b Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Thu, 19 Dec 2019 10:31:43 +0000
Subject: [PATCH] ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio
codec
This patch adds bindings for wcd9340/wcd9341 audio codec which can
support both SLIMbus and I2S/I2C interface.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20191219103153.14875-2-srinivas.kandagatla@linaro.org
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../bindings/sound/qcom,wcd934x.yaml | 175 ++++++++++++++++++
1 file changed, 175 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml
diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml
new file mode 100644
index 000000000000..38eaf0c028f9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml
@@ -0,0 +1,175 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/qcom,wcd934x.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Bindings for Qualcomm WCD9340/WCD9341 Audio Codec
+
+maintainers:
+ - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
+
+description: |
+ Qualcomm WCD9340/WCD9341 Codec is a standalone Hi-Fi audio codec IC.
+ It has in-built Soundwire controller, pin controller, interrupt mux and
+ supports both I2S/I2C and SLIMbus audio interfaces.
+
+properties:
+ compatible:
+ const: slim217,250
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ reset-gpios:
+ description: GPIO spec for reset line to use
+ maxItems: 1
+
+ slim-ifc-dev: true
+
+ clocks:
+ maxItems: 1
+
+ clock-names:
+ const: extclk
+
+ vdd-buck-supply:
+ description: A reference to the 1.8V buck supply
+
+ vdd-buck-sido-supply:
+ description: A reference to the 1.8V SIDO buck supply
+
+ vdd-rx-supply:
+ description: A reference to the 1.8V rx supply
+
+ vdd-tx-supply:
+ description: A reference to the 1.8V tx supply
+
+ vdd-vbat-supply:
+ description: A reference to the vbat supply
+
+ vdd-io-supply:
+ description: A reference to the 1.8V I/O supply
+
+ vdd-micbias-supply:
+ description: A reference to the micbias supply
+
+ qcom,micbias1-microvolt:
+ description: micbias1 voltage
+ minimum: 1800000
+ maximum: 2850000
+
+ qcom,micbias2-microvolt:
+ description: micbias2 voltage
+ minimum: 1800000
+ maximum: 2850000
+
+ qcom,micbias3-microvolt:
+ description: micbias3 voltage
+ minimum: 1800000
+ maximum: 2850000
+
+ qcom,micbias4-microvolt:
+ description: micbias4 voltage
+ minimum: 1800000
+ maximum: 2850000
+
+ clock-output-names:
+ const: mclk
+
+ clock-frequency:
+ description: Clock frequency of output clk in Hz
+
+ interrupt-controller: true
+
+ '#interrupt-cells':
+ const: 1
+
+ '#clock-cells':
+ const: 0
+
+ '#sound-dai-cells':
+ const: 1
+
+ "#address-cells":
+ const: 1
+
+ "#size-cells":
+ const: 1
+
+ gpio@42:
+ type: object
+ allOf:
+ - $ref: ../gpio/qcom,wcd934x-gpio.yaml#
+
+patternProperties:
+ "^.*@[0-9a-f]+$":
+ type: object
+ description: |
+ WCD934x subnode for each slave devices. Bindings of each subnodes
+ depends on the specific driver providing the functionality and
+ documented in their respective bindings.
+
+ properties:
+ reg:
+ maxItems: 1
+
+ required:
+ - reg
+
+required:
+ - compatible
+ - reg
+ - reset-gpios
+ - slim-ifc-dev
+ - interrupts
+ - interrupt-controller
+ - clock-frequency
+ - clock-output-names
+ - qcom,micbias1-microvolt
+ - qcom,micbias2-microvolt
+ - qcom,micbias3-microvolt
+ - qcom,micbias4-microvolt
+ - "#interrupt-cells"
+ - "#clock-cells"
+ - "#sound-dai-cells"
+ - "#address-cells"
+ - "#size-cells"
+
+examples:
+ - |
+ codec@1,0{
+ compatible = "slim217,250";
+ reg = <1 0>;
+ reset-gpios = <&tlmm 64 0>;
+ slim-ifc-dev = <&wcd9340_ifd>;
+ #sound-dai-cells = <1>;
+ interrupt-parent = <&tlmm>;
+ interrupts = <54 4>;
+ interrupt-controller;
+ #interrupt-cells = <1>;
+ #clock-cells = <0>;
+ clock-frequency = <9600000>;
+ clock-output-names = "mclk";
+ qcom,micbias1-microvolt = <1800000>;
+ qcom,micbias2-microvolt = <1800000>;
+ qcom,micbias3-microvolt = <1800000>;
+ qcom,micbias4-microvolt = <1800000>;
+ clock-names = "extclk";
+ clocks = <&rpmhcc 2>;
+
+ #address-cells = <1>;
+ #size-cells = <1>;
+
+ gpio@42 {
+ compatible = "qcom,wcd9340-gpio";
+ reg = <0x42 0x2>;
+ gpio-controller;
+ #gpio-cells = <2>;
+ };
+ };
+
+...
--
2.20.1
_______________________________________________
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Alsa-devel@alsa-project.org
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: amd: Create multiple I2S platform device endpoint" to the asoc tree
@ 2019-12-27 1:08 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2019-12-27 1:08 UTC (permalink / raw)
To: Ravulapati Vishnu vardhan rao; +Cc: alsa-devel, Mark Brown
The patch
ASoC: amd: Create multiple I2S platform device endpoint
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 5def0136c609062091b03a91ac1d987f4afc5363 Mon Sep 17 00:00:00 2001
From: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Date: Thu, 5 Dec 2019 19:07:26 +0530
Subject: [PATCH] ASoC: amd: Create multiple I2S platform device endpoint
Creates Platform Device endpoints for multiple
I2S instances: SP and BT endpoints device.
Pass PCI resources like MMIO, irq to the platform devices.
Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Reviewed-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/1575553053-18344-2-git-send-email-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/amd/raven/acp3x.h | 5 ++
sound/soc/amd/raven/pci-acp3x.c | 95 ++++++++++++++++++++++-----------
2 files changed, 68 insertions(+), 32 deletions(-)
diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h
index 4f2cadd90a87..2f15fe10ead5 100644
--- a/sound/soc/amd/raven/acp3x.h
+++ b/sound/soc/amd/raven/acp3x.h
@@ -7,10 +7,15 @@
#include "chip_offset_byte.h"
+#define ACP3x_DEVS 3
#define ACP3x_PHY_BASE_ADDRESS 0x1240000
#define ACP3x_I2S_MODE 0
#define ACP3x_REG_START 0x1240000
#define ACP3x_REG_END 0x1250200
+#define ACP3x_I2STDM_REG_START 0x1242400
+#define ACP3x_I2STDM_REG_END 0x1242410
+#define ACP3x_BT_TDM_REG_START 0x1242800
+#define ACP3x_BT_TDM_REG_END 0x1242810
#define I2S_MODE 0x04
#define BT_TX_THRESHOLD 26
#define BT_RX_THRESHOLD 25
diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c
index facec2472b34..94f5f21d9a53 100644
--- a/sound/soc/amd/raven/pci-acp3x.c
+++ b/sound/soc/amd/raven/pci-acp3x.c
@@ -16,17 +16,17 @@ struct acp3x_dev_data {
void __iomem *acp3x_base;
bool acp3x_audio_mode;
struct resource *res;
- struct platform_device *pdev;
+ struct platform_device *pdev[ACP3x_DEVS];
};
static int snd_acp3x_probe(struct pci_dev *pci,
const struct pci_device_id *pci_id)
{
- int ret;
- u32 addr, val;
struct acp3x_dev_data *adata;
- struct platform_device_info pdevinfo;
+ struct platform_device_info pdevinfo[ACP3x_DEVS];
unsigned int irqflags;
+ int ret, i;
+ u32 addr, val;
if (pci_enable_device(pci)) {
dev_err(&pci->dev, "pci_enable_device failed\n");
@@ -56,10 +56,11 @@ static int snd_acp3x_probe(struct pci_dev *pci,
irqflags = 0;
addr = pci_resource_start(pci, 0);
- adata->acp3x_base = ioremap(addr, pci_resource_len(pci, 0));
+ adata->acp3x_base = devm_ioremap(&pci->dev, addr,
+ pci_resource_len(pci, 0));
if (!adata->acp3x_base) {
ret = -ENOMEM;
- goto release_regions;
+ goto disable_msi;
}
pci_set_master(pci);
pci_set_drvdata(pci, adata);
@@ -68,11 +69,11 @@ static int snd_acp3x_probe(struct pci_dev *pci,
switch (val) {
case I2S_MODE:
adata->res = devm_kzalloc(&pci->dev,
- sizeof(struct resource) * 2,
+ sizeof(struct resource) * 4,
GFP_KERNEL);
if (!adata->res) {
ret = -ENOMEM;
- goto unmap_mmio;
+ goto disable_msi;
}
adata->res[0].name = "acp3x_i2s_iomem";
@@ -80,40 +81,67 @@ static int snd_acp3x_probe(struct pci_dev *pci,
adata->res[0].start = addr;
adata->res[0].end = addr + (ACP3x_REG_END - ACP3x_REG_START);
- adata->res[1].name = "acp3x_i2s_irq";
- adata->res[1].flags = IORESOURCE_IRQ;
- adata->res[1].start = pci->irq;
- adata->res[1].end = pci->irq;
+ adata->res[1].name = "acp3x_i2s_sp";
+ adata->res[1].flags = IORESOURCE_MEM;
+ adata->res[1].start = addr + ACP3x_I2STDM_REG_START;
+ adata->res[1].end = addr + ACP3x_I2STDM_REG_END;
+
+ adata->res[2].name = "acp3x_i2s_bt";
+ adata->res[2].flags = IORESOURCE_MEM;
+ adata->res[2].start = addr + ACP3x_BT_TDM_REG_START;
+ adata->res[2].end = addr + ACP3x_BT_TDM_REG_END;
+
+ adata->res[3].name = "acp3x_i2s_irq";
+ adata->res[3].flags = IORESOURCE_IRQ;
+ adata->res[3].start = pci->irq;
+ adata->res[3].end = adata->res[3].start;
adata->acp3x_audio_mode = ACP3x_I2S_MODE;
memset(&pdevinfo, 0, sizeof(pdevinfo));
- pdevinfo.name = "acp3x_rv_i2s";
- pdevinfo.id = 0;
- pdevinfo.parent = &pci->dev;
- pdevinfo.num_res = 2;
- pdevinfo.res = adata->res;
- pdevinfo.data = &irqflags;
- pdevinfo.size_data = sizeof(irqflags);
-
- adata->pdev = platform_device_register_full(&pdevinfo);
- if (IS_ERR(adata->pdev)) {
- dev_err(&pci->dev, "cannot register %s device\n",
- pdevinfo.name);
- ret = PTR_ERR(adata->pdev);
- goto unmap_mmio;
+ pdevinfo[0].name = "acp3x_rv_i2s_dma";
+ pdevinfo[0].id = 0;
+ pdevinfo[0].parent = &pci->dev;
+ pdevinfo[0].num_res = 4;
+ pdevinfo[0].res = &adata->res[0];
+ pdevinfo[0].data = &irqflags;
+ pdevinfo[0].size_data = sizeof(irqflags);
+
+ pdevinfo[1].name = "acp3x_i2s_playcap";
+ pdevinfo[1].id = 0;
+ pdevinfo[1].parent = &pci->dev;
+ pdevinfo[1].num_res = 1;
+ pdevinfo[1].res = &adata->res[1];
+
+ pdevinfo[2].name = "acp3x_i2s_playcap";
+ pdevinfo[2].id = 1;
+ pdevinfo[2].parent = &pci->dev;
+ pdevinfo[2].num_res = 1;
+ pdevinfo[2].res = &adata->res[2];
+ for (i = 0; i < ACP3x_DEVS ; i++) {
+ adata->pdev[i] =
+ platform_device_register_full(&pdevinfo[i]);
+ if (IS_ERR(adata->pdev[i])) {
+ dev_err(&pci->dev, "cannot register %s device\n",
+ pdevinfo[i].name);
+ ret = PTR_ERR(adata->pdev[i]);
+ goto unregister_devs;
+ }
}
break;
default:
dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val);
ret = -ENODEV;
- goto unmap_mmio;
+ goto disable_msi;
}
return 0;
-unmap_mmio:
+unregister_devs:
+ if (val == I2S_MODE)
+ for (i = 0 ; i < ACP3x_DEVS ; i++)
+ platform_device_unregister(adata->pdev[i]);
+disable_msi:
pci_disable_msi(pci);
- iounmap(adata->acp3x_base);
release_regions:
pci_release_regions(pci);
disable_pci:
@@ -125,10 +153,12 @@ static int snd_acp3x_probe(struct pci_dev *pci,
static void snd_acp3x_remove(struct pci_dev *pci)
{
struct acp3x_dev_data *adata = pci_get_drvdata(pci);
+ int i;
- platform_device_unregister(adata->pdev);
- iounmap(adata->acp3x_base);
-
+ if (adata->acp3x_audio_mode == ACP3x_I2S_MODE) {
+ for (i = 0 ; i < ACP3x_DEVS ; i++)
+ platform_device_unregister(adata->pdev[i]);
+ }
pci_disable_msi(pci);
pci_release_regions(pci);
pci_disable_device(pci);
@@ -151,6 +181,7 @@ static struct pci_driver acp3x_driver = {
module_pci_driver(acp3x_driver);
+MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati@amd.com");
MODULE_AUTHOR("Maruthi.Bayyavarapu@amd.com");
MODULE_DESCRIPTION("AMD ACP3x PCI driver");
MODULE_LICENSE("GPL v2");
--
2.20.1
_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: max98090: Drop incorrectly applied duplicate commit" to the asoc tree
@ 2020-01-10 13:38 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-01-10 13:38 UTC (permalink / raw)
To: Mark Brown; +Cc: alsa-devel, Marek Szyprowski
The patch
ASoC: max98090: Drop incorrectly applied duplicate commit
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From a84188eced6109983af54f9435a26d21eac3f8cc Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@kernel.org>
Date: Fri, 10 Jan 2020 13:24:02 +0000
Subject: [PATCH] ASoC: max98090: Drop incorrectly applied duplicate commit
This reverts commit 4e93c1294f4b051 (ASoC: max98090: fix incorrect
helper in max98090_dapm_put_enum_double()) which was misapplied.
Reported-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/max98090.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index ba0e3ba162f8..454cb8e5b0a1 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -98,7 +98,7 @@ static int max98090_put_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component =
- snd_soc_dapm_kcontrol_component(kcontrol);
+ snd_soc_kcontrol_component(kcontrol);
struct max98090_priv *max98090 =
snd_soc_component_get_drvdata(component);
int ret;
--
2.20.1
_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: sti: fix possible sleep-in-atomic" to the asoc tree
@ 2020-01-13 15:13 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-01-13 15:13 UTC (permalink / raw)
To: Arnaud Pouliquen; +Cc: alsa-devel, Mark Brown
The patch
ASoC: sti: fix possible sleep-in-atomic
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From ce780a47c3c01e1e179d0792df6b853a913928f1 Mon Sep 17 00:00:00 2001
From: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Date: Mon, 13 Jan 2020 11:04:00 +0100
Subject: [PATCH] ASoC: sti: fix possible sleep-in-atomic
Change mutex and spinlock management to avoid sleep
in atomic issue.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Link: https://lore.kernel.org/r/20200113100400.30472-1-arnaud.pouliquen@st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sti/uniperif_player.c | 7 ++++---
1 file changed, 4 insertions(+), 3 deletions(-)
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index 48ea915b24ba..2ed92c990b97 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -226,7 +226,6 @@ static void uni_player_set_channel_status(struct uniperif *player,
* sampling frequency. If no sample rate is already specified, then
* set one.
*/
- mutex_lock(&player->ctrl_lock);
if (runtime) {
switch (runtime->rate) {
case 22050:
@@ -303,7 +302,6 @@ static void uni_player_set_channel_status(struct uniperif *player,
player->stream_settings.iec958.status[3 + (n * 4)] << 24;
SET_UNIPERIF_CHANNEL_STA_REGN(player, n, status);
}
- mutex_unlock(&player->ctrl_lock);
/* Update the channel status */
if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
@@ -365,8 +363,10 @@ static int uni_player_prepare_iec958(struct uniperif *player,
SET_UNIPERIF_CTRL_ZERO_STUFF_HW(player);
+ mutex_lock(&player->ctrl_lock);
/* Update the channel status */
uni_player_set_channel_status(player, runtime);
+ mutex_unlock(&player->ctrl_lock);
/* Clear the user validity user bits */
SET_UNIPERIF_USER_VALIDITY_VALIDITY_LR(player, 0);
@@ -598,7 +598,6 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol,
iec958->status[1] = ucontrol->value.iec958.status[1];
iec958->status[2] = ucontrol->value.iec958.status[2];
iec958->status[3] = ucontrol->value.iec958.status[3];
- mutex_unlock(&player->ctrl_lock);
spin_lock_irqsave(&player->irq_lock, flags);
if (player->substream && player->substream->runtime)
@@ -608,6 +607,8 @@ static int uni_player_ctl_iec958_put(struct snd_kcontrol *kcontrol,
uni_player_set_channel_status(player, NULL);
spin_unlock_irqrestore(&player->irq_lock, flags);
+ mutex_unlock(&player->ctrl_lock);
+
return 0;
}
--
2.20.1
_______________________________________________
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Alsa-devel@alsa-project.org
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: SOF: Intel: fix HDA codec driver probe with multiple controllers" to the asoc tree
@ 2020-01-13 15:13 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-01-13 15:13 UTC (permalink / raw)
To: Kai Vehmanen; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: SOF: Intel: fix HDA codec driver probe with multiple controllers
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 2c63bea714780f8e1fc9cb7bc10deda26fada25b Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Fri, 10 Jan 2020 17:57:50 -0600
Subject: [PATCH] ASoC: SOF: Intel: fix HDA codec driver probe with multiple
controllers
In case system has multiple HDA controllers, it can happen that
same HDA codec driver is used for codecs of multiple controllers.
In this case, SOF may fail to probe the HDA driver and SOF
initialization fails.
SOF HDA code currently relies that a call to request_module() will
also run device matching logic to attach driver to the codec instance.
However if driver for another HDA controller was already loaded and it
already loaded the HDA codec driver, this breaks current logic in SOF.
In this case the request_module() SOF does becomes a no-op and HDA
Codec driver is not attached to the codec instance sitting on the HDA
bus SOF is controlling. Typical scenario would be a system with both
external and internal GPUs, with driver of the external GPU loaded
first.
Fix this by adding similar logic as is used in legacy HDA driver
where an explicit device_attach() call is done after request_module().
Also add logic to propagate errors reported by device_attach() back
to caller. This also works in the case where drivers are not built
as modules.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200110235751.3404-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-codec.c | 19 ++++++++++++-------
1 file changed, 12 insertions(+), 7 deletions(-)
diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c
index 827f84a0722e..fbfa225d1c5a 100644
--- a/sound/soc/sof/intel/hda-codec.c
+++ b/sound/soc/sof/intel/hda-codec.c
@@ -24,19 +24,18 @@
#define IDISP_VID_INTEL 0x80860000
/* load the legacy HDA codec driver */
-#ifdef MODULE
-static void hda_codec_load_module(struct hda_codec *codec)
+static int hda_codec_load_module(struct hda_codec *codec)
{
+#ifdef MODULE
char alias[MODULE_NAME_LEN];
const char *module = alias;
snd_hdac_codec_modalias(&codec->core, alias, sizeof(alias));
dev_dbg(&codec->core.dev, "loading codec module: %s\n", module);
request_module(module);
-}
-#else
-static void hda_codec_load_module(struct hda_codec *codec) {}
#endif
+ return device_attach(hda_codec_dev(codec));
+}
/* enable controller wake up event for all codecs with jack connectors */
void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev)
@@ -129,10 +128,16 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address)
if ((mach_params && mach_params->common_hdmi_codec_drv) ||
(resp & 0xFFFF0000) != IDISP_VID_INTEL) {
hdev->type = HDA_DEV_LEGACY;
- hda_codec_load_module(&hda_priv->codec);
+ ret = hda_codec_load_module(&hda_priv->codec);
+ /*
+ * handle ret==0 (no driver bound) as an error, but pass
+ * other return codes without modification
+ */
+ if (ret == 0)
+ ret = -ENOENT;
}
- return 0;
+ return ret;
#else
hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL);
if (!hdev)
--
2.20.1
_______________________________________________
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Alsa-devel@alsa-project.org
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: rt1308-sdw: add rt1308 SdW amplifier driver" to the asoc tree
@ 2020-01-13 16:11 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-01-13 16:11 UTC (permalink / raw)
To: Shuming Fan; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: rt1308-sdw: add rt1308 SdW amplifier driver
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From a87a6653a28c0d511b3fdf3d62302bed5f325e03 Mon Sep 17 00:00:00 2001
From: Shuming Fan <shumingf@realtek.com>
Date: Fri, 10 Jan 2020 09:46:06 +0800
Subject: [PATCH] ASoC: rt1308-sdw: add rt1308 SdW amplifier driver
This is the initial amplifier driver for rt1308-sdw.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200110014606.17333-1-shumingf@realtek.com
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/Kconfig | 6 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/rt1308-sdw.c | 736 ++++++++++++++++++++++++++++++++++
sound/soc/codecs/rt1308-sdw.h | 169 ++++++++
4 files changed, 913 insertions(+)
create mode 100644 sound/soc/codecs/rt1308-sdw.c
create mode 100644 sound/soc/codecs/rt1308-sdw.h
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index cdfd912d5f8f..2f52e4cac96a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -168,6 +168,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_RT700_SDW if SOUNDWIRE
select SND_SOC_RT711_SDW if SOUNDWIRE
select SND_SOC_RT715_SDW if SOUNDWIRE
+ select SND_SOC_RT1308_SDW if SOUNDWIRE
select SND_SOC_SGTL5000 if I2C
select SND_SOC_SI476X if MFD_SI476X_CORE
select SND_SOC_SIMPLE_AMPLIFIER
@@ -1005,6 +1006,11 @@ config SND_SOC_RT1305
config SND_SOC_RT1308
tristate
+config SND_SOC_RT1308_SDW
+ tristate "Realtek RT1308 Codec - SDW"
+ depends on SOUNDWIRE
+ select REGMAP_SOUNDWIRE
+
config SND_SOC_RT5514
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index ce285b33a806..66aa452f5aa2 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -154,6 +154,7 @@ snd-soc-rl6347a-objs := rl6347a.o
snd-soc-rt1011-objs := rt1011.o
snd-soc-rt1305-objs := rt1305.o
snd-soc-rt1308-objs := rt1308.o
+snd-soc-rt1308-sdw-objs := rt1308-sdw.o
snd-soc-rt274-objs := rt274.o
snd-soc-rt286-objs := rt286.o
snd-soc-rt298-objs := rt298.o
@@ -448,6 +449,7 @@ obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o
obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o
obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o
obj-$(CONFIG_SND_SOC_RT1308) += snd-soc-rt1308.o
+obj-$(CONFIG_SND_SOC_RT1308_SDW) += snd-soc-rt1308-sdw.o
obj-$(CONFIG_SND_SOC_RT274) += snd-soc-rt274.o
obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o
obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o
diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c
new file mode 100644
index 000000000000..8a03dbfe7906
--- /dev/null
+++ b/sound/soc/codecs/rt1308-sdw.c
@@ -0,0 +1,736 @@
+// SPDX-License-Identifier: GPL-2.0
+//
+// rt1308-sdw.c -- rt1308 ALSA SoC audio driver
+//
+// Copyright(c) 2019 Realtek Semiconductor Corp.
+//
+//
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/pm_runtime.h>
+#include <linux/mod_devicetable.h>
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_type.h>
+#include <linux/soundwire/sdw_registers.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "rt1308.h"
+#include "rt1308-sdw.h"
+
+static bool rt1308_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case 0x00e0:
+ case 0x00f0:
+ case 0x2f01 ... 0x2f07:
+ case 0x3000 ... 0x3001:
+ case 0x3004 ... 0x3005:
+ case 0x3008:
+ case 0x300a:
+ case 0xc000 ... 0xcff3:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool rt1308_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case 0x2f01 ... 0x2f07:
+ case 0x3000 ... 0x3001:
+ case 0x3004 ... 0x3005:
+ case 0x3008:
+ case 0x300a:
+ case 0xc000:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct regmap_config rt1308_sdw_regmap = {
+ .reg_bits = 32,
+ .val_bits = 8,
+ .readable_reg = rt1308_readable_register,
+ .volatile_reg = rt1308_volatile_register,
+ .max_register = 0xcfff,
+ .reg_defaults = rt1308_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(rt1308_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
+ .use_single_read = true,
+ .use_single_write = true,
+};
+
+/* Bus clock frequency */
+#define RT1308_CLK_FREQ_9600000HZ 9600000
+#define RT1308_CLK_FREQ_12000000HZ 12000000
+#define RT1308_CLK_FREQ_6000000HZ 6000000
+#define RT1308_CLK_FREQ_4800000HZ 4800000
+#define RT1308_CLK_FREQ_2400000HZ 2400000
+#define RT1308_CLK_FREQ_12288000HZ 12288000
+
+static int rt1308_clock_config(struct device *dev)
+{
+ struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev);
+ unsigned int clk_freq, value;
+
+ clk_freq = (rt1308->params.curr_dr_freq >> 1);
+
+ switch (clk_freq) {
+ case RT1308_CLK_FREQ_12000000HZ:
+ value = 0x0;
+ break;
+ case RT1308_CLK_FREQ_6000000HZ:
+ value = 0x1;
+ break;
+ case RT1308_CLK_FREQ_9600000HZ:
+ value = 0x2;
+ break;
+ case RT1308_CLK_FREQ_4800000HZ:
+ value = 0x3;
+ break;
+ case RT1308_CLK_FREQ_2400000HZ:
+ value = 0x4;
+ break;
+ case RT1308_CLK_FREQ_12288000HZ:
+ value = 0x5;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ regmap_write(rt1308->regmap, 0xe0, value);
+ regmap_write(rt1308->regmap, 0xf0, value);
+
+ dev_dbg(dev, "%s complete, clk_freq=%d\n", __func__, clk_freq);
+
+ return 0;
+}
+
+static int rt1308_read_prop(struct sdw_slave *slave)
+{
+ struct sdw_slave_prop *prop = &slave->prop;
+ int nval, i, num_of_ports = 1;
+ u32 bit;
+ unsigned long addr;
+ struct sdw_dpn_prop *dpn;
+
+ prop->paging_support = true;
+
+ /* first we need to allocate memory for set bits in port lists */
+ prop->source_ports = 0x00; /* BITMAP: 00010100 (not enable yet) */
+ prop->sink_ports = 0x2; /* BITMAP: 00000010 */
+
+ /* for sink */
+ nval = hweight32(prop->sink_ports);
+ num_of_ports += nval;
+ prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval,
+ sizeof(*prop->sink_dpn_prop),
+ GFP_KERNEL);
+ if (!prop->sink_dpn_prop)
+ return -ENOMEM;
+
+ i = 0;
+ dpn = prop->sink_dpn_prop;
+ addr = prop->sink_ports;
+ for_each_set_bit(bit, &addr, 32) {
+ dpn[i].num = bit;
+ dpn[i].type = SDW_DPN_FULL;
+ dpn[i].simple_ch_prep_sm = true;
+ dpn[i].ch_prep_timeout = 10;
+ i++;
+ }
+
+ /* Allocate port_ready based on num_of_ports */
+ slave->port_ready = devm_kcalloc(&slave->dev, num_of_ports,
+ sizeof(*slave->port_ready),
+ GFP_KERNEL);
+ if (!slave->port_ready)
+ return -ENOMEM;
+
+ /* Initialize completion */
+ for (i = 0; i < num_of_ports; i++)
+ init_completion(&slave->port_ready[i]);
+
+ /* set the timeout values */
+ prop->clk_stop_timeout = 20;
+
+ dev_dbg(&slave->dev, "%s\n", __func__);
+
+ return 0;
+}
+
+static int rt1308_io_init(struct device *dev, struct sdw_slave *slave)
+{
+ struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev);
+ int ret = 0;
+ unsigned int efuse_m_btl_l, efuse_m_btl_r, tmp;
+ unsigned int efuse_c_btl_l, efuse_c_btl_r;
+
+ if (rt1308->hw_init)
+ return 0;
+
+ ret = rt1308_read_prop(slave);
+ if (ret < 0)
+ goto _io_init_err_;
+
+ if (rt1308->first_hw_init) {
+ regcache_cache_only(rt1308->regmap, false);
+ regcache_cache_bypass(rt1308->regmap, true);
+ }
+
+ /*
+ * PM runtime is only enabled when a Slave reports as Attached
+ */
+ if (!rt1308->first_hw_init) {
+ /* set autosuspend parameters */
+ pm_runtime_set_autosuspend_delay(&slave->dev, 3000);
+ pm_runtime_use_autosuspend(&slave->dev);
+
+ /* update count of parent 'active' children */
+ pm_runtime_set_active(&slave->dev);
+
+ /* make sure the device does not suspend immediately */
+ pm_runtime_mark_last_busy(&slave->dev);
+
+ pm_runtime_enable(&slave->dev);
+ }
+
+ pm_runtime_get_noresume(&slave->dev);
+
+ /* sw reset */
+ regmap_write(rt1308->regmap, RT1308_SDW_RESET, 0);
+
+ /* read efuse */
+ regmap_write(rt1308->regmap, 0xc360, 0x01);
+ regmap_write(rt1308->regmap, 0xc361, 0x80);
+ regmap_write(rt1308->regmap, 0xc7f0, 0x04);
+ regmap_write(rt1308->regmap, 0xc7f1, 0xfe);
+ msleep(100);
+ regmap_write(rt1308->regmap, 0xc7f0, 0x44);
+ msleep(20);
+ regmap_write(rt1308->regmap, 0xc240, 0x10);
+
+ regmap_read(rt1308->regmap, 0xc861, &tmp);
+ efuse_m_btl_l = tmp;
+ regmap_read(rt1308->regmap, 0xc860, &tmp);
+ efuse_m_btl_l = efuse_m_btl_l | (tmp << 8);
+ regmap_read(rt1308->regmap, 0xc863, &tmp);
+ efuse_c_btl_l = tmp;
+ regmap_read(rt1308->regmap, 0xc862, &tmp);
+ efuse_c_btl_l = efuse_c_btl_l | (tmp << 8);
+ regmap_read(rt1308->regmap, 0xc871, &tmp);
+ efuse_m_btl_r = tmp;
+ regmap_read(rt1308->regmap, 0xc870, &tmp);
+ efuse_m_btl_r = efuse_m_btl_r | (tmp << 8);
+ regmap_read(rt1308->regmap, 0xc873, &tmp);
+ efuse_c_btl_r = tmp;
+ regmap_read(rt1308->regmap, 0xc872, &tmp);
+ efuse_c_btl_r = efuse_c_btl_r | (tmp << 8);
+ dev_info(&slave->dev, "%s m_btl_l=0x%x, m_btl_r=0x%x\n", __func__,
+ efuse_m_btl_l, efuse_m_btl_r);
+ dev_info(&slave->dev, "%s c_btl_l=0x%x, c_btl_r=0x%x\n", __func__,
+ efuse_c_btl_l, efuse_c_btl_r);
+
+ /* initial settings */
+ regmap_write(rt1308->regmap, 0xc103, 0xc0);
+ regmap_write(rt1308->regmap, 0xc030, 0x17);
+ regmap_write(rt1308->regmap, 0xc031, 0x81);
+ regmap_write(rt1308->regmap, 0xc032, 0x26);
+ regmap_write(rt1308->regmap, 0xc040, 0x80);
+ regmap_write(rt1308->regmap, 0xc041, 0x80);
+ regmap_write(rt1308->regmap, 0xc042, 0x06);
+ regmap_write(rt1308->regmap, 0xc052, 0x0a);
+ regmap_write(rt1308->regmap, 0xc080, 0x0a);
+ regmap_write(rt1308->regmap, 0xc060, 0x02);
+ regmap_write(rt1308->regmap, 0xc061, 0x75);
+ regmap_write(rt1308->regmap, 0xc062, 0x05);
+ regmap_write(rt1308->regmap, 0xc171, 0x07);
+ regmap_write(rt1308->regmap, 0xc173, 0x0d);
+ regmap_write(rt1308->regmap, 0xc311, 0x7f);
+ regmap_write(rt1308->regmap, 0xc900, 0x90);
+ regmap_write(rt1308->regmap, 0xc1a0, 0x84);
+ regmap_write(rt1308->regmap, 0xc1a1, 0x01);
+ regmap_write(rt1308->regmap, 0xc360, 0x78);
+ regmap_write(rt1308->regmap, 0xc361, 0x87);
+ regmap_write(rt1308->regmap, 0xc0a1, 0x71);
+ regmap_write(rt1308->regmap, 0xc210, 0x00);
+ regmap_write(rt1308->regmap, 0xc070, 0x00);
+ regmap_write(rt1308->regmap, 0xc100, 0xd7);
+ regmap_write(rt1308->regmap, 0xc101, 0xd7);
+ regmap_write(rt1308->regmap, 0xc300, 0x09);
+
+ if (rt1308->first_hw_init) {
+ regcache_cache_bypass(rt1308->regmap, false);
+ regcache_mark_dirty(rt1308->regmap);
+ } else
+ rt1308->first_hw_init = true;
+
+ /* Mark Slave initialization complete */
+ rt1308->hw_init = true;
+
+ pm_runtime_mark_last_busy(&slave->dev);
+ pm_runtime_put_autosuspend(&slave->dev);
+
+ dev_dbg(&slave->dev, "%s hw_init complete\n", __func__);
+
+_io_init_err_:
+ return ret;
+}
+
+static int rt1308_update_status(struct sdw_slave *slave,
+ enum sdw_slave_status status)
+{
+ struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(&slave->dev);
+
+ /* Update the status */
+ rt1308->status = status;
+
+ if (status == SDW_SLAVE_UNATTACHED)
+ rt1308->hw_init = false;
+
+ /*
+ * Perform initialization only if slave status is present and
+ * hw_init flag is false
+ */
+ if (rt1308->hw_init || rt1308->status != SDW_SLAVE_ATTACHED)
+ return 0;
+
+ /* perform I/O transfers required for Slave initialization */
+ return rt1308_io_init(&slave->dev, slave);
+}
+
+static int rt1308_bus_config(struct sdw_slave *slave,
+ struct sdw_bus_params *params)
+{
+ struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(&slave->dev);
+ int ret;
+
+ memcpy(&rt1308->params, params, sizeof(*params));
+
+ ret = rt1308_clock_config(&slave->dev);
+ if (ret < 0)
+ dev_err(&slave->dev, "Invalid clk config");
+
+ return ret;
+}
+
+static int rt1308_interrupt_callback(struct sdw_slave *slave,
+ struct sdw_slave_intr_status *status)
+{
+ dev_dbg(&slave->dev,
+ "%s control_port_stat=%x", __func__, status->control_port);
+
+ return 0;
+}
+
+static int rt1308_classd_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *component =
+ snd_soc_dapm_to_component(w->dapm);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ msleep(30);
+ snd_soc_component_update_bits(component,
+ RT1308_SDW_OFFSET | (RT1308_POWER_STATUS << 4),
+ 0x3, 0x3);
+ msleep(40);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_component_update_bits(component,
+ RT1308_SDW_OFFSET | (RT1308_POWER_STATUS << 4),
+ 0x3, 0);
+ usleep_range(150000, 200000);
+ break;
+
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static const char * const rt1308_rx_data_ch_select[] = {
+ "LR",
+ "LL",
+ "RL",
+ "RR",
+};
+
+static SOC_ENUM_SINGLE_DECL(rt1308_rx_data_ch_enum,
+ RT1308_SDW_OFFSET | (RT1308_DATA_PATH << 4), 0,
+ rt1308_rx_data_ch_select);
+
+static const struct snd_kcontrol_new rt1308_snd_controls[] = {
+
+ /* I2S Data Channel Selection */
+ SOC_ENUM("RX Channel Select", rt1308_rx_data_ch_enum),
+};
+
+static const struct snd_kcontrol_new rt1308_sto_dac_l =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch",
+ RT1308_SDW_OFFSET_BYTE3 | (RT1308_DAC_SET << 4),
+ RT1308_DVOL_MUTE_L_EN_SFT, 1, 1);
+
+static const struct snd_kcontrol_new rt1308_sto_dac_r =
+ SOC_DAPM_SINGLE_AUTODISABLE("Switch",
+ RT1308_SDW_OFFSET_BYTE3 | (RT1308_DAC_SET << 4),
+ RT1308_DVOL_MUTE_R_EN_SFT, 1, 1);
+
+static const struct snd_soc_dapm_widget rt1308_dapm_widgets[] = {
+ /* Audio Interface */
+ SND_SOC_DAPM_AIF_IN("AIF1RX", "DP1 Playback", 0, SND_SOC_NOPM, 0, 0),
+
+ /* Supply Widgets */
+ SND_SOC_DAPM_SUPPLY("MBIAS20U",
+ RT1308_SDW_OFFSET | (RT1308_POWER << 4), 7, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ALDO",
+ RT1308_SDW_OFFSET | (RT1308_POWER << 4), 6, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DBG",
+ RT1308_SDW_OFFSET | (RT1308_POWER << 4), 5, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DACL",
+ RT1308_SDW_OFFSET | (RT1308_POWER << 4), 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("CLK25M",
+ RT1308_SDW_OFFSET | (RT1308_POWER << 4), 2, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC_R",
+ RT1308_SDW_OFFSET | (RT1308_POWER << 4), 1, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("ADC_L",
+ RT1308_SDW_OFFSET | (RT1308_POWER << 4), 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("DAC Power",
+ RT1308_SDW_OFFSET | (RT1308_POWER << 4), 3, 0, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("DLDO",
+ RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 5, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("VREF",
+ RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MIXER_R",
+ RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 2, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MIXER_L",
+ RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 1, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MBIAS4U",
+ RT1308_SDW_OFFSET_BYTE1 | (RT1308_POWER << 4), 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_SUPPLY("PLL2_LDO",
+ RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLL2B",
+ RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 3, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLL2F",
+ RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 2, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLL2F2",
+ RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 1, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("PLL2B2",
+ RT1308_SDW_OFFSET_BYTE2 | (RT1308_POWER << 4), 0, 0, NULL, 0),
+
+ /* Digital Interface */
+ SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_SWITCH("DAC L", SND_SOC_NOPM, 0, 0, &rt1308_sto_dac_l),
+ SND_SOC_DAPM_SWITCH("DAC R", SND_SOC_NOPM, 0, 0, &rt1308_sto_dac_r),
+
+ /* Output Lines */
+ SND_SOC_DAPM_PGA_E("CLASS D", SND_SOC_NOPM, 0, 0, NULL, 0,
+ rt1308_classd_event,
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_OUTPUT("SPOL"),
+ SND_SOC_DAPM_OUTPUT("SPOR"),
+};
+
+static const struct snd_soc_dapm_route rt1308_dapm_routes[] = {
+
+ { "DAC", NULL, "AIF1RX" },
+
+ { "DAC", NULL, "MBIAS20U" },
+ { "DAC", NULL, "ALDO" },
+ { "DAC", NULL, "DBG" },
+ { "DAC", NULL, "DACL" },
+ { "DAC", NULL, "CLK25M" },
+ { "DAC", NULL, "ADC_R" },
+ { "DAC", NULL, "ADC_L" },
+ { "DAC", NULL, "DLDO" },
+ { "DAC", NULL, "VREF" },
+ { "DAC", NULL, "MIXER_R" },
+ { "DAC", NULL, "MIXER_L" },
+ { "DAC", NULL, "MBIAS4U" },
+ { "DAC", NULL, "PLL2_LDO" },
+ { "DAC", NULL, "PLL2B" },
+ { "DAC", NULL, "PLL2F" },
+ { "DAC", NULL, "PLL2F2" },
+ { "DAC", NULL, "PLL2B2" },
+
+ { "DAC L", "Switch", "DAC" },
+ { "DAC R", "Switch", "DAC" },
+ { "DAC L", NULL, "DAC Power" },
+ { "DAC R", NULL, "DAC Power" },
+
+ { "CLASS D", NULL, "DAC L" },
+ { "CLASS D", NULL, "DAC R" },
+ { "SPOL", NULL, "CLASS D" },
+ { "SPOR", NULL, "CLASS D" },
+};
+
+static int rt1308_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream,
+ int direction)
+{
+ struct sdw_stream_data *stream;
+
+ stream = kzalloc(sizeof(*stream), GFP_KERNEL);
+ if (!stream)
+ return -ENOMEM;
+
+ stream->sdw_stream = (struct sdw_stream_runtime *)sdw_stream;
+
+ /* Use tx_mask or rx_mask to configure stream tag and set dma_data */
+ if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+ dai->playback_dma_data = stream;
+ else
+ dai->capture_dma_data = stream;
+
+ return 0;
+}
+
+static void rt1308_sdw_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct sdw_stream_data *stream;
+
+ stream = snd_soc_dai_get_dma_data(dai, substream);
+ snd_soc_dai_set_dma_data(dai, substream, NULL);
+ kfree(stream);
+}
+
+static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct rt1308_sdw_priv *rt1308 =
+ snd_soc_component_get_drvdata(component);
+ struct sdw_stream_config stream_config;
+ struct sdw_port_config port_config;
+ enum sdw_data_direction direction;
+ struct sdw_stream_data *stream;
+ int retval, port, num_channels;
+
+ dev_dbg(dai->dev, "%s %s", __func__, dai->name);
+ stream = snd_soc_dai_get_dma_data(dai, substream);
+
+ if (!stream)
+ return -EINVAL;
+
+ if (!rt1308->sdw_slave)
+ return -EINVAL;
+
+ /* SoundWire specific configuration */
+ /* port 1 for playback */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ direction = SDW_DATA_DIR_RX;
+ port = 1;
+ } else {
+ return -EINVAL;
+ }
+
+ stream_config.frame_rate = params_rate(params);
+ stream_config.ch_count = params_channels(params);
+ stream_config.bps = snd_pcm_format_width(params_format(params));
+ stream_config.direction = direction;
+
+ num_channels = params_channels(params);
+ port_config.ch_mask = (1 << (num_channels)) - 1;
+ port_config.num = port;
+
+ retval = sdw_stream_add_slave(rt1308->sdw_slave, &stream_config,
+ &port_config, 1, stream->sdw_stream);
+ if (retval) {
+ dev_err(dai->dev, "Unable to configure port\n");
+ return retval;
+ }
+
+ return retval;
+}
+
+static int rt1308_sdw_pcm_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct rt1308_sdw_priv *rt1308 =
+ snd_soc_component_get_drvdata(component);
+ struct sdw_stream_data *stream =
+ snd_soc_dai_get_dma_data(dai, substream);
+
+ if (!rt1308->sdw_slave)
+ return -EINVAL;
+
+ sdw_stream_remove_slave(rt1308->sdw_slave, stream->sdw_stream);
+ return 0;
+}
+
+/*
+ * slave_ops: callbacks for get_clock_stop_mode, clock_stop and
+ * port_prep are not defined for now
+ */
+static struct sdw_slave_ops rt1308_slave_ops = {
+ .read_prop = rt1308_read_prop,
+ .interrupt_callback = rt1308_interrupt_callback,
+ .update_status = rt1308_update_status,
+ .bus_config = rt1308_bus_config,
+};
+
+static const struct snd_soc_component_driver soc_component_sdw_rt1308 = {
+ .controls = rt1308_snd_controls,
+ .num_controls = ARRAY_SIZE(rt1308_snd_controls),
+ .dapm_widgets = rt1308_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(rt1308_dapm_widgets),
+ .dapm_routes = rt1308_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(rt1308_dapm_routes),
+};
+
+static const struct snd_soc_dai_ops rt1308_aif_dai_ops = {
+ .hw_params = rt1308_sdw_hw_params,
+ .hw_free = rt1308_sdw_pcm_hw_free,
+ .set_sdw_stream = rt1308_set_sdw_stream,
+ .shutdown = rt1308_sdw_shutdown,
+};
+
+#define RT1308_STEREO_RATES SNDRV_PCM_RATE_48000
+#define RT1308_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_driver rt1308_sdw_dai[] = {
+ {
+ .name = "rt1308-aif",
+ .playback = {
+ .stream_name = "DP1 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = RT1308_STEREO_RATES,
+ .formats = RT1308_FORMATS,
+ },
+ .ops = &rt1308_aif_dai_ops,
+ },
+};
+
+static int rt1308_sdw_init(struct device *dev, struct regmap *regmap,
+ struct sdw_slave *slave)
+{
+ struct rt1308_sdw_priv *rt1308;
+ int ret;
+
+ rt1308 = devm_kzalloc(dev, sizeof(*rt1308), GFP_KERNEL);
+ if (!rt1308)
+ return -ENOMEM;
+
+ dev_set_drvdata(dev, rt1308);
+ rt1308->sdw_slave = slave;
+ rt1308->regmap = regmap;
+
+ /*
+ * Mark hw_init to false
+ * HW init will be performed when device reports present
+ */
+ rt1308->hw_init = false;
+ rt1308->first_hw_init = false;
+
+ ret = devm_snd_soc_register_component(dev,
+ &soc_component_sdw_rt1308,
+ rt1308_sdw_dai,
+ ARRAY_SIZE(rt1308_sdw_dai));
+
+ dev_dbg(&slave->dev, "%s\n", __func__);
+
+ return ret;
+}
+
+static int rt1308_sdw_probe(struct sdw_slave *slave,
+ const struct sdw_device_id *id)
+{
+ struct regmap *regmap;
+
+ /* Assign ops */
+ slave->ops = &rt1308_slave_ops;
+
+ /* Regmap Initialization */
+ regmap = devm_regmap_init_sdw(slave, &rt1308_sdw_regmap);
+ if (!regmap)
+ return -EINVAL;
+
+ rt1308_sdw_init(&slave->dev, regmap, slave);
+
+ return 0;
+}
+
+static const struct sdw_device_id rt1308_id[] = {
+ SDW_SLAVE_ENTRY(0x025d, 0x1308, 0),
+ {},
+};
+MODULE_DEVICE_TABLE(sdw, rt1308_id);
+
+static int rt1308_dev_suspend(struct device *dev)
+{
+ struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev);
+
+ if (!rt1308->hw_init)
+ return 0;
+
+ regcache_cache_only(rt1308->regmap, true);
+
+ return 0;
+}
+
+#define RT1308_PROBE_TIMEOUT 2000
+
+static int rt1308_dev_resume(struct device *dev)
+{
+ struct sdw_slave *slave = dev_to_sdw_dev(dev);
+ struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev);
+ unsigned long time;
+
+ if (!rt1308->hw_init)
+ return 0;
+
+ if (!slave->unattach_request)
+ goto regmap_sync;
+
+ time = wait_for_completion_timeout(&slave->initialization_complete,
+ msecs_to_jiffies(RT1308_PROBE_TIMEOUT));
+ if (!time) {
+ dev_err(&slave->dev, "Initialization not complete, timed out\n");
+ return -ETIMEDOUT;
+ }
+
+regmap_sync:
+ slave->unattach_request = 0;
+ regcache_cache_only(rt1308->regmap, false);
+ regcache_sync_region(rt1308->regmap, 0xc000, 0xcfff);
+
+ return 0;
+}
+
+static const struct dev_pm_ops rt1308_pm = {
+ SET_SYSTEM_SLEEP_PM_OPS(rt1308_dev_suspend, rt1308_dev_resume)
+ SET_RUNTIME_PM_OPS(rt1308_dev_suspend, rt1308_dev_resume, NULL)
+};
+
+static struct sdw_driver rt1308_sdw_driver = {
+ .driver = {
+ .name = "rt1308",
+ .owner = THIS_MODULE,
+ .pm = &rt1308_pm,
+ },
+ .probe = rt1308_sdw_probe,
+ .ops = &rt1308_slave_ops,
+ .id_table = rt1308_id,
+};
+module_sdw_driver(rt1308_sdw_driver);
+
+MODULE_DESCRIPTION("ASoC RT1308 driver SDW");
+MODULE_AUTHOR("Shuming Fan <shumingf@realtek.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h
new file mode 100644
index 000000000000..c9341e70d6cf
--- /dev/null
+++ b/sound/soc/codecs/rt1308-sdw.h
@@ -0,0 +1,169 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * rt1308-sdw.h -- RT1308 ALSA SoC audio driver header
+ *
+ * Copyright(c) 2019 Realtek Semiconductor Corp.
+ */
+
+#ifndef __RT1308_SDW_H__
+#define __RT1308_SDW_H__
+
+static const struct reg_default rt1308_reg_defaults[] = {
+ { 0x0000, 0x00 },
+ { 0x0001, 0x00 },
+ { 0x0002, 0x00 },
+ { 0x0003, 0x00 },
+ { 0x0004, 0x00 },
+ { 0x0005, 0x01 },
+ { 0x0020, 0x00 },
+ { 0x0022, 0x00 },
+ { 0x0023, 0x00 },
+ { 0x0024, 0x00 },
+ { 0x0025, 0x00 },
+ { 0x0026, 0x00 },
+ { 0x0030, 0x00 },
+ { 0x0032, 0x00 },
+ { 0x0033, 0x00 },
+ { 0x0034, 0x00 },
+ { 0x0035, 0x00 },
+ { 0x0036, 0x00 },
+ { 0x0040, 0x00 },
+ { 0x0041, 0x00 },
+ { 0x0042, 0x00 },
+ { 0x0043, 0x00 },
+ { 0x0044, 0x20 },
+ { 0x0045, 0x01 },
+ { 0x0046, 0x01 },
+ { 0x0048, 0x00 },
+ { 0x0049, 0x00 },
+ { 0x0050, 0x20 },
+ { 0x0051, 0x02 },
+ { 0x0052, 0x5D },
+ { 0x0053, 0x13 },
+ { 0x0054, 0x08 },
+ { 0x0055, 0x00 },
+ { 0x0060, 0x00 },
+ { 0x0070, 0x00 },
+ { 0x00E0, 0x00 },
+ { 0x00F0, 0x00 },
+ { 0x0100, 0x00 },
+ { 0x0101, 0x00 },
+ { 0x0102, 0x20 },
+ { 0x0103, 0x00 },
+ { 0x0104, 0x00 },
+ { 0x0105, 0x03 },
+ { 0x0120, 0x00 },
+ { 0x0122, 0x00 },
+ { 0x0123, 0x00 },
+ { 0x0124, 0x00 },
+ { 0x0125, 0x00 },
+ { 0x0126, 0x00 },
+ { 0x0127, 0x00 },
+ { 0x0130, 0x00 },
+ { 0x0132, 0x00 },
+ { 0x0133, 0x00 },
+ { 0x0134, 0x00 },
+ { 0x0135, 0x00 },
+ { 0x0136, 0x00 },
+ { 0x0137, 0x00 },
+ { 0x0200, 0x00 },
+ { 0x0201, 0x00 },
+ { 0x0202, 0x00 },
+ { 0x0203, 0x00 },
+ { 0x0204, 0x00 },
+ { 0x0205, 0x03 },
+ { 0x0220, 0x00 },
+ { 0x0222, 0x00 },
+ { 0x0223, 0x00 },
+ { 0x0224, 0x00 },
+ { 0x0225, 0x00 },
+ { 0x0226, 0x00 },
+ { 0x0227, 0x00 },
+ { 0x0230, 0x00 },
+ { 0x0232, 0x00 },
+ { 0x0233, 0x00 },
+ { 0x0234, 0x00 },
+ { 0x0235, 0x00 },
+ { 0x0236, 0x00 },
+ { 0x0237, 0x00 },
+ { 0x0400, 0x00 },
+ { 0x0401, 0x00 },
+ { 0x0402, 0x00 },
+ { 0x0403, 0x00 },
+ { 0x0404, 0x00 },
+ { 0x0405, 0x03 },
+ { 0x0420, 0x00 },
+ { 0x0422, 0x00 },
+ { 0x0423, 0x00 },
+ { 0x0424, 0x00 },
+ { 0x0425, 0x00 },
+ { 0x0426, 0x00 },
+ { 0x0427, 0x00 },
+ { 0x0430, 0x00 },
+ { 0x0432, 0x00 },
+ { 0x0433, 0x00 },
+ { 0x0434, 0x00 },
+ { 0x0435, 0x00 },
+ { 0x0436, 0x00 },
+ { 0x0437, 0x00 },
+ { 0x0f00, 0x00 },
+ { 0x0f01, 0x00 },
+ { 0x0f02, 0x00 },
+ { 0x0f03, 0x00 },
+ { 0x0f04, 0x00 },
+ { 0x0f05, 0x00 },
+ { 0x0f20, 0x00 },
+ { 0x0f22, 0x00 },
+ { 0x0f23, 0x00 },
+ { 0x0f24, 0x00 },
+ { 0x0f25, 0x00 },
+ { 0x0f26, 0x00 },
+ { 0x0f27, 0x00 },
+ { 0x0f30, 0x00 },
+ { 0x0f32, 0x00 },
+ { 0x0f33, 0x00 },
+ { 0x0f34, 0x00 },
+ { 0x0f35, 0x00 },
+ { 0x0f36, 0x00 },
+ { 0x0f37, 0x00 },
+ { 0x2f01, 0x01 },
+ { 0x2f02, 0x09 },
+ { 0x2f03, 0x00 },
+ { 0x2f04, 0x0f },
+ { 0x2f05, 0x0b },
+ { 0x2f06, 0x01 },
+ { 0x2f07, 0x8e },
+ { 0x3000, 0x00 },
+ { 0x3001, 0x00 },
+ { 0x3004, 0x01 },
+ { 0x3005, 0x23 },
+ { 0x3008, 0x02 },
+ { 0x300a, 0x00 },
+ { 0xc003 | (RT1308_DAC_SET << 4), 0x00 },
+ { 0xc001 | (RT1308_POWER << 4), 0x00 },
+ { 0xc002 | (RT1308_POWER << 4), 0x00 },
+};
+
+#define RT1308_SDW_OFFSET 0xc000
+#define RT1308_SDW_OFFSET_BYTE0 0xc000
+#define RT1308_SDW_OFFSET_BYTE1 0xc001
+#define RT1308_SDW_OFFSET_BYTE2 0xc002
+#define RT1308_SDW_OFFSET_BYTE3 0xc003
+
+#define RT1308_SDW_RESET (RT1308_SDW_OFFSET | (RT1308_RESET << 4))
+
+struct rt1308_sdw_priv {
+ struct snd_soc_component *component;
+ struct regmap *regmap;
+ struct sdw_slave *sdw_slave;
+ enum sdw_slave_status status;
+ struct sdw_bus_params params;
+ bool hw_init;
+ bool first_hw_init;
+};
+
+struct sdw_stream_data {
+ struct sdw_stream_runtime *sdw_stream;
+};
+
+#endif /* __RT1308_SDW_H__ */
--
2.20.1
_______________________________________________
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Alsa-devel@alsa-project.org
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m" to the asoc tree
@ 2020-01-13 16:11 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-01-13 16:11 UTC (permalink / raw)
To: Chen Zhou; +Cc: Hulk Robot, alsa-devel, Mark Brown
The patch
ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 8fea78029f5e6ed734ae1957bef23cfda1af4354 Mon Sep 17 00:00:00 2001
From: Chen Zhou <chenzhou10@huawei.com>
Date: Mon, 13 Jan 2020 21:32:42 +0800
Subject: [PATCH] ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m
If CONFIG_SND_ATMEL_SOC_DMA=m, build error:
sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio':
(.text+0x7cd): undefined reference to `atmel_pcm_dma_platform_register'
Function atmel_pcm_dma_platform_register is defined under
CONFIG SND_ATMEL_SOC_DMA, so select SND_ATMEL_SOC_DMA in
CONFIG SND_ATMEL_SOC_SSC, same to CONFIG_SND_ATMEL_SOC_PDC.
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Chen Zhou <chenzhou10@huawei.com>
Link: https://lore.kernel.org/r/20200113133242.144550-1-chenzhou10@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/atmel/Kconfig | 2 ++
1 file changed, 2 insertions(+)
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index f118c229ed82..d1dc8e6366dc 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -19,6 +19,8 @@ config SND_ATMEL_SOC_DMA
config SND_ATMEL_SOC_SSC
tristate
+ select SND_ATMEL_SOC_DMA
+ select SND_ATMEL_SOC_PDC
config SND_ATMEL_SOC_SSC_PDC
tristate "SoC PCM DAI support for AT91 SSC controller using PDC"
--
2.20.1
_______________________________________________
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Alsa-devel@alsa-project.org
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "regulator fix for "regulator: core: Add regulator_is_equal() helper"" to the asoc tree
@ 2020-01-15 14:12 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-01-15 14:12 UTC (permalink / raw)
To: Stephen Rothwell; +Cc: Marek Vasut, alsa-devel, Mark Brown
The patch
regulator fix for "regulator: core: Add regulator_is_equal() helper"
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 0468e667a5bead9c1b7ded92861b5a98d8d78745 Mon Sep 17 00:00:00 2001
From: Stephen Rothwell <sfr@canb.auug.org.au>
Date: Wed, 15 Jan 2020 12:02:58 +1100
Subject: [PATCH] regulator fix for "regulator: core: Add regulator_is_equal()
helper"
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Link: https://lore.kernel.org/r/20200115120258.0e535fcb@canb.auug.org.au
Acked-by: Marek Vasut <marex@denx.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/linux/regulator/consumer.h | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/include/linux/regulator/consumer.h b/include/linux/regulator/consumer.h
index 2c89d886595c..6a92fd3105a3 100644
--- a/include/linux/regulator/consumer.h
+++ b/include/linux/regulator/consumer.h
@@ -596,7 +596,7 @@ regulator_bulk_set_supply_names(struct regulator_bulk_data *consumers,
}
static inline bool
-regulator_is_equal(struct regulator *reg1, struct regulator *reg2);
+regulator_is_equal(struct regulator *reg1, struct regulator *reg2)
{
return false;
}
--
2.20.1
_______________________________________________
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Alsa-devel@alsa-project.org
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: mediatek: mt8183-da7219: apply some refactors" to the asoc tree
@ 2020-02-13 13:32 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-13 13:32 UTC (permalink / raw)
To: Tzung-Bi Shih; +Cc: alsa-devel, Mark Brown
The patch
ASoC: mediatek: mt8183-da7219: apply some refactors
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From ff0035e4c22371a29e4e0d4a07cdce5726fe50aa Mon Sep 17 00:00:00 2001
From: Tzung-Bi Shih <tzungbi@google.com>
Date: Thu, 13 Feb 2020 11:27:28 +0800
Subject: [PATCH] ASoC: mediatek: mt8183-da7219: apply some refactors
1. Moves headset jack to card-specific storage.
2. Removes trailing blank line.
3. Moves card registration to the end of probe.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200213112003.4.Ia542007f51d3de753a9e0a83135ee074581dbf71@changeid
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../mediatek/mt8183/mt8183-da7219-max98357.c | 19 ++++++-------------
1 file changed, 6 insertions(+), 13 deletions(-)
diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
index c7f766f24e44..c0c85972cfb7 100644
--- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
@@ -16,8 +16,6 @@
#include "../../codecs/da7219-aad.h"
#include "../../codecs/da7219.h"
-static struct snd_soc_jack headset_jack;
-
enum PINCTRL_PIN_STATE {
PIN_STATE_DEFAULT = 0,
PIN_TDM_OUT_ON,
@@ -32,6 +30,7 @@ static const char * const mt8183_pin_str[PIN_STATE_MAX] = {
struct mt8183_da7219_max98357_priv {
struct pinctrl *pinctrl;
struct pinctrl_state *pin_states[PIN_STATE_MAX];
+ struct snd_soc_jack headset_jack;
};
static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream,
@@ -510,6 +509,8 @@ static int
mt8183_da7219_max98357_headset_init(struct snd_soc_component *component)
{
int ret;
+ struct mt8183_da7219_max98357_priv *priv =
+ snd_soc_card_get_drvdata(component->card);
/* Enable Headset and 4 Buttons Jack detection */
ret = snd_soc_card_jack_new(&mt8183_da7219_max98357_card,
@@ -517,12 +518,12 @@ mt8183_da7219_max98357_headset_init(struct snd_soc_component *component)
SND_JACK_HEADSET |
SND_JACK_BTN_0 | SND_JACK_BTN_1 |
SND_JACK_BTN_2 | SND_JACK_BTN_3,
- &headset_jack,
+ &priv->headset_jack,
NULL, 0);
if (ret)
return ret;
- da7219_aad_jack_det(component, &headset_jack);
+ da7219_aad_jack_det(component, &priv->headset_jack);
return ret;
}
@@ -559,13 +560,6 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev)
return -EINVAL;
}
- ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret) {
- dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
- __func__, ret);
- return ret;
- }
-
priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
@@ -607,7 +601,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev)
__func__, ret);
}
- return ret;
+ return devm_snd_soc_register_card(&pdev->dev, card);
}
#ifdef CONFIG_OF
@@ -634,4 +628,3 @@ MODULE_DESCRIPTION("MT8183-DA7219-MAX98357 ALSA SoC machine driver");
MODULE_AUTHOR("Shunli Wang <shunli.wang@mediatek.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("mt8183_da7219_max98357 soc card");
-
--
2.20.1
_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: mediatek: mt8183-da7219: support TDM out and 8ch I2S out" to the asoc tree
@ 2020-02-13 13:32 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-13 13:32 UTC (permalink / raw)
To: Tzung-Bi Shih; +Cc: alsa-devel, Mark Brown
The patch
ASoC: mediatek: mt8183-da7219: support TDM out and 8ch I2S out
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 195a6431710543ace009ff8e32dcf1e087506199 Mon Sep 17 00:00:00 2001
From: Tzung-Bi Shih <tzungbi@google.com>
Date: Thu, 13 Feb 2020 11:27:27 +0800
Subject: [PATCH] ASoC: mediatek: mt8183-da7219: support TDM out and 8ch I2S
out
Supports TDM out and 8ch I2S out.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200213112003.3.I30f0b8c87d5ec2a0e5f1b0fabf0a8ccef374f5ea@changeid
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 3 +++
1 file changed, 3 insertions(+)
diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
index d7685916a5cb..c7f766f24e44 100644
--- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
@@ -450,6 +450,9 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = {
{
.name = "TDM",
.no_pcm = 1,
+ .dai_fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_IB_IF |
+ SND_SOC_DAIFMT_CBM_CFM,
.dpcm_playback = 1,
.ignore_suspend = 1,
.be_hw_params_fixup = mt8183_i2s_hw_params_fixup,
--
2.20.1
_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed" to the asoc tree
@ 2020-02-13 13:32 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-13 13:32 UTC (permalink / raw)
To: Tzung-Bi Shih; +Cc: alsa-devel, Mark Brown
The patch
ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 8726ee6148fe24e2b29d4a961ad95c4ff8025d1d Mon Sep 17 00:00:00 2001
From: Tzung-Bi Shih <tzungbi@google.com>
Date: Thu, 13 Feb 2020 11:27:26 +0800
Subject: [PATCH] ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when
probed
1. Switch TDM GPIO pins according to playback on or off.
2. Pull TDM GPIO pins down when probed to avoid current leakage.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200213112003.2.I1d568b0c99742c6e755d051aadfd52e4be3cc0a5@changeid
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../mediatek/mt8183/mt8183-da7219-max98357.c | 104 +++++++++++++++++-
1 file changed, 98 insertions(+), 6 deletions(-)
diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
index b52ffed882a7..d7685916a5cb 100644
--- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
@@ -18,6 +18,22 @@
static struct snd_soc_jack headset_jack;
+enum PINCTRL_PIN_STATE {
+ PIN_STATE_DEFAULT = 0,
+ PIN_TDM_OUT_ON,
+ PIN_TDM_OUT_OFF,
+ PIN_STATE_MAX
+};
+
+static const char * const mt8183_pin_str[PIN_STATE_MAX] = {
+ "default", "aud_tdm_out_on", "aud_tdm_out_off",
+};
+
+struct mt8183_da7219_max98357_priv {
+ struct pinctrl *pinctrl;
+ struct pinctrl_state *pin_states[PIN_STATE_MAX];
+};
+
static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -244,6 +260,47 @@ SND_SOC_DAILINK_DEFS(tdm,
DAILINK_COMP_ARRAY(COMP_DUMMY()),
DAILINK_COMP_ARRAY(COMP_EMPTY()));
+static int mt8183_da7219_tdm_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct mt8183_da7219_max98357_priv *priv =
+ snd_soc_card_get_drvdata(rtd->card);
+ int ret;
+
+ if (IS_ERR(priv->pin_states[PIN_TDM_OUT_ON]))
+ return PTR_ERR(priv->pin_states[PIN_TDM_OUT_ON]);
+
+ ret = pinctrl_select_state(priv->pinctrl,
+ priv->pin_states[PIN_TDM_OUT_ON]);
+ if (ret)
+ dev_err(rtd->card->dev, "%s failed to select state %d\n",
+ __func__, ret);
+
+ return ret;
+}
+
+static void mt8183_da7219_tdm_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct mt8183_da7219_max98357_priv *priv =
+ snd_soc_card_get_drvdata(rtd->card);
+ int ret;
+
+ if (IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF]))
+ return;
+
+ ret = pinctrl_select_state(priv->pinctrl,
+ priv->pin_states[PIN_TDM_OUT_OFF]);
+ if (ret)
+ dev_err(rtd->card->dev, "%s failed to select state %d\n",
+ __func__, ret);
+}
+
+static struct snd_soc_ops mt8183_da7219_tdm_ops = {
+ .startup = mt8183_da7219_tdm_startup,
+ .shutdown = mt8183_da7219_tdm_shutdown,
+};
+
static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = {
/* FE */
{
@@ -395,6 +452,8 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = {
.no_pcm = 1,
.dpcm_playback = 1,
.ignore_suspend = 1,
+ .be_hw_params_fixup = mt8183_i2s_hw_params_fixup,
+ .ops = &mt8183_da7219_tdm_ops,
SND_SOC_DAILINK_REG(tdm),
},
};
@@ -470,7 +529,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev)
struct snd_soc_card *card = &mt8183_da7219_max98357_card;
struct device_node *platform_node;
struct snd_soc_dai_link *dai_link;
- struct pinctrl *default_pins;
+ struct mt8183_da7219_max98357_priv *priv;
int ret, i;
card->dev = &pdev->dev;
@@ -504,12 +563,45 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev)
return ret;
}
- default_pins =
- devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT);
- if (IS_ERR(default_pins)) {
- dev_err(&pdev->dev, "%s set pins failed\n",
+ priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ snd_soc_card_set_drvdata(card, priv);
+
+ priv->pinctrl = devm_pinctrl_get(&pdev->dev);
+ if (IS_ERR(priv->pinctrl)) {
+ dev_err(&pdev->dev, "%s devm_pinctrl_get failed\n",
__func__);
- return PTR_ERR(default_pins);
+ return PTR_ERR(priv->pinctrl);
+ }
+
+ for (i = 0; i < PIN_STATE_MAX; i++) {
+ priv->pin_states[i] = pinctrl_lookup_state(priv->pinctrl,
+ mt8183_pin_str[i]);
+ if (IS_ERR(priv->pin_states[i])) {
+ ret = PTR_ERR(priv->pin_states[i]);
+ dev_info(&pdev->dev, "%s Can't find pin state %s %d\n",
+ __func__, mt8183_pin_str[i], ret);
+ }
+ }
+
+ if (!IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) {
+ ret = pinctrl_select_state(priv->pinctrl,
+ priv->pin_states[PIN_TDM_OUT_OFF]);
+ if (ret)
+ dev_info(&pdev->dev,
+ "%s failed to select state %d\n",
+ __func__, ret);
+ }
+
+ if (!IS_ERR(priv->pin_states[PIN_STATE_DEFAULT])) {
+ ret = pinctrl_select_state(priv->pinctrl,
+ priv->pin_states[PIN_STATE_DEFAULT]);
+ if (ret)
+ dev_info(&pdev->dev,
+ "%s failed to select state %d\n",
+ __func__, ret);
}
return ret;
--
2.20.1
_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
^ permalink raw reply related [flat|nested] 116+ messages in thread
* [alsa-devel] Applied "ASoC: mediatek: mt8183-da7219: change supported formats of DL2 and UL1" to the asoc tree
@ 2020-02-13 13:32 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-13 13:32 UTC (permalink / raw)
To: Tzung-Bi Shih; +Cc: alsa-devel, Mark Brown
The patch
ASoC: mediatek: mt8183-da7219: change supported formats of DL2 and UL1
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From ec7ba9e1500b0af5bf30b4e56bfaaf3a88850bbf Mon Sep 17 00:00:00 2001
From: Tzung-Bi Shih <tzungbi@google.com>
Date: Thu, 13 Feb 2020 11:27:25 +0800
Subject: [PATCH] ASoC: mediatek: mt8183-da7219: change supported formats of
DL2 and UL1
DL2 and UL1 are for BTSCO. Provides only 16-bit, mono, 8kHz and
16kHz to userspace.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200213112003.1.Ie5aedb9d34ebfc7f05ceb382bfe346c45331cd63@changeid
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../mediatek/mt8183/mt8183-da7219-max98357.c | 42 +++++++++++++++++++
1 file changed, 42 insertions(+)
diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
index 1626541cc0d6..b52ffed882a7 100644
--- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
@@ -116,6 +116,46 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
return 0;
}
+static int
+mt8183_da7219_max98357_bt_sco_startup(
+ struct snd_pcm_substream *substream)
+{
+ static const unsigned int rates[] = {
+ 8000, 16000
+ };
+ static const struct snd_pcm_hw_constraint_list constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+ };
+ static const unsigned int channels[] = {
+ 1,
+ };
+ static const struct snd_pcm_hw_constraint_list constraints_channels = {
+ .count = ARRAY_SIZE(channels),
+ .list = channels,
+ .mask = 0,
+ };
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
+ runtime->hw.channels_max = 1;
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ &constraints_channels);
+
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+ snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
+
+ return 0;
+}
+
+static const struct snd_soc_ops mt8183_da7219_max98357_bt_sco_ops = {
+ .startup = mt8183_da7219_max98357_bt_sco_startup,
+};
+
/* FE */
SND_SOC_DAILINK_DEFS(playback1,
DAILINK_COMP_ARRAY(COMP_CPU("DL1")),
@@ -222,6 +262,7 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = {
SND_SOC_DPCM_TRIGGER_PRE},
.dynamic = 1,
.dpcm_playback = 1,
+ .ops = &mt8183_da7219_max98357_bt_sco_ops,
SND_SOC_DAILINK_REG(playback2),
},
{
@@ -240,6 +281,7 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = {
SND_SOC_DPCM_TRIGGER_PRE},
.dynamic = 1,
.dpcm_capture = 1,
+ .ops = &mt8183_da7219_max98357_bt_sco_ops,
SND_SOC_DAILINK_REG(capture1),
},
{
--
2.20.1
_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ALSA: usx2y: use for_each_pcm_streams() macro" to the asoc tree
@ 2020-02-19 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:09 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ALSA: usx2y: use for_each_pcm_streams() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From ffd11d1e7ad4602d5d1c2b4517ad316f7587d4d9 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:28:51 +0900
Subject: [PATCH] ALSA: usx2y: use for_each_pcm_streams() macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87sgj9aa8e.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/usb/usx2y/usbusx2yaudio.c | 9 +++++----
1 file changed, 5 insertions(+), 4 deletions(-)
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 772f6f3ccbb1..37d290fe9d43 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -906,11 +906,12 @@ static const struct snd_pcm_ops snd_usX2Y_pcm_ops =
*/
static void usX2Y_audio_stream_free(struct snd_usX2Y_substream **usX2Y_substream)
{
- kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]);
- usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL;
+ int stream;
- kfree(usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]);
- usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE] = NULL;
+ for_each_pcm_streams(stream) {
+ kfree(usX2Y_substream[stream]);
+ usX2Y_substream[stream] = NULL;
+ }
}
static void snd_usX2Y_pcm_private_free(struct snd_pcm *pcm)
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: sof: sof-audio: use for_each_pcm_streams() macro" to the asoc tree
@ 2020-02-19 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:09 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: sof: sof-audio: use for_each_pcm_streams() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 525c4107da8c0a86aa3548dc6e1d0014749e95f7 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:28:47 +0900
Subject: [PATCH] ASoC: sof: sof-audio: use for_each_pcm_streams() macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87tv3paa8i.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/sof-audio.c | 19 +++++++------------
1 file changed, 7 insertions(+), 12 deletions(-)
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
index 75f2ef2bd94b..fc4ed2a8a914 100644
--- a/sound/soc/sof/sof-audio.c
+++ b/sound/soc/sof/sof-audio.c
@@ -23,7 +23,7 @@ bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev)
int dir;
list_for_each_entry(spcm, &sdev->pcm_list, list) {
- for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) {
+ for_each_pcm_streams(dir) {
substream = spcm->stream[dir].substream;
if (!substream || !substream->runtime)
continue;
@@ -71,7 +71,7 @@ int sof_set_hw_params_upon_resume(struct device *dev)
* have been suspended.
*/
list_for_each_entry(spcm, &sdev->pcm_list, list) {
- for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) {
+ for_each_pcm_streams(dir) {
/*
* do not reset hw_params upon resume for streams that
* were kept running during suspend
@@ -319,16 +319,11 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp,
int dir;
list_for_each_entry(spcm, &sdev->pcm_list, list) {
- dir = SNDRV_PCM_STREAM_PLAYBACK;
- if (spcm->stream[dir].comp_id == comp_id) {
- *direction = dir;
- return spcm;
- }
-
- dir = SNDRV_PCM_STREAM_CAPTURE;
- if (spcm->stream[dir].comp_id == comp_id) {
- *direction = dir;
- return spcm;
+ for_each_pcm_streams(dir) {
+ if (spcm->stream[dir].comp_id == comp_id) {
+ *direction = dir;
+ return spcm;
+ }
}
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro" to the asoc tree
@ 2020-02-19 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:09 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 4c260c3f19bd16e6b11841aad1162f5a105ed24e Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:28:44 +0900
Subject: [PATCH] ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87v9o5aa8m.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/lpass-platform.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index b05091c283b7..5d1bc5757169 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -529,7 +529,7 @@ static void lpass_platform_pcm_free(struct snd_soc_component *component,
struct snd_pcm_substream *substream;
int i;
- for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) {
+ for_each_pcm_streams(i) {
substream = pcm->streams[i].substream;
if (substream) {
snd_dma_free_pages(&substream->dma_buffer);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro" to the asoc tree
@ 2020-02-19 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:09 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From fa7b2a1fcb92906a284b0824b45866a7b8afb599 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:28:40 +0900
Subject: [PATCH] ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87wo8laa8p.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/fsl/fsl_asrc_dma.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index ece130f59d15..44e5924be870 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -400,7 +400,7 @@ static int fsl_asrc_dma_pcm_new(struct snd_soc_component *component,
return ret;
}
- for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
+ for_each_pcm_streams(i) {
substream = pcm->streams[i].substream;
if (!substream)
continue;
@@ -428,7 +428,7 @@ static void fsl_asrc_dma_pcm_free(struct snd_soc_component *component,
struct snd_pcm_substream *substream;
int i;
- for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
+ for_each_pcm_streams(i) {
substream = pcm->streams[i].substream;
if (!substream)
continue;
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro" to the asoc tree
@ 2020-02-19 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:09 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 0a170be9631ea8335e494f3c5f7ab720287023a2 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:28:36 +0900
Subject: [PATCH] ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87y2t1aa8t.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/dwc/dwc-i2s.c | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c
index a8bff6f08a69..515f88456dbd 100644
--- a/sound/soc/dwc/dwc-i2s.c
+++ b/sound/soc/dwc/dwc-i2s.c
@@ -422,15 +422,15 @@ static int dw_i2s_resume(struct snd_soc_component *component)
{
struct dw_i2s_dev *dev = snd_soc_component_get_drvdata(component);
struct snd_soc_dai *dai;
+ int stream;
if (dev->capability & DW_I2S_MASTER)
clk_enable(dev->clk);
for_each_component_dais(component, dai) {
- if (dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
- dw_i2s_config(dev, SNDRV_PCM_STREAM_PLAYBACK);
- if (dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])
- dw_i2s_config(dev, SNDRV_PCM_STREAM_CAPTURE);
+ for_each_pcm_streams(stream)
+ if (dai->stream_active[stream])
+ dw_i2s_config(dev, stream);
}
return 0;
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams() macro" to the asoc tree
@ 2020-02-19 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:09 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From ee10fbe1cdf7cb4ae62f5e23ccd771e696b8f404 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:28:32 +0900
Subject: [PATCH] ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams()
macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87zhdhaa8x.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-generic-dmaengine-pcm.c | 8 +++-----
1 file changed, 3 insertions(+), 5 deletions(-)
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 2cc25651661c..d6b4831e8aec 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -237,7 +237,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component,
max_buffer_size = SIZE_MAX;
}
- for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
+ for_each_pcm_streams(i) {
substream = rtd->pcm->streams[i].substream;
if (!substream)
continue;
@@ -371,8 +371,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
dev = config->dma_dev;
}
- for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
- i++) {
+ for_each_pcm_streams(i) {
if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
name = "rx-tx";
else
@@ -401,8 +400,7 @@ static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm)
{
unsigned int i;
- for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
- i++) {
+ for_each_pcm_streams(i) {
if (!pcm->chan[i])
continue;
dma_release_channel(pcm->chan[i]);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: use for_each_pcm_streams() macro" to the asoc tree
@ 2020-02-19 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:09 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: use for_each_pcm_streams() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 7083f877ea66e106f90e9a1a0dabb19ebbacc4e6 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:28:28 +0900
Subject: [PATCH] ASoC: soc-pcm: use for_each_pcm_streams() macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/871rqtboth.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 97 ++++++++++++++++-----------------------------
1 file changed, 35 insertions(+), 62 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 6fd69574ca31..63f67eb7c077 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2612,6 +2612,7 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream)
static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
{
struct snd_soc_dapm_widget_list *list;
+ int stream;
int count, paths;
if (!fe->dai_link->dynamic)
@@ -2625,69 +2626,42 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n",
new ? "new" : "old", fe->dai_link->name);
- /* skip if FE doesn't have playback capability */
- if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK) ||
- !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_PLAYBACK))
- goto capture;
-
- /* skip if FE isn't currently playing */
- if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] ||
- !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
- goto capture;
-
- paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
- if (paths < 0) {
- dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
- fe->dai_link->name, "playback");
- return paths;
- }
-
- /* update any playback paths */
- count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new);
- if (count) {
- if (new)
- dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
- else
- dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
+ for_each_pcm_streams(stream) {
- dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
- dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
- }
-
- dpcm_path_put(&list);
+ /* skip if FE doesn't have playback/capture capability */
+ if (!snd_soc_dai_stream_valid(fe->cpu_dai, stream) ||
+ !snd_soc_dai_stream_valid(fe->codec_dai, stream))
+ continue;
-capture:
- /* skip if FE doesn't have capture capability */
- if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE) ||
- !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_CAPTURE))
- return 0;
+ /* skip if FE isn't currently playing/capturing */
+ if (!fe->cpu_dai->stream_active[stream] ||
+ !fe->codec_dai->stream_active[stream])
+ continue;
- /* skip if FE isn't currently capturing */
- if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] ||
- !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])
- return 0;
+ paths = dpcm_path_get(fe, stream, &list);
+ if (paths < 0) {
+ dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
+ fe->dai_link->name,
+ stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ "playback" : "capture");
+ return paths;
+ }
- paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
- if (paths < 0) {
- dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
- fe->dai_link->name, "capture");
- return paths;
- }
+ /* update any playback/capture paths */
+ count = dpcm_process_paths(fe, stream, &list, new);
+ if (count) {
+ if (new)
+ dpcm_run_new_update(fe, stream);
+ else
+ dpcm_run_old_update(fe, stream);
- /* update any old capture paths */
- count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new);
- if (count) {
- if (new)
- dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE);
- else
- dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE);
+ dpcm_clear_pending_state(fe, stream);
+ dpcm_be_disconnect(fe, stream);
+ }
- dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
- dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
+ dpcm_path_put(&list);
}
- dpcm_path_put(&list);
-
return 0;
}
@@ -3114,19 +3088,18 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf,
{
struct snd_soc_pcm_runtime *fe = file->private_data;
ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0;
+ int stream;
char *buf;
buf = kmalloc(out_count, GFP_KERNEL);
if (!buf)
return -ENOMEM;
- if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK))
- offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK,
- buf + offset, out_count - offset);
-
- if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE))
- offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE,
- buf + offset, out_count - offset);
+ for_each_pcm_streams(stream)
+ if (snd_soc_dai_stream_valid(fe->cpu_dai, stream))
+ offset += dpcm_show_state(fe, stream,
+ buf + offset,
+ out_count - offset);
ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-core: use for_each_pcm_streams() macro" to the asoc tree
@ 2020-02-19 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:09 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-core: use for_each_pcm_streams() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From d74c2a156b710e9ad81193a60e037430f8894c0c Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:28:25 +0900
Subject: [PATCH] ASoC: soc-core: use for_each_pcm_streams() macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/8736b9botk.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-core.c | 31 +++++++++++++------------------
1 file changed, 13 insertions(+), 18 deletions(-)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index f0ae1a7d7e09..30c17fde14ca 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -432,6 +432,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
struct snd_soc_component *component;
struct device *dev;
int ret;
+ int stream;
/*
* for rtd->dev
@@ -466,10 +467,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
rtd->dev = dev;
INIT_LIST_HEAD(&rtd->list);
- INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);
- INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients);
- INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients);
- INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
+ for_each_pcm_streams(stream) {
+ INIT_LIST_HEAD(&rtd->dpcm[stream].be_clients);
+ INIT_LIST_HEAD(&rtd->dpcm[stream].fe_clients);
+ }
dev_set_drvdata(dev, rtd);
INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
@@ -559,17 +560,14 @@ int snd_soc_suspend(struct device *dev)
snd_soc_flush_all_delayed_work(card);
for_each_card_rtds(card, rtd) {
+ int stream;
if (rtd->dai_link->ignore_suspend)
continue;
- snd_soc_dapm_stream_event(rtd,
- SNDRV_PCM_STREAM_PLAYBACK,
- SND_SOC_DAPM_STREAM_SUSPEND);
-
- snd_soc_dapm_stream_event(rtd,
- SNDRV_PCM_STREAM_CAPTURE,
- SND_SOC_DAPM_STREAM_SUSPEND);
+ for_each_pcm_streams(stream)
+ snd_soc_dapm_stream_event(rtd, stream,
+ SND_SOC_DAPM_STREAM_SUSPEND);
}
/* Recheck all endpoints too, their state is affected by suspend */
@@ -665,17 +663,14 @@ static void soc_resume_deferred(struct work_struct *work)
}
for_each_card_rtds(card, rtd) {
+ int stream;
if (rtd->dai_link->ignore_suspend)
continue;
- snd_soc_dapm_stream_event(rtd,
- SNDRV_PCM_STREAM_PLAYBACK,
- SND_SOC_DAPM_STREAM_RESUME);
-
- snd_soc_dapm_stream_event(rtd,
- SNDRV_PCM_STREAM_CAPTURE,
- SND_SOC_DAPM_STREAM_RESUME);
+ for_each_pcm_streams(stream)
+ snd_soc_dapm_stream_event(rtd, stream,
+ SND_SOC_DAPM_STREAM_RESUME);
}
/* unmute any active DACs */
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ALSA: pcm.h: add for_each_pcm_streams()" to the asoc tree
@ 2020-02-19 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:09 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: Takashi Iwai, alsa-devel, Mark Brown
The patch
ALSA: pcm.h: add for_each_pcm_streams()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 3193abd26b515ccac65e1c323533cb7f53d06176 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:28:19 +0900
Subject: [PATCH] ALSA: pcm.h: add for_each_pcm_streams()
ALSA code has SNDRV_PCM_STREAM_PLAYBACK/CAPTURE everywhere.
Having for_each_xxxx macro is useful.
This patch adds for_each_pcm_streams() for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/874kvpbotq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/pcm.h | 5 +++++
1 file changed, 5 insertions(+)
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index f657ff08f317..2628246b76fa 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -644,6 +644,11 @@ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream,
#define snd_pcm_group_for_each_entry(s, substream) \
list_for_each_entry(s, &substream->group->substreams, link_list)
+#define for_each_pcm_streams(stream) \
+ for (stream = SNDRV_PCM_STREAM_PLAYBACK; \
+ stream <= SNDRV_PCM_STREAM_LAST; \
+ stream++)
+
/**
* snd_pcm_running - Check whether the substream is in a running state
* @substream: substream to check
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: merge playback/cature_active into stream_active" to the asoc tree
@ 2020-02-19 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:09 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: merge playback/cature_active into stream_active
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 0f6011fd79a2fb92cb80177fd6bdc8aac3a3cd93 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:28:15 +0900
Subject: [PATCH] ASoC: soc-pcm: merge playback/cature_active into
stream_active
DAI has playback_active and capture_active to care usage count.
OTOH, we have SNDRV_PCM_STREAM_PLAYBACK/CAPTURE.
But because of this kind of implementation mismatch,
ALSA SoC has many verbose code.
To solve this issue, this patch merge playback_active/capture_active
into stream_active[2];
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/875zg5botu.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc-dai.h | 3 +--
sound/soc/codecs/cs4271.c | 4 ++--
sound/soc/dwc/dwc-i2s.c | 4 ++--
sound/soc/soc-core.c | 17 +++++++++--------
sound/soc/soc-pcm.c | 25 ++++++++++++-------------
5 files changed, 26 insertions(+), 27 deletions(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 04c23ac0dfff..7481e468be39 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -322,8 +322,7 @@ struct snd_soc_dai {
struct snd_soc_dai_driver *driver;
/* DAI runtime info */
- unsigned int capture_active; /* stream usage count */
- unsigned int playback_active; /* stream usage count */
+ unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1]; /* usage count */
unsigned int active;
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 04b86a51e055..62f412d6f9f2 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -356,9 +356,9 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
*/
if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
- !dai->capture_active) ||
+ !dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) ||
(substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
- !dai->playback_active)) {
+ !dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])) {
ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
CS4271_MODE2_PDN,
CS4271_MODE2_PDN);
diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c
index 7eeca2150b2d..a8bff6f08a69 100644
--- a/sound/soc/dwc/dwc-i2s.c
+++ b/sound/soc/dwc/dwc-i2s.c
@@ -427,9 +427,9 @@ static int dw_i2s_resume(struct snd_soc_component *component)
clk_enable(dev->clk);
for_each_component_dais(component, dai) {
- if (dai->playback_active)
+ if (dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
dw_i2s_config(dev, SNDRV_PCM_STREAM_PLAYBACK);
- if (dai->capture_active)
+ if (dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])
dw_i2s_config(dev, SNDRV_PCM_STREAM_CAPTURE);
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 6a58a8f6e3c4..f0ae1a7d7e09 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -365,19 +365,20 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime);
void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int playback = SNDRV_PCM_STREAM_PLAYBACK;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
dev_dbg(rtd->dev,
"ASoC: pop wq checking: %s status: %s waiting: %s\n",
codec_dai->driver->playback.stream_name,
- codec_dai->playback_active ? "active" : "inactive",
+ codec_dai->stream_active[playback] ? "active" : "inactive",
rtd->pop_wait ? "yes" : "no");
/* are we waiting on this codec DAI stream */
if (rtd->pop_wait == 1) {
rtd->pop_wait = 0;
- snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+ snd_soc_dapm_stream_event(rtd, playback,
SND_SOC_DAPM_STREAM_STOP);
}
@@ -514,6 +515,7 @@ int snd_soc_suspend(struct device *dev)
struct snd_soc_card *card = dev_get_drvdata(dev);
struct snd_soc_component *component;
struct snd_soc_pcm_runtime *rtd;
+ int playback = SNDRV_PCM_STREAM_PLAYBACK;
int i;
/* If the card is not initialized yet there is nothing to do */
@@ -537,9 +539,8 @@ int snd_soc_suspend(struct device *dev)
continue;
for_each_rtd_codec_dai(rtd, i, dai) {
- if (dai->playback_active)
- snd_soc_dai_digital_mute(dai, 1,
- SNDRV_PCM_STREAM_PLAYBACK);
+ if (dai->stream_active[playback])
+ snd_soc_dai_digital_mute(dai, 1, playback);
}
}
@@ -680,14 +681,14 @@ static void soc_resume_deferred(struct work_struct *work)
/* unmute any active DACs */
for_each_card_rtds(card, rtd) {
struct snd_soc_dai *dai;
+ int playback = SNDRV_PCM_STREAM_PLAYBACK;
if (rtd->dai_link->ignore_suspend)
continue;
for_each_rtd_codec_dai(rtd, i, dai) {
- if (dai->playback_active)
- snd_soc_dai_digital_mute(dai, 0,
- SNDRV_PCM_STREAM_PLAYBACK);
+ if (dai->stream_active[playback])
+ snd_soc_dai_digital_mute(dai, 0, playback);
}
}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 5a79a830ee18..6fd69574ca31 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -100,15 +100,9 @@ static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd,
lockdep_assert_held(&rtd->card->pcm_mutex);
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- cpu_dai->playback_active += action;
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- codec_dai->playback_active += action;
- } else {
- cpu_dai->capture_active += action;
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- codec_dai->capture_active += action;
- }
+ cpu_dai->stream_active[stream] += action;
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ codec_dai->stream_active[stream] += action;
cpu_dai->active += action;
cpu_dai->component->active += action;
@@ -967,8 +961,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
/* apply codec digital mute */
for_each_rtd_codec_dai(rtd, i, codec_dai) {
- if ((playback && codec_dai->playback_active == 1) ||
- (!playback && codec_dai->capture_active == 1))
+ int playback_active = codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK];
+ int capture_active = codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE];
+
+ if ((playback && playback_active == 1) ||
+ (!playback && capture_active == 1))
snd_soc_dai_digital_mute(codec_dai, 1,
substream->stream);
}
@@ -2634,7 +2631,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
goto capture;
/* skip if FE isn't currently playing */
- if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active)
+ if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] ||
+ !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
goto capture;
paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
@@ -2665,7 +2663,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
return 0;
/* skip if FE isn't currently capturing */
- if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active)
+ if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] ||
+ !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])
return 0;
paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: use goto and remove multi return" to the asoc tree
@ 2020-02-19 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:09 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: use goto and remove multi return
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From cae06eb92557f0a073835380e57abee5f8173d73 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:28:11 +0900
Subject: [PATCH] ASoC: soc-pcm: use goto and remove multi return
When we use some kind of lock, we need to do unlock.
In that time, multi unlock/return is not good implementation.
This patch add label and use goto at dpcm_fe_dai_open()
to reduce such code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/877e0lboty.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index d77a2c22a04f..5a79a830ee18 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2733,8 +2733,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
ret = dpcm_path_get(fe, stream, &list);
if (ret < 0) {
- mutex_unlock(&fe->card->mutex);
- return ret;
+ goto open_end;
} else if (ret == 0) {
dev_dbg(fe->dev, "ASoC: %s no valid %s route\n",
fe->dai_link->name, stream ? "capture" : "playback");
@@ -2755,6 +2754,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
dpcm_clear_pending_state(fe, stream);
dpcm_path_put(&list);
+open_end:
mutex_unlock(&fe->card->mutex);
return ret;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code" to the asoc tree
@ 2020-02-19 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:09 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 085d22be035db245c44714cf879a73eae06c9f6b Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:28:07 +0900
Subject: [PATCH] ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate
code
Below functions are doing very similar things, the difference is
used state only.
snd_soc_dpcm_can_be_free_stop()
snd_soc_dpcm_can_be_params()
This patch adds common snd_soc_dpcm_check_state(), and use it from
snd_soc_dpcm_can_be_free_stop() / snd_soc_dpcm_can_be_params().
It can reduce duplicate code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/878sl1bou2.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 70 ++++++++++++++++++++++-----------------------
1 file changed, 35 insertions(+), 35 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index bd4e4f86f5b2..d77a2c22a04f 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2954,17 +2954,17 @@ struct snd_pcm_substream *
}
EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream);
-/*
- * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE
- * are not running, paused or suspended for the specified stream direction.
- */
-int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
- struct snd_soc_pcm_runtime *be, int stream)
+static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be,
+ int stream,
+ const enum snd_soc_dpcm_state *states,
+ int num_states)
{
struct snd_soc_dpcm *dpcm;
int state;
int ret = 1;
unsigned long flags;
+ int i;
spin_lock_irqsave(&fe->card->dpcm_lock, flags);
for_each_dpcm_fe(be, stream, dpcm) {
@@ -2973,18 +2973,34 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
continue;
state = dpcm->fe->dpcm[stream].state;
- if (state == SND_SOC_DPCM_STATE_START ||
- state == SND_SOC_DPCM_STATE_PAUSED ||
- state == SND_SOC_DPCM_STATE_SUSPEND) {
- ret = 0;
- break;
+ for (i = 0; i < num_states; i++) {
+ if (state == states[i]) {
+ ret = 0;
+ break;
+ }
}
}
spin_unlock_irqrestore(&fe->card->dpcm_lock, flags);
- /* it's safe to free/stop this BE DAI */
+ /* it's safe to do this BE DAI */
return ret;
}
+
+/*
+ * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE
+ * are not running, paused or suspended for the specified stream direction.
+ */
+int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream)
+{
+ const enum snd_soc_dpcm_state state[] = {
+ SND_SOC_DPCM_STATE_START,
+ SND_SOC_DPCM_STATE_PAUSED,
+ SND_SOC_DPCM_STATE_SUSPEND,
+ };
+
+ return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
+}
EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop);
/*
@@ -2994,30 +3010,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop);
int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
struct snd_soc_pcm_runtime *be, int stream)
{
- struct snd_soc_dpcm *dpcm;
- int state;
- int ret = 1;
- unsigned long flags;
-
- spin_lock_irqsave(&fe->card->dpcm_lock, flags);
- for_each_dpcm_fe(be, stream, dpcm) {
-
- if (dpcm->fe == fe)
- continue;
+ const enum snd_soc_dpcm_state state[] = {
+ SND_SOC_DPCM_STATE_START,
+ SND_SOC_DPCM_STATE_PAUSED,
+ SND_SOC_DPCM_STATE_SUSPEND,
+ SND_SOC_DPCM_STATE_PREPARE,
+ };
- state = dpcm->fe->dpcm[stream].state;
- if (state == SND_SOC_DPCM_STATE_START ||
- state == SND_SOC_DPCM_STATE_PAUSED ||
- state == SND_SOC_DPCM_STATE_SUSPEND ||
- state == SND_SOC_DPCM_STATE_PREPARE) {
- ret = 0;
- break;
- }
- }
- spin_unlock_irqrestore(&fe->card->dpcm_lock, flags);
-
- /* it's safe to change hw_params */
- return ret;
+ return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
}
EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state()" to the asoc tree
@ 2020-02-19 0:09 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:09 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 289a7e64f8583aaa45847c7fa3b7fabf8d48fd6b Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:28:04 +0900
Subject: [PATCH] ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state()
No one is using snd_soc_dpcm_be_get/set_state().
If it exists only by assumption that "it may be necessary someday",
let's remove it now. Otherwise code maintenance will be difficult.
We can revive it when we really needed it.
Let's remove it, so far.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87a75hbou7.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc-dpcm.h | 8 --------
sound/soc/soc-pcm.c | 16 ----------------
2 files changed, 24 deletions(-)
diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h
index 665516387671..3e7819d2a6aa 100644
--- a/include/sound/soc-dpcm.h
+++ b/include/sound/soc-dpcm.h
@@ -132,14 +132,6 @@ int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe,
struct snd_pcm_substream *
snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream);
-/* get the BE runtime state */
-enum snd_soc_dpcm_state
- snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream);
-
-/* set the BE runtime state */
-void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream,
- enum snd_soc_dpcm_state state);
-
/* internal use only */
int soc_dpcm_runtime_update(struct snd_soc_card *);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b8ea4d892031..bd4e4f86f5b2 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2954,22 +2954,6 @@ struct snd_pcm_substream *
}
EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream);
-/* get the BE runtime state */
-enum snd_soc_dpcm_state
- snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream)
-{
- return be->dpcm[stream].state;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_get_state);
-
-/* set the BE runtime state */
-void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be,
- int stream, enum snd_soc_dpcm_state state)
-{
- be->dpcm[stream].state = state;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_set_state);
-
/*
* We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE
* are not running, paused or suspended for the specified stream direction.
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: remove soc_dpcm_be_digital_mute()" to the asoc tree
@ 2020-02-19 0:10 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:10 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: remove soc_dpcm_be_digital_mute()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From c9645d2a952b7925b6708b24242cd5ed04975648 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:27:57 +0900
Subject: [PATCH] ASoC: soc-pcm: remove soc_dpcm_be_digital_mute()
No one is using soc_dpcm_be_digital_mute().
If it exists only by assumption that "it may be necessary someday",
let's remove it now. Otherwise code maintenance will be difficult.
We can revive it when we really needed it.
Let's remove it, so far.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87blpxbouc.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc-dpcm.h | 1 -
sound/soc/soc-pcm.c | 27 ---------------------------
2 files changed, 28 deletions(-)
diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h
index b654ebfc8766..665516387671 100644
--- a/include/sound/soc-dpcm.h
+++ b/include/sound/soc-dpcm.h
@@ -141,7 +141,6 @@ void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream,
enum snd_soc_dpcm_state state);
/* internal use only */
-int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute);
int soc_dpcm_runtime_update(struct snd_soc_card *);
#ifdef CONFIG_DEBUG_FS
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 1d48be24bfaa..b8ea4d892031 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2719,33 +2719,6 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
mutex_unlock(&card->mutex);
return ret;
}
-int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
-{
- struct snd_soc_dpcm *dpcm;
- struct snd_soc_dai *dai;
-
- for_each_dpcm_be(fe, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
-
- struct snd_soc_pcm_runtime *be = dpcm->be;
- int i;
-
- if (be->dai_link->ignore_suspend)
- continue;
-
- for_each_rtd_codec_dai(be, i, dai) {
- struct snd_soc_dai_driver *drv = dai->driver;
-
- dev_dbg(be->dev, "ASoC: BE digital mute %s\n",
- be->dai_link->name);
-
- if (drv->ops && drv->ops->digital_mute &&
- dai->playback_active)
- drv->ops->digital_mute(dai, mute);
- }
- }
-
- return 0;
-}
static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
{
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be()" to the asoc tree
@ 2020-02-19 0:10 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:10 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 027a483871832044fa0cb8e9df208cca5230ae91 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:27:53 +0900
Subject: [PATCH] ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be()
dpcm_end_walk_at_be() and dpcm_get_be() are almost same code.
This patch uses dpcm_get_be() from dpcm_end_walk_at_be().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87d0adbouh.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 20 +++-----------------
1 file changed, 3 insertions(+), 17 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 7d4419ae63f6..1d48be24bfaa 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1297,10 +1297,7 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget,
{
struct snd_soc_card *card = widget->dapm->card;
struct snd_soc_pcm_runtime *rtd;
- struct snd_soc_dapm_widget *w;
- struct snd_soc_dai *dai;
int stream;
- int i;
/* adjust dir to stream */
if (dir == SND_SOC_DAPM_DIR_OUT)
@@ -1308,20 +1305,9 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget,
else
stream = SNDRV_PCM_STREAM_CAPTURE;
- for_each_card_rtds(card, rtd) {
- if (!rtd->dai_link->no_pcm)
- continue;
-
- w = dai_get_widget(rtd->cpu_dai, stream);
- if (w == widget)
- return true;
-
- for_each_rtd_codec_dai(rtd, i, dai) {
- w = dai_get_widget(dai, stream);
- if (w == widget)
- return true;
- }
- }
+ rtd = dpcm_get_be(card, widget, stream);
+ if (rtd)
+ return true;
return false;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be()" to the asoc tree
@ 2020-02-19 0:10 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:10 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From c2cd821603c216a6a7242b2b4c1a093051e26aaf Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:27:48 +0900
Subject: [PATCH] ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be()
dpcm_end_walk_at_be() has very duplicate code.
dpcm_end_walk_at_be() {
...
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
(1) /* code for Playback */
} else {
(2) /* code for Capture */
}
}
The difference between Playback (1) and Capture (2) code is pointer only
(= "playback_widget" or "caputre_widget").
OTOH, now we already has dai_get_widget() for it.
This means we can merge (1) and (2).
This patch do it and remove duplicated code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87eeutboul.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 37 ++++++++++++++++---------------------
1 file changed, 16 insertions(+), 21 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b708db972310..7d4419ae63f6 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1297,34 +1297,29 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget,
{
struct snd_soc_card *card = widget->dapm->card;
struct snd_soc_pcm_runtime *rtd;
+ struct snd_soc_dapm_widget *w;
struct snd_soc_dai *dai;
+ int stream;
int i;
- if (dir == SND_SOC_DAPM_DIR_OUT) {
- for_each_card_rtds(card, rtd) {
- if (!rtd->dai_link->no_pcm)
- continue;
+ /* adjust dir to stream */
+ if (dir == SND_SOC_DAPM_DIR_OUT)
+ stream = SNDRV_PCM_STREAM_PLAYBACK;
+ else
+ stream = SNDRV_PCM_STREAM_CAPTURE;
- if (rtd->cpu_dai->playback_widget == widget)
- return true;
+ for_each_card_rtds(card, rtd) {
+ if (!rtd->dai_link->no_pcm)
+ continue;
- for_each_rtd_codec_dai(rtd, i, dai) {
- if (dai->playback_widget == widget)
- return true;
- }
- }
- } else { /* SND_SOC_DAPM_DIR_IN */
- for_each_card_rtds(card, rtd) {
- if (!rtd->dai_link->no_pcm)
- continue;
+ w = dai_get_widget(rtd->cpu_dai, stream);
+ if (w == widget)
+ return true;
- if (rtd->cpu_dai->capture_widget == widget)
+ for_each_rtd_codec_dai(rtd, i, dai) {
+ w = dai_get_widget(dai, stream);
+ if (w == widget)
return true;
-
- for_each_rtd_codec_dai(rtd, i, dai) {
- if (dai->capture_widget == widget)
- return true;
- }
}
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be()" to the asoc tree
@ 2020-02-19 0:10 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:10 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 93597fae552a35d27cd1f399ffab6a6862cf9dc3 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:27:43 +0900
Subject: [PATCH] ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be()
dpcm_get_be() has very duplicate code.
dpcm_get_be() {
...
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
(1) /* code for Playback */
} else {
(2) /* code for Capture */
}
}
The difference between Playback (1) and Capture (2) code is pointer only
(= "playback_widget" or "caputre_widget").
OTOH, now we already has dai_get_widget() for it.
This means we can merge (1) and (2).
This patch do it and remove duplicated code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87ftf9bouq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 41 ++++++++++++-----------------------------
1 file changed, 12 insertions(+), 29 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 23e36f4f965c..b708db972310 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1246,47 +1246,30 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
struct snd_soc_dapm_widget *widget, int stream)
{
struct snd_soc_pcm_runtime *be;
+ struct snd_soc_dapm_widget *w;
struct snd_soc_dai *dai;
int i;
dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name);
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- for_each_card_rtds(card, be) {
+ for_each_card_rtds(card, be) {
- if (!be->dai_link->no_pcm)
- continue;
+ if (!be->dai_link->no_pcm)
+ continue;
- dev_dbg(card->dev, "ASoC: try BE : %s\n",
- be->cpu_dai->playback_widget ?
- be->cpu_dai->playback_widget->name : "(not set)");
+ w = dai_get_widget(be->cpu_dai, stream);
- if (be->cpu_dai->playback_widget == widget)
- return be;
+ dev_dbg(card->dev, "ASoC: try BE : %s\n",
+ w ? w->name : "(not set)");
- for_each_rtd_codec_dai(be, i, dai) {
- if (dai->playback_widget == widget)
- return be;
- }
- }
- } else {
-
- for_each_card_rtds(card, be) {
-
- if (!be->dai_link->no_pcm)
- continue;
+ if (w == widget)
+ return be;
- dev_dbg(card->dev, "ASoC: try BE %s\n",
- be->cpu_dai->capture_widget ?
- be->cpu_dai->capture_widget->name : "(not set)");
+ for_each_rtd_codec_dai(be, i, dai) {
+ w = dai_get_widget(dai, stream);
- if (be->cpu_dai->capture_widget == widget)
+ if (w == widget)
return be;
-
- for_each_rtd_codec_dai(be, i, dai) {
- if (dai->capture_widget == widget)
- return be;
- }
}
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: move dai_get_widget()" to the asoc tree
@ 2020-02-19 0:10 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:10 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: move dai_get_widget()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From d9303690f753dfdae51304fc89f4b04c0549a9f7 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 17 Feb 2020 17:27:39 +0900
Subject: [PATCH] ASoC: soc-pcm: move dai_get_widget()
This patch moves dai_get_widget() to top side.
This is prepare for cleanup soc-pcm.c
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87h7zpbouu.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 18 +++++++++---------
1 file changed, 9 insertions(+), 9 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 65a3856be250..23e36f4f965c 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -82,6 +82,15 @@ static int soc_rtd_trigger(struct snd_soc_pcm_runtime *rtd,
return 0;
}
+static inline
+struct snd_soc_dapm_widget *dai_get_widget(struct snd_soc_dai *dai, int stream)
+{
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return dai->playback_widget;
+ else
+ return dai->capture_widget;
+}
+
static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd,
int stream, int action)
{
@@ -1287,15 +1296,6 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
return NULL;
}
-static inline struct snd_soc_dapm_widget *
- dai_get_widget(struct snd_soc_dai *dai, int stream)
-{
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- return dai->playback_widget;
- else
- return dai->capture_widget;
-}
-
static int widget_in_list(struct snd_soc_dapm_widget_list *list,
struct snd_soc_dapm_widget *widget)
{
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: rt5682: Add DAI clock binding info for WCLK/BCLK CCF usage" to the asoc tree
@ 2020-02-19 0:10 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:10 UTC (permalink / raw)
To: Derek Fang; +Cc: alsa-devel, Mark Brown
The patch
ASoC: rt5682: Add DAI clock binding info for WCLK/BCLK CCF usage
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 8b59e642d05f0ae9800b057350c063fe7debd6bc Mon Sep 17 00:00:00 2001
From: Derek Fang <derek.fang@realtek.com>
Date: Tue, 18 Feb 2020 21:51:52 +0800
Subject: [PATCH] ASoC: rt5682: Add DAI clock binding info for WCLK/BCLK CCF
usage
This patch describes that rt5682 can expose WCLK and BCLK clocks
and how to use.
Signed-off-by: Derek Fang <derek.fang@realtek.com>
Link: https://lore.kernel.org/r/1582033912-6841-2-git-send-email-derek.fang@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/rt5682.txt | 12 ++++++++++++
1 file changed, 12 insertions(+)
diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt
index 30e927a28369..ac98151d29e4 100644
--- a/Documentation/devicetree/bindings/sound/rt5682.txt
+++ b/Documentation/devicetree/bindings/sound/rt5682.txt
@@ -32,6 +32,12 @@ Optional properties:
The delay time is realtek,btndet-delay value multiple of 8.192 ms.
If absent, the default is 16.
+- #clock-cells : Should be set to '<1>', wclk and bclk sources provided.
+- clock-output-names : Name given for DAI clocks output.
+
+- clocks : phandle and clock specifier for codec MCLK.
+- clock-names : Clock name string for 'clocks' attribute, should be "mclk".
+
Pins on the device (for linking into audio routes) for RT5682:
* DMIC L1
@@ -53,4 +59,10 @@ rt5682 {
realtek,dmic1-clk-pin = <1>;
realtek,jd-src = <1>;
realtek,btndet-delay = <16>;
+
+ #clock-cells = <1>;
+ clock-output-names = "rt5682-dai-wclk", "rt5682-dai-bclk";
+
+ clocks = <&osc>;
+ clock-names = "mclk";
};
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: Intel: Add Probe compress CPU DAIs" to the asoc tree
@ 2020-02-19 0:10 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:10 UTC (permalink / raw)
To: Cezary Rojewski; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: SOF: Intel: Add Probe compress CPU DAIs
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 70368106467cd8c420176bf3ab0acc797f6584bf Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Tue, 18 Feb 2020 15:39:24 +0100
Subject: [PATCH] ASoC: SOF: Intel: Add Probe compress CPU DAIs
Declare extraction CPU DAI as well as sof_probe_compr_ops. FE DAIs can
link against these new CPU DAI to create new compress devices.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200218143924.10565-10-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/compress.c | 5 +++++
sound/soc/sof/compress.h | 2 ++
sound/soc/sof/intel/hda-dai.c | 28 ++++++++++++++++++++++++++++
sound/soc/sof/intel/hda.h | 6 ++++++
sound/soc/sof/pcm.c | 7 +++++++
5 files changed, 48 insertions(+)
diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c
index e87cc81a0599..7354dc6a49cf 100644
--- a/sound/soc/sof/compress.c
+++ b/sound/soc/sof/compress.c
@@ -13,6 +13,11 @@
#include "ops.h"
#include "probe.h"
+struct snd_compr_ops sof_probe_compressed_ops = {
+ .copy = sof_probe_compr_copy,
+};
+EXPORT_SYMBOL(sof_probe_compressed_ops);
+
int sof_probe_compr_open(struct snd_compr_stream *cstream,
struct snd_soc_dai *dai)
{
diff --git a/sound/soc/sof/compress.h b/sound/soc/sof/compress.h
index dccc9e008f81..800f163603e1 100644
--- a/sound/soc/sof/compress.h
+++ b/sound/soc/sof/compress.h
@@ -13,6 +13,8 @@
#include <sound/compress_driver.h>
+extern struct snd_compr_ops sof_probe_compressed_ops;
+
int sof_probe_compr_open(struct snd_compr_stream *cstream,
struct snd_soc_dai *dai);
int sof_probe_compr_free(struct snd_compr_stream *cstream,
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 9c6e3f990ee3..ed5e7d2c0d43 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -399,6 +399,19 @@ static const struct snd_soc_dai_ops hda_link_dai_ops = {
.trigger = hda_link_pcm_trigger,
.prepare = hda_link_pcm_prepare,
};
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+#include "../compress.h"
+
+static struct snd_soc_cdai_ops sof_probe_compr_ops = {
+ .startup = sof_probe_compr_open,
+ .shutdown = sof_probe_compr_free,
+ .set_params = sof_probe_compr_set_params,
+ .trigger = sof_probe_compr_trigger,
+ .pointer = sof_probe_compr_pointer,
+};
+
+#endif
#endif
/*
@@ -460,5 +473,20 @@ struct snd_soc_dai_driver skl_dai[] = {
.name = "Alt Analog CPU DAI",
.ops = &hda_link_dai_ops,
},
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+{
+ .name = "Probe Extraction CPU DAI",
+ .compress_new = snd_soc_new_compress,
+ .cops = &sof_probe_compr_ops,
+ .capture = {
+ .stream_name = "Probe Extraction",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ },
+},
+#endif
#endif
};
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index ca44ecb76534..537c0a930a15 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -349,7 +349,13 @@
/* Number of DAIs */
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
+
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES)
+#define SOF_SKL_NUM_DAIS 16
+#else
#define SOF_SKL_NUM_DAIS 15
+#endif
+
#else
#define SOF_SKL_NUM_DAIS 8
#endif
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index db3df02c7398..b239bbff4b5c 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -16,6 +16,9 @@
#include "sof-priv.h"
#include "sof-audio.h"
#include "ops.h"
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+#include "compress.h"
+#endif
/* Create DMA buffer page table for DSP */
static int create_page_table(struct snd_soc_component *component,
@@ -787,6 +790,10 @@ void snd_sof_new_platform_drv(struct snd_sof_dev *sdev)
#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMPRESS)
pd->compr_ops = &sof_compressed_ops;
+#endif
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+ /* override cops when probe support is enabled */
+ pd->compr_ops = &sof_probe_compressed_ops;
#endif
pd->pcm_construct = sof_pcm_new;
pd->ignore_machine = drv_name;
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: Provide probe debugfs support" to the asoc tree
@ 2020-02-19 0:10 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:10 UTC (permalink / raw)
To: Cezary Rojewski; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: SOF: Provide probe debugfs support
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 394695f410c1cd3208906451ba5420f45c420a51 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Tue, 18 Feb 2020 15:39:23 +0100
Subject: [PATCH] ASoC: SOF: Provide probe debugfs support
Define debugfs subdirectory delegated for IPC communication with DSP.
Input format: uint,uint,(...) which are later translated into DWORDS
sequence and further into instances of struct of interest given the IPC
type.
For Extractor probes, following have been enabled:
- PROBE_POINT_ADD (echo <..> probe_points)
- PROBE_POINT_REMOVE (echo <..> probe_points_remove)
- PROBE_POINT_INFO (cat probe_points)
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200218143924.10565-9-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/debug.c | 226 ++++++++++++++++++++++++++++++++++++++++++
1 file changed, 226 insertions(+)
diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c
index d2b3b99d3a20..b5c0d6cf72cc 100644
--- a/sound/soc/sof/debug.c
+++ b/sound/soc/sof/debug.c
@@ -17,6 +17,221 @@
#include "sof-priv.h"
#include "ops.h"
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+#include "probe.h"
+
+/**
+ * strsplit_u32 - Split string into sequence of u32 tokens
+ * @buf: String to split into tokens.
+ * @delim: String containing delimiter characters.
+ * @tkns: Returned u32 sequence pointer.
+ * @num_tkns: Returned number of tokens obtained.
+ */
+static int
+strsplit_u32(char **buf, const char *delim, u32 **tkns, size_t *num_tkns)
+{
+ char *s;
+ u32 *data, *tmp;
+ size_t count = 0;
+ size_t cap = 32;
+ int ret = 0;
+
+ *tkns = NULL;
+ *num_tkns = 0;
+ data = kcalloc(cap, sizeof(*data), GFP_KERNEL);
+ if (!data)
+ return -ENOMEM;
+
+ while ((s = strsep(buf, delim)) != NULL) {
+ ret = kstrtouint(s, 0, data + count);
+ if (ret)
+ goto exit;
+ if (++count >= cap) {
+ cap *= 2;
+ tmp = krealloc(data, cap * sizeof(*data), GFP_KERNEL);
+ if (!tmp) {
+ ret = -ENOMEM;
+ goto exit;
+ }
+ data = tmp;
+ }
+ }
+
+ if (!count)
+ goto exit;
+ *tkns = kmemdup(data, count * sizeof(*data), GFP_KERNEL);
+ if (*tkns == NULL) {
+ ret = -ENOMEM;
+ goto exit;
+ }
+ *num_tkns = count;
+
+exit:
+ kfree(data);
+ return ret;
+}
+
+static int tokenize_input(const char __user *from, size_t count,
+ loff_t *ppos, u32 **tkns, size_t *num_tkns)
+{
+ char *buf;
+ int ret;
+
+ buf = kmalloc(count + 1, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ ret = simple_write_to_buffer(buf, count, ppos, from, count);
+ if (ret != count) {
+ ret = ret >= 0 ? -EIO : ret;
+ goto exit;
+ }
+
+ buf[count] = '\0';
+ ret = strsplit_u32((char **)&buf, ",", tkns, num_tkns);
+exit:
+ kfree(buf);
+ return ret;
+}
+
+static ssize_t probe_points_read(struct file *file,
+ char __user *to, size_t count, loff_t *ppos)
+{
+ struct snd_sof_dfsentry *dfse = file->private_data;
+ struct snd_sof_dev *sdev = dfse->sdev;
+ struct sof_probe_point_desc *desc;
+ size_t num_desc, len = 0;
+ char *buf;
+ int i, ret;
+
+ if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) {
+ dev_warn(sdev->dev, "no extractor stream running\n");
+ return -ENOENT;
+ }
+
+ buf = kzalloc(PAGE_SIZE, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc);
+ if (ret < 0)
+ goto exit;
+
+ for (i = 0; i < num_desc; i++) {
+ ret = snprintf(buf + len, PAGE_SIZE - len,
+ "Id: %#010x Purpose: %d Node id: %#x\n",
+ desc[i].buffer_id, desc[i].purpose, desc[i].stream_tag);
+ if (ret < 0)
+ goto free_desc;
+ len += ret;
+ }
+
+ ret = simple_read_from_buffer(to, count, ppos, buf, len);
+free_desc:
+ kfree(desc);
+exit:
+ kfree(buf);
+ return ret;
+}
+
+static ssize_t probe_points_write(struct file *file,
+ const char __user *from, size_t count, loff_t *ppos)
+{
+ struct snd_sof_dfsentry *dfse = file->private_data;
+ struct snd_sof_dev *sdev = dfse->sdev;
+ struct sof_probe_point_desc *desc;
+ size_t num_tkns, bytes;
+ u32 *tkns;
+ int ret;
+
+ if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) {
+ dev_warn(sdev->dev, "no extractor stream running\n");
+ return -ENOENT;
+ }
+
+ ret = tokenize_input(from, count, ppos, &tkns, &num_tkns);
+ if (ret < 0)
+ return ret;
+ bytes = sizeof(*tkns) * num_tkns;
+ if (!num_tkns || (bytes % sizeof(*desc))) {
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ desc = (struct sof_probe_point_desc *)tkns;
+ ret = sof_ipc_probe_points_add(sdev,
+ desc, bytes / sizeof(*desc));
+ if (!ret)
+ ret = count;
+exit:
+ kfree(tkns);
+ return ret;
+}
+
+static const struct file_operations probe_points_fops = {
+ .open = simple_open,
+ .read = probe_points_read,
+ .write = probe_points_write,
+ .llseek = default_llseek,
+};
+
+static ssize_t probe_points_remove_write(struct file *file,
+ const char __user *from, size_t count, loff_t *ppos)
+{
+ struct snd_sof_dfsentry *dfse = file->private_data;
+ struct snd_sof_dev *sdev = dfse->sdev;
+ size_t num_tkns;
+ u32 *tkns;
+ int ret;
+
+ if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) {
+ dev_warn(sdev->dev, "no extractor stream running\n");
+ return -ENOENT;
+ }
+
+ ret = tokenize_input(from, count, ppos, &tkns, &num_tkns);
+ if (ret < 0)
+ return ret;
+ if (!num_tkns) {
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ ret = sof_ipc_probe_points_remove(sdev, tkns, num_tkns);
+ if (!ret)
+ ret = count;
+exit:
+ kfree(tkns);
+ return ret;
+}
+
+static const struct file_operations probe_points_remove_fops = {
+ .open = simple_open,
+ .write = probe_points_remove_write,
+ .llseek = default_llseek,
+};
+
+static int snd_sof_debugfs_probe_item(struct snd_sof_dev *sdev,
+ const char *name, mode_t mode,
+ const struct file_operations *fops)
+{
+ struct snd_sof_dfsentry *dfse;
+
+ dfse = devm_kzalloc(sdev->dev, sizeof(*dfse), GFP_KERNEL);
+ if (!dfse)
+ return -ENOMEM;
+
+ dfse->type = SOF_DFSENTRY_TYPE_BUF;
+ dfse->sdev = sdev;
+
+ debugfs_create_file(name, mode, sdev->debugfs_root, dfse, fops);
+ /* add to dfsentry list */
+ list_add(&dfse->list, &sdev->dfsentry_list);
+
+ return 0;
+}
+#endif
+
#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST)
#define MAX_IPC_FLOOD_DURATION_MS 1000
#define MAX_IPC_FLOOD_COUNT 10000
@@ -436,6 +651,17 @@ int snd_sof_dbg_init(struct snd_sof_dev *sdev)
return err;
}
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+ err = snd_sof_debugfs_probe_item(sdev, "probe_points",
+ 0644, &probe_points_fops);
+ if (err < 0)
+ return err;
+ err = snd_sof_debugfs_probe_item(sdev, "probe_points_remove",
+ 0200, &probe_points_remove_fops);
+ if (err < 0)
+ return err;
+#endif
+
#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST)
/* create read-write ipc_flood_count debugfs entry */
err = snd_sof_debugfs_buf_item(sdev, NULL, 0,
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: Generic probe compress operations" to the asoc tree
@ 2020-02-19 0:10 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:10 UTC (permalink / raw)
To: Cezary Rojewski; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: SOF: Generic probe compress operations
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From e145e9af231adff081e0e16e1dacfb6e3c4e968f Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Tue, 18 Feb 2020 15:39:20 +0100
Subject: [PATCH] ASoC: SOF: Generic probe compress operations
Define system-agnostic probe compress flow which serves as a base for
actual, hardware-dependent implementations.
As per firmware spec, maximum of one extraction stream is allowed, while
for injection, there can be plenty.
Apart from probe_pointer, all probe compress operations are mandatory.
Copy operation is defined as unified as its flow should be shared across
all SOF systems.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200218143924.10565-6-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/Kconfig | 1 +
sound/soc/sof/Makefile | 2 +-
sound/soc/sof/compress.c | 141 +++++++++++++++++++++++++++++++++++++++
sound/soc/sof/compress.h | 29 ++++++++
sound/soc/sof/core.c | 6 ++
sound/soc/sof/ops.h | 43 ++++++++++++
sound/soc/sof/sof-priv.h | 25 +++++++
7 files changed, 246 insertions(+), 1 deletion(-)
create mode 100644 sound/soc/sof/compress.c
create mode 100644 sound/soc/sof/compress.h
diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig
index 65c3cfbcb812..4dda4b62509f 100644
--- a/sound/soc/sof/Kconfig
+++ b/sound/soc/sof/Kconfig
@@ -43,6 +43,7 @@ config SND_SOC_SOF_OF
config SND_SOC_SOF_DEBUG_PROBES
bool "SOF enable data probing"
+ select SND_SOC_COMPRESS
help
This option enables the data probing feature that can be used to
gather data directly from specific points of the audio pipeline.
diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile
index 18d7cab9046e..8eca2f85c90e 100644
--- a/sound/soc/sof/Makefile
+++ b/sound/soc/sof/Makefile
@@ -2,7 +2,7 @@
snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\
control.o trace.o utils.o sof-audio.o
-snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o
+snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o compress.o
snd-sof-pci-objs := sof-pci-dev.o
snd-sof-acpi-objs := sof-acpi-dev.o
diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c
new file mode 100644
index 000000000000..e87cc81a0599
--- /dev/null
+++ b/sound/soc/sof/compress.c
@@ -0,0 +1,141 @@
+// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause)
+//
+// This file is provided under a dual BSD/GPLv2 license. When using or
+// redistributing this file, you may do so under either license.
+//
+// Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+//
+// Author: Cezary Rojewski <cezary.rojewski@intel.com>
+//
+
+#include <sound/soc.h>
+#include "compress.h"
+#include "ops.h"
+#include "probe.h"
+
+int sof_probe_compr_open(struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_sof_dev *sdev =
+ snd_soc_component_get_drvdata(dai->component);
+ int ret;
+
+ ret = snd_sof_probe_compr_assign(sdev, cstream, dai);
+ if (ret < 0) {
+ dev_err(dai->dev, "Failed to assign probe stream: %d\n", ret);
+ return ret;
+ }
+
+ sdev->extractor_stream_tag = ret;
+ return 0;
+}
+EXPORT_SYMBOL(sof_probe_compr_open);
+
+int sof_probe_compr_free(struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_sof_dev *sdev =
+ snd_soc_component_get_drvdata(dai->component);
+ struct sof_probe_point_desc *desc;
+ size_t num_desc;
+ int i, ret;
+
+ /* disconnect all probe points */
+ ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc);
+ if (ret < 0) {
+ dev_err(dai->dev, "Failed to get probe points: %d\n", ret);
+ goto exit;
+ }
+
+ for (i = 0; i < num_desc; i++)
+ sof_ipc_probe_points_remove(sdev, &desc[i].buffer_id, 1);
+ kfree(desc);
+
+exit:
+ ret = sof_ipc_probe_deinit(sdev);
+ if (ret < 0)
+ dev_err(dai->dev, "Failed to deinit probe: %d\n", ret);
+
+ sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID;
+ snd_compr_free_pages(cstream);
+
+ return snd_sof_probe_compr_free(sdev, cstream, dai);
+}
+EXPORT_SYMBOL(sof_probe_compr_free);
+
+int sof_probe_compr_set_params(struct snd_compr_stream *cstream,
+ struct snd_compr_params *params, struct snd_soc_dai *dai)
+{
+ struct snd_compr_runtime *rtd = cstream->runtime;
+ struct snd_sof_dev *sdev =
+ snd_soc_component_get_drvdata(dai->component);
+ int ret;
+
+ cstream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_SG;
+ cstream->dma_buffer.dev.dev = sdev->dev;
+ ret = snd_compr_malloc_pages(cstream, rtd->buffer_size);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_sof_probe_compr_set_params(sdev, cstream, params, dai);
+ if (ret < 0)
+ return ret;
+
+ ret = sof_ipc_probe_init(sdev, sdev->extractor_stream_tag,
+ rtd->dma_bytes);
+ if (ret < 0) {
+ dev_err(dai->dev, "Failed to init probe: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL(sof_probe_compr_set_params);
+
+int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_sof_dev *sdev =
+ snd_soc_component_get_drvdata(dai->component);
+
+ return snd_sof_probe_compr_trigger(sdev, cstream, cmd, dai);
+}
+EXPORT_SYMBOL(sof_probe_compr_trigger);
+
+int sof_probe_compr_pointer(struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai)
+{
+ struct snd_sof_dev *sdev =
+ snd_soc_component_get_drvdata(dai->component);
+
+ return snd_sof_probe_compr_pointer(sdev, cstream, tstamp, dai);
+}
+EXPORT_SYMBOL(sof_probe_compr_pointer);
+
+int sof_probe_compr_copy(struct snd_compr_stream *cstream,
+ char __user *buf, size_t count)
+{
+ struct snd_compr_runtime *rtd = cstream->runtime;
+ unsigned int offset, n;
+ void *ptr;
+ int ret;
+
+ if (count > rtd->buffer_size)
+ count = rtd->buffer_size;
+
+ div_u64_rem(rtd->total_bytes_transferred, rtd->buffer_size, &offset);
+ ptr = rtd->dma_area + offset;
+ n = rtd->buffer_size - offset;
+
+ if (count < n) {
+ ret = copy_to_user(buf, ptr, count);
+ } else {
+ ret = copy_to_user(buf, ptr, n);
+ ret += copy_to_user(buf + n, rtd->dma_area, count - n);
+ }
+
+ if (ret)
+ return count - ret;
+ return count;
+}
+EXPORT_SYMBOL(sof_probe_compr_copy);
diff --git a/sound/soc/sof/compress.h b/sound/soc/sof/compress.h
new file mode 100644
index 000000000000..dccc9e008f81
--- /dev/null
+++ b/sound/soc/sof/compress.h
@@ -0,0 +1,29 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/*
+ * This file is provided under a dual BSD/GPLv2 license. When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+ *
+ * Author: Cezary Rojewski <cezary.rojewski@intel.com>
+ */
+
+#ifndef __SOF_COMPRESS_H
+#define __SOF_COMPRESS_H
+
+#include <sound/compress_driver.h>
+
+int sof_probe_compr_open(struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai);
+int sof_probe_compr_free(struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai);
+int sof_probe_compr_set_params(struct snd_compr_stream *cstream,
+ struct snd_compr_params *params, struct snd_soc_dai *dai);
+int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd,
+ struct snd_soc_dai *dai);
+int sof_probe_compr_pointer(struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai);
+int sof_probe_compr_copy(struct snd_compr_stream *cstream,
+ char __user *buf, size_t count);
+
+#endif
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 1d07450aff77..91acfae7935c 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -14,6 +14,9 @@
#include <sound/sof.h>
#include "sof-priv.h"
#include "ops.h"
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+#include "probe.h"
+#endif
/* see SOF_DBG_ flags */
int sof_core_debug;
@@ -292,6 +295,9 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data)
sdev->pdata = plat_data;
sdev->first_boot = true;
sdev->fw_state = SOF_FW_BOOT_NOT_STARTED;
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+ sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID;
+#endif
dev_set_drvdata(dev, sdev);
/* check all mandatory ops */
diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h
index 7f532bcc8e9d..a771500ac442 100644
--- a/sound/soc/sof/ops.h
+++ b/sound/soc/sof/ops.h
@@ -393,6 +393,49 @@ snd_sof_pcm_platform_pointer(struct snd_sof_dev *sdev,
return 0;
}
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+static inline int
+snd_sof_probe_compr_assign(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream, struct snd_soc_dai *dai)
+{
+ return sof_ops(sdev)->probe_assign(sdev, cstream, dai);
+}
+
+static inline int
+snd_sof_probe_compr_free(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream, struct snd_soc_dai *dai)
+{
+ return sof_ops(sdev)->probe_free(sdev, cstream, dai);
+}
+
+static inline int
+snd_sof_probe_compr_set_params(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_compr_params *params, struct snd_soc_dai *dai)
+{
+ return sof_ops(sdev)->probe_set_params(sdev, cstream, params, dai);
+}
+
+static inline int
+snd_sof_probe_compr_trigger(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ return sof_ops(sdev)->probe_trigger(sdev, cstream, cmd, dai);
+}
+
+static inline int
+snd_sof_probe_compr_pointer(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai)
+{
+ if (sof_ops(sdev) && sof_ops(sdev)->probe_pointer)
+ return sof_ops(sdev)->probe_pointer(sdev, cstream, tstamp, dai);
+
+ return 0;
+}
+#endif
+
/* machine driver */
static inline int
snd_sof_machine_register(struct snd_sof_dev *sdev, void *pdata)
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index 00084471d0de..5d16f668d16a 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -170,6 +170,27 @@ struct snd_sof_dsp_ops {
snd_pcm_uframes_t (*pcm_pointer)(struct snd_sof_dev *sdev,
struct snd_pcm_substream *substream); /* optional */
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+ /* Except for probe_pointer, all probe ops are mandatory */
+ int (*probe_assign)(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai); /* mandatory */
+ int (*probe_free)(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_soc_dai *dai); /* mandatory */
+ int (*probe_set_params)(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_compr_params *params,
+ struct snd_soc_dai *dai); /* mandatory */
+ int (*probe_trigger)(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream, int cmd,
+ struct snd_soc_dai *dai); /* mandatory */
+ int (*probe_pointer)(struct snd_sof_dev *sdev,
+ struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *tstamp,
+ struct snd_soc_dai *dai); /* optional */
+#endif
+
/* host read DSP stream data */
void (*ipc_msg_data)(struct snd_sof_dev *sdev,
struct snd_pcm_substream *substream,
@@ -405,6 +426,10 @@ struct snd_sof_dev {
wait_queue_head_t waitq;
int code_loading;
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES)
+ unsigned int extractor_stream_tag;
+#endif
+
/* DMA for Trace */
struct snd_dma_buffer dmatb;
struct snd_dma_buffer dmatp;
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: Implement Probe IPC API" to the asoc tree
@ 2020-02-19 0:10 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:10 UTC (permalink / raw)
To: Cezary Rojewski; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: SOF: Implement Probe IPC API
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From f3b433e4699fa358ce5b7bd7688bebe36068c199 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Tue, 18 Feb 2020 15:39:19 +0100
Subject: [PATCH] ASoC: SOF: Implement Probe IPC API
Add all required types and methods to support each and every request
that driver could sent to firmware. Probe is one of SOF firmware
features which allows for data extraction and injection directly from
or to DMA stream.
Exposes eight IPCs:
- addition and removal of injection DMAs
- addition and removal of probe points
- info retrieval of injection DMAs and probe points
- probe initialization and cleanup
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200218143924.10565-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
| 11 ++
sound/soc/sof/Kconfig | 8 +
sound/soc/sof/Makefile | 1 +
sound/soc/sof/intel/hda-ipc.c | 4 +-
sound/soc/sof/probe.c | 286 ++++++++++++++++++++++++++++++++++
sound/soc/sof/probe.h | 85 ++++++++++
6 files changed, 394 insertions(+), 1 deletion(-)
create mode 100644 sound/soc/sof/probe.c
create mode 100644 sound/soc/sof/probe.h
--git a/include/sound/sof/header.h b/include/sound/sof/header.h
index bf3edd9c08b4..b79479575cc8 100644
--- a/include/sound/sof/header.h
+++ b/include/sound/sof/header.h
@@ -51,6 +51,7 @@
#define SOF_IPC_GLB_TRACE_MSG SOF_GLB_TYPE(0x9U)
#define SOF_IPC_GLB_GDB_DEBUG SOF_GLB_TYPE(0xAU)
#define SOF_IPC_GLB_TEST_MSG SOF_GLB_TYPE(0xBU)
+#define SOF_IPC_GLB_PROBE SOF_GLB_TYPE(0xCU)
/*
* DSP Command Message Types
@@ -102,6 +103,16 @@
#define SOF_IPC_STREAM_VORBIS_PARAMS SOF_CMD_TYPE(0x010)
#define SOF_IPC_STREAM_VORBIS_FREE SOF_CMD_TYPE(0x011)
+/* probe */
+#define SOF_IPC_PROBE_INIT SOF_CMD_TYPE(0x001)
+#define SOF_IPC_PROBE_DEINIT SOF_CMD_TYPE(0x002)
+#define SOF_IPC_PROBE_DMA_ADD SOF_CMD_TYPE(0x003)
+#define SOF_IPC_PROBE_DMA_INFO SOF_CMD_TYPE(0x004)
+#define SOF_IPC_PROBE_DMA_REMOVE SOF_CMD_TYPE(0x005)
+#define SOF_IPC_PROBE_POINT_ADD SOF_CMD_TYPE(0x006)
+#define SOF_IPC_PROBE_POINT_INFO SOF_CMD_TYPE(0x007)
+#define SOF_IPC_PROBE_POINT_REMOVE SOF_CMD_TYPE(0x008)
+
/* trace */
#define SOF_IPC_TRACE_DMA_PARAMS SOF_CMD_TYPE(0x001)
#define SOF_IPC_TRACE_DMA_POSITION SOF_CMD_TYPE(0x002)
diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig
index 827b0ec92522..65c3cfbcb812 100644
--- a/sound/soc/sof/Kconfig
+++ b/sound/soc/sof/Kconfig
@@ -41,6 +41,14 @@ config SND_SOC_SOF_OF
required to enable i.MX8 devices.
Say Y if you need this option. If unsure select "N".
+config SND_SOC_SOF_DEBUG_PROBES
+ bool "SOF enable data probing"
+ help
+ This option enables the data probing feature that can be used to
+ gather data directly from specific points of the audio pipeline.
+ Say Y if you want to enable probes.
+ If unsure, select "N".
+
config SND_SOC_SOF_DEVELOPER_SUPPORT
bool "SOF developer options support"
depends on EXPERT
diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile
index 0a8bc72c28a5..18d7cab9046e 100644
--- a/sound/soc/sof/Makefile
+++ b/sound/soc/sof/Makefile
@@ -2,6 +2,7 @@
snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\
control.o trace.o utils.o sof-audio.o
+snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o
snd-sof-pci-objs := sof-pci-dev.o
snd-sof-acpi-objs := sof-acpi-dev.o
diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c
index 1837f66e361f..922052883b0a 100644
--- a/sound/soc/sof/intel/hda-ipc.c
+++ b/sound/soc/sof/intel/hda-ipc.c
@@ -106,7 +106,9 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev)
ret = reply.error;
} else {
/* reply correct size ? */
- if (reply.hdr.size != msg->reply_size) {
+ if (reply.hdr.size != msg->reply_size &&
+ /* getter payload is never known upfront */
+ !(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) {
dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n",
msg->reply_size, reply.hdr.size);
ret = -EINVAL;
diff --git a/sound/soc/sof/probe.c b/sound/soc/sof/probe.c
new file mode 100644
index 000000000000..2b2f3dcfc7e9
--- /dev/null
+++ b/sound/soc/sof/probe.c
@@ -0,0 +1,286 @@
+// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause)
+//
+// This file is provided under a dual BSD/GPLv2 license. When using or
+// redistributing this file, you may do so under either license.
+//
+// Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+//
+// Author: Cezary Rojewski <cezary.rojewski@intel.com>
+//
+
+#include "sof-priv.h"
+#include "probe.h"
+
+/**
+ * sof_ipc_probe_init - initialize data probing
+ * @sdev: SOF sound device
+ * @stream_tag: Extractor stream tag
+ * @buffer_size: DMA buffer size to set for extractor
+ *
+ * Host chooses whether extraction is supported or not by providing
+ * valid stream tag to DSP. Once specified, stream described by that
+ * tag will be tied to DSP for extraction for the entire lifetime of
+ * probe.
+ *
+ * Probing is initialized only once and each INIT request must be
+ * matched by DEINIT call.
+ */
+int sof_ipc_probe_init(struct snd_sof_dev *sdev,
+ u32 stream_tag, size_t buffer_size)
+{
+ struct sof_ipc_probe_dma_add_params *msg;
+ struct sof_ipc_reply reply;
+ size_t size = struct_size(msg, dma, 1);
+ int ret;
+
+ msg = kmalloc(size, GFP_KERNEL);
+ if (!msg)
+ return -ENOMEM;
+ msg->hdr.size = size;
+ msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_INIT;
+ msg->num_elems = 1;
+ msg->dma[0].stream_tag = stream_tag;
+ msg->dma[0].dma_buffer_size = buffer_size;
+
+ ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+ &reply, sizeof(reply));
+ kfree(msg);
+ return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_init);
+
+/**
+ * sof_ipc_probe_deinit - cleanup after data probing
+ * @sdev: SOF sound device
+ *
+ * Host sends DEINIT request to free previously initialized probe
+ * on DSP side once it is no longer needed. DEINIT only when there
+ * are no probes connected and with all injectors detached.
+ */
+int sof_ipc_probe_deinit(struct snd_sof_dev *sdev)
+{
+ struct sof_ipc_cmd_hdr msg;
+ struct sof_ipc_reply reply;
+
+ msg.size = sizeof(msg);
+ msg.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DEINIT;
+
+ return sof_ipc_tx_message(sdev->ipc, msg.cmd, &msg, msg.size,
+ &reply, sizeof(reply));
+}
+EXPORT_SYMBOL(sof_ipc_probe_deinit);
+
+static int sof_ipc_probe_info(struct snd_sof_dev *sdev, unsigned int cmd,
+ void **params, size_t *num_params)
+{
+ struct sof_ipc_probe_info_params msg = {{{0}}};
+ struct sof_ipc_probe_info_params *reply;
+ size_t bytes;
+ int ret;
+
+ *params = NULL;
+ *num_params = 0;
+
+ reply = kzalloc(SOF_IPC_MSG_MAX_SIZE, GFP_KERNEL);
+ if (!reply)
+ return -ENOMEM;
+ msg.rhdr.hdr.size = sizeof(msg);
+ msg.rhdr.hdr.cmd = SOF_IPC_GLB_PROBE | cmd;
+
+ ret = sof_ipc_tx_message(sdev->ipc, msg.rhdr.hdr.cmd, &msg,
+ msg.rhdr.hdr.size, reply, SOF_IPC_MSG_MAX_SIZE);
+ if (ret < 0 || reply->rhdr.error < 0)
+ goto exit;
+
+ if (!reply->num_elems)
+ goto exit;
+
+ bytes = reply->num_elems * sizeof(reply->dma[0]);
+ *params = kmemdup(&reply->dma[0], bytes, GFP_KERNEL);
+ if (!*params) {
+ ret = -ENOMEM;
+ goto exit;
+ }
+ *num_params = msg.num_elems;
+
+exit:
+ kfree(reply);
+ return ret;
+}
+
+/**
+ * sof_ipc_probe_dma_info - retrieve list of active injection dmas
+ * @sdev: SOF sound device
+ * @dma: Returned list of active dmas
+ * @num_dma: Returned count of active dmas
+ *
+ * Host sends DMA_INFO request to obtain list of injection dmas it
+ * can use to transfer data over with.
+ *
+ * Note that list contains only injection dmas as there is only one
+ * extractor (dma) and it is always assigned on probing init.
+ * DSP knows exactly where data from extraction probes is going to,
+ * which is not the case for injection where multiple streams
+ * could be engaged.
+ */
+int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev,
+ struct sof_probe_dma **dma, size_t *num_dma)
+{
+ return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_DMA_INFO,
+ (void **)dma, num_dma);
+}
+EXPORT_SYMBOL(sof_ipc_probe_dma_info);
+
+/**
+ * sof_ipc_probe_dma_add - attach to specified dmas
+ * @sdev: SOF sound device
+ * @dma: List of streams (dmas) to attach to
+ * @num_dma: Number of elements in @dma
+ *
+ * Contrary to extraction, injection streams are never assigned
+ * on init. Before attempting any data injection, host is responsible
+ * for specifying streams which will be later used to transfer data
+ * to connected probe points.
+ */
+int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev,
+ struct sof_probe_dma *dma, size_t num_dma)
+{
+ struct sof_ipc_probe_dma_add_params *msg;
+ struct sof_ipc_reply reply;
+ size_t size = struct_size(msg, dma, num_dma);
+ int ret;
+
+ msg = kmalloc(size, GFP_KERNEL);
+ if (!msg)
+ return -ENOMEM;
+ msg->hdr.size = size;
+ msg->num_elems = num_dma;
+ msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_ADD;
+ memcpy(&msg->dma[0], dma, size - sizeof(*msg));
+
+ ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+ &reply, sizeof(reply));
+ kfree(msg);
+ return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_dma_add);
+
+/**
+ * sof_ipc_probe_dma_remove - detach from specified dmas
+ * @sdev: SOF sound device
+ * @stream_tag: List of stream tags to detach from
+ * @num_stream_tag: Number of elements in @stream_tag
+ *
+ * Host sends DMA_REMOVE request to free previously attached stream
+ * from being occupied for injection. Each detach operation should
+ * match equivalent DMA_ADD. Detach only when all probes tied to
+ * given stream have been disconnected.
+ */
+int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev,
+ unsigned int *stream_tag, size_t num_stream_tag)
+{
+ struct sof_ipc_probe_dma_remove_params *msg;
+ struct sof_ipc_reply reply;
+ size_t size = struct_size(msg, stream_tag, num_stream_tag);
+ int ret;
+
+ msg = kmalloc(size, GFP_KERNEL);
+ if (!msg)
+ return -ENOMEM;
+ msg->hdr.size = size;
+ msg->num_elems = num_stream_tag;
+ msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_REMOVE;
+ memcpy(&msg->stream_tag[0], stream_tag, size - sizeof(*msg));
+
+ ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+ &reply, sizeof(reply));
+ kfree(msg);
+ return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_dma_remove);
+
+/**
+ * sof_ipc_probe_points_info - retrieve list of active probe points
+ * @sdev: SOF sound device
+ * @desc: Returned list of active probes
+ * @num_desc: Returned count of active probes
+ *
+ * Host sends PROBE_POINT_INFO request to obtain list of active probe
+ * points, valid for disconnection when given probe is no longer
+ * required.
+ */
+int sof_ipc_probe_points_info(struct snd_sof_dev *sdev,
+ struct sof_probe_point_desc **desc, size_t *num_desc)
+{
+ return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_POINT_INFO,
+ (void **)desc, num_desc);
+}
+EXPORT_SYMBOL(sof_ipc_probe_points_info);
+
+/**
+ * sof_ipc_probe_points_add - connect specified probes
+ * @sdev: SOF sound device
+ * @desc: List of probe points to connect
+ * @num_desc: Number of elements in @desc
+ *
+ * Dynamically connects to provided set of endpoints. Immediately
+ * after connection is established, host must be prepared to
+ * transfer data from or to target stream given the probing purpose.
+ *
+ * Each probe point should be removed using PROBE_POINT_REMOVE
+ * request when no longer needed.
+ */
+int sof_ipc_probe_points_add(struct snd_sof_dev *sdev,
+ struct sof_probe_point_desc *desc, size_t num_desc)
+{
+ struct sof_ipc_probe_point_add_params *msg;
+ struct sof_ipc_reply reply;
+ size_t size = struct_size(msg, desc, num_desc);
+ int ret;
+
+ msg = kmalloc(size, GFP_KERNEL);
+ if (!msg)
+ return -ENOMEM;
+ msg->hdr.size = size;
+ msg->num_elems = num_desc;
+ msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_ADD;
+ memcpy(&msg->desc[0], desc, size - sizeof(*msg));
+
+ ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+ &reply, sizeof(reply));
+ kfree(msg);
+ return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_points_add);
+
+/**
+ * sof_ipc_probe_points_remove - disconnect specified probes
+ * @sdev: SOF sound device
+ * @buffer_id: List of probe points to disconnect
+ * @num_buffer_id: Number of elements in @desc
+ *
+ * Removes previously connected probes from list of active probe
+ * points and frees all resources on DSP side.
+ */
+int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev,
+ unsigned int *buffer_id, size_t num_buffer_id)
+{
+ struct sof_ipc_probe_point_remove_params *msg;
+ struct sof_ipc_reply reply;
+ size_t size = struct_size(msg, buffer_id, num_buffer_id);
+ int ret;
+
+ msg = kmalloc(size, GFP_KERNEL);
+ if (!msg)
+ return -ENOMEM;
+ msg->hdr.size = size;
+ msg->num_elems = num_buffer_id;
+ msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_REMOVE;
+ memcpy(&msg->buffer_id[0], buffer_id, size - sizeof(*msg));
+
+ ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size,
+ &reply, sizeof(reply));
+ kfree(msg);
+ return ret;
+}
+EXPORT_SYMBOL(sof_ipc_probe_points_remove);
diff --git a/sound/soc/sof/probe.h b/sound/soc/sof/probe.h
new file mode 100644
index 000000000000..45daa5552834
--- /dev/null
+++ b/sound/soc/sof/probe.h
@@ -0,0 +1,85 @@
+/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */
+/*
+ * This file is provided under a dual BSD/GPLv2 license. When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * Copyright(c) 2019-2020 Intel Corporation. All rights reserved.
+ *
+ * Author: Cezary Rojewski <cezary.rojewski@intel.com>
+ */
+
+#ifndef __SOF_PROBE_H
+#define __SOF_PROBE_H
+
+#include <sound/sof/header.h>
+
+struct snd_sof_dev;
+
+#define SOF_PROBE_INVALID_NODE_ID UINT_MAX
+
+struct sof_probe_dma {
+ unsigned int stream_tag;
+ unsigned int dma_buffer_size;
+} __packed;
+
+enum sof_connection_purpose {
+ SOF_CONNECTION_PURPOSE_EXTRACT = 1,
+ SOF_CONNECTION_PURPOSE_INJECT,
+};
+
+struct sof_probe_point_desc {
+ unsigned int buffer_id;
+ unsigned int purpose;
+ unsigned int stream_tag;
+} __packed;
+
+struct sof_ipc_probe_dma_add_params {
+ struct sof_ipc_cmd_hdr hdr;
+ unsigned int num_elems;
+ struct sof_probe_dma dma[0];
+} __packed;
+
+struct sof_ipc_probe_info_params {
+ struct sof_ipc_reply rhdr;
+ unsigned int num_elems;
+ union {
+ struct sof_probe_dma dma[0];
+ struct sof_probe_point_desc desc[0];
+ };
+} __packed;
+
+struct sof_ipc_probe_dma_remove_params {
+ struct sof_ipc_cmd_hdr hdr;
+ unsigned int num_elems;
+ unsigned int stream_tag[0];
+} __packed;
+
+struct sof_ipc_probe_point_add_params {
+ struct sof_ipc_cmd_hdr hdr;
+ unsigned int num_elems;
+ struct sof_probe_point_desc desc[0];
+} __packed;
+
+struct sof_ipc_probe_point_remove_params {
+ struct sof_ipc_cmd_hdr hdr;
+ unsigned int num_elems;
+ unsigned int buffer_id[0];
+} __packed;
+
+int sof_ipc_probe_init(struct snd_sof_dev *sdev,
+ u32 stream_tag, size_t buffer_size);
+int sof_ipc_probe_deinit(struct snd_sof_dev *sdev);
+int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev,
+ struct sof_probe_dma **dma, size_t *num_dma);
+int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev,
+ struct sof_probe_dma *dma, size_t num_dma);
+int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev,
+ unsigned int *stream_tag, size_t num_stream_tag);
+int sof_ipc_probe_points_info(struct snd_sof_dev *sdev,
+ struct sof_probe_point_desc **desc, size_t *num_desc);
+int sof_ipc_probe_points_add(struct snd_sof_dev *sdev,
+ struct sof_probe_point_desc *desc, size_t num_desc);
+int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev,
+ unsigned int *buffer_id, size_t num_buffer_id);
+
+#endif
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ALSA: core: Implement compress page allocation and free routines" to the asoc tree
@ 2020-02-19 0:11 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:11 UTC (permalink / raw)
To: Cezary Rojewski
Cc: Divya Prakash, alsa-devel, Mark Brown, Pierre-Louis Bossart, Vinod Koul
The patch
ALSA: core: Implement compress page allocation and free routines
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From b9759ef2fd1acb0d3f3dce7991c44a4c5e9e68a3 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Tue, 18 Feb 2020 15:39:17 +0100
Subject: [PATCH] ALSA: core: Implement compress page allocation and free
routines
Add simple malloc and free methods for memory management for compress
streams. Based on snd_pcm_lib_malloc_pages and snd_pcm_lib_free_pages
implementation.
Signed-off-by: Divya Prakash <divya1.prakash@intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Vinod Koul <vkoul@kernel.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200218143924.10565-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/compress_driver.h | 5 ++++
sound/core/compress_offload.c | 42 +++++++++++++++++++++++++++++++++
2 files changed, 47 insertions(+)
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index 00f633c0c3ba..6ce8effa0b12 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -67,6 +67,7 @@ struct snd_compr_runtime {
* @metadata_set: metadata set flag, true when set
* @next_track: has userspace signal next track transition, true when set
* @private_data: pointer to DSP private data
+ * @dma_buffer: allocated buffer if any
*/
struct snd_compr_stream {
const char *name;
@@ -78,6 +79,7 @@ struct snd_compr_stream {
bool metadata_set;
bool next_track;
void *private_data;
+ struct snd_dma_buffer dma_buffer;
};
/**
@@ -212,6 +214,9 @@ snd_compr_set_runtime_buffer(struct snd_compr_stream *stream,
}
}
+int snd_compr_malloc_pages(struct snd_compr_stream *stream, size_t size);
+int snd_compr_free_pages(struct snd_compr_stream *stream);
+
int snd_compr_stop_error(struct snd_compr_stream *stream,
snd_pcm_state_t state);
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 9de1c9a0173e..509290f2efa8 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -488,6 +488,48 @@ snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg)
}
#endif /* !COMPR_CODEC_CAPS_OVERFLOW */
+int snd_compr_malloc_pages(struct snd_compr_stream *stream, size_t size)
+{
+ struct snd_dma_buffer *dmab;
+ int ret;
+
+ if (snd_BUG_ON(!(stream) || !(stream)->runtime))
+ return -EINVAL;
+ dmab = kzalloc(sizeof(*dmab), GFP_KERNEL);
+ if (!dmab)
+ return -ENOMEM;
+ dmab->dev = stream->dma_buffer.dev;
+ ret = snd_dma_alloc_pages(dmab->dev.type, dmab->dev.dev, size, dmab);
+ if (ret < 0) {
+ kfree(dmab);
+ return ret;
+ }
+
+ snd_compr_set_runtime_buffer(stream, dmab);
+ stream->runtime->dma_bytes = size;
+ return 1;
+}
+EXPORT_SYMBOL(snd_compr_malloc_pages);
+
+int snd_compr_free_pages(struct snd_compr_stream *stream)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+
+ if (snd_BUG_ON(!(stream) || !(stream)->runtime))
+ return -EINVAL;
+ if (runtime->dma_area == NULL)
+ return 0;
+ if (runtime->dma_buffer_p != &stream->dma_buffer) {
+ /* It's a newly allocated buffer. Release it now. */
+ snd_dma_free_pages(runtime->dma_buffer_p);
+ kfree(runtime->dma_buffer_p);
+ }
+
+ snd_compr_set_runtime_buffer(stream, NULL);
+ return 0;
+}
+EXPORT_SYMBOL(snd_compr_free_pages);
+
/* revisit this with snd_pcm_preallocate_xxx */
static int snd_compr_allocate_buffer(struct snd_compr_stream *stream,
struct snd_compr_params *params)
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ALSA: core: Expand DMA buffer information" to the asoc tree
@ 2020-02-19 0:11 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:11 UTC (permalink / raw)
To: Cezary Rojewski; +Cc: Vinod Koul, alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ALSA: core: Expand DMA buffer information
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 386dd54b3a2eedb91aa6e465e7c3a57db04f3960 Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Tue, 18 Feb 2020 15:39:16 +0100
Subject: [PATCH] ALSA: core: Expand DMA buffer information
Update DMA buffer definition for snd_compr_runtime so it is represented
similarly as in snd_pcm_runtime. While at it, modify
snd_compr_set_runtime_buffer to account for newly added members.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Vinod Koul <vkoul@kernel.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200218143924.10565-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/compress_driver.h | 35 ++++++++++++++++++++++++---------
1 file changed, 26 insertions(+), 9 deletions(-)
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index bc88d6f964da..00f633c0c3ba 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -23,7 +23,6 @@ struct snd_compr_ops;
* struct snd_compr_runtime: runtime stream description
* @state: stream state
* @ops: pointer to DSP callbacks
- * @dma_buffer_p: runtime dma buffer pointer
* @buffer: pointer to kernel buffer, valid only when not in mmap mode or
* DSP doesn't implement copy
* @buffer_size: size of the above buffer
@@ -34,11 +33,14 @@ struct snd_compr_ops;
* @total_bytes_transferred: cumulative bytes transferred by offload DSP
* @sleep: poll sleep
* @private_data: driver private data pointer
+ * @dma_area: virtual buffer address
+ * @dma_addr: physical buffer address (not accessible from main CPU)
+ * @dma_bytes: size of DMA area
+ * @dma_buffer_p: runtime dma buffer pointer
*/
struct snd_compr_runtime {
snd_pcm_state_t state;
struct snd_compr_ops *ops;
- struct snd_dma_buffer *dma_buffer_p;
void *buffer;
u64 buffer_size;
u32 fragment_size;
@@ -47,6 +49,11 @@ struct snd_compr_runtime {
u64 total_bytes_transferred;
wait_queue_head_t sleep;
void *private_data;
+
+ unsigned char *dma_area;
+ dma_addr_t dma_addr;
+ size_t dma_bytes;
+ struct snd_dma_buffer *dma_buffer_p;
};
/**
@@ -180,19 +187,29 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream)
/**
* snd_compr_set_runtime_buffer - Set the Compress runtime buffer
- * @substream: compress substream to set
+ * @stream: compress stream to set
* @bufp: the buffer information, NULL to clear
*
* Copy the buffer information to runtime buffer when @bufp is non-NULL.
* Otherwise it clears the current buffer information.
*/
-static inline void snd_compr_set_runtime_buffer(
- struct snd_compr_stream *substream,
- struct snd_dma_buffer *bufp)
+static inline void
+snd_compr_set_runtime_buffer(struct snd_compr_stream *stream,
+ struct snd_dma_buffer *bufp)
{
- struct snd_compr_runtime *runtime = substream->runtime;
-
- runtime->dma_buffer_p = bufp;
+ struct snd_compr_runtime *runtime = stream->runtime;
+
+ if (bufp) {
+ runtime->dma_buffer_p = bufp;
+ runtime->dma_area = bufp->area;
+ runtime->dma_addr = bufp->addr;
+ runtime->dma_bytes = bufp->bytes;
+ } else {
+ runtime->dma_buffer_p = NULL;
+ runtime->dma_area = NULL;
+ runtime->dma_addr = 0;
+ runtime->dma_bytes = 0;
+ }
}
int snd_compr_stop_error(struct snd_compr_stream *stream,
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: tas2562: Return invalid for when bitwidth is invalid" to the asoc tree
@ 2020-02-19 0:11 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 0:11 UTC (permalink / raw)
To: Dan Murphy; +Cc: alsa-devel, Mark Brown
The patch
ASoC: tas2562: Return invalid for when bitwidth is invalid
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 310006cab991fa895cc3a42c53609778b0d50d46 Mon Sep 17 00:00:00 2001
From: Dan Murphy <dmurphy@ti.com>
Date: Tue, 18 Feb 2020 12:52:52 -0600
Subject: [PATCH] ASoC: tas2562: Return invalid for when bitwidth is invalid
If the bitwidth passed in to the set_bitwidth function is not supported
then return an error.
Fixes: 29b74236bd57 ("ASoC: tas2562: Introduce the TAS2562 amplifier")
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200218185252.26290-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/tas2562.c | 3 ++-
1 file changed, 2 insertions(+), 1 deletion(-)
diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c
index 729acd874c48..5b4803a76719 100644
--- a/sound/soc/codecs/tas2562.c
+++ b/sound/soc/codecs/tas2562.c
@@ -215,7 +215,8 @@ static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth)
break;
default:
- dev_info(tas2562->dev, "Not supported params format\n");
+ dev_info(tas2562->dev, "Unsupported bitwidth format\n");
+ return -EINVAL;
}
ret = snd_soc_component_update_bits(tas2562->component,
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: meson: add t9015 internal codec binding documentation" to the asoc tree
@ 2020-02-19 20:56 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 20:56 UTC (permalink / raw)
To: Jerome Brunet; +Cc: alsa-devel, Mark Brown
The patch
ASoC: meson: add t9015 internal codec binding documentation
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 5c36abcd2621adc3d50d05628f0ef0be6e7840a9 Mon Sep 17 00:00:00 2001
From: Jerome Brunet <jbrunet@baylibre.com>
Date: Wed, 19 Feb 2020 18:35:02 +0100
Subject: [PATCH] ASoC: meson: add t9015 internal codec binding documentation
Add the DT binding documention of the internal DAC found in the Amlogic
gxl, g12a and sm1 SoC family.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200219173503.1112561-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
| 58 +++++++++++++++++++
1 file changed, 58 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/amlogic,t9015.yaml
--git a/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml
new file mode 100644
index 000000000000..b7c38c2b5b54
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml
@@ -0,0 +1,58 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/amlogic,t9015.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Amlogic T9015 Internal Audio DAC
+
+maintainers:
+ - Jerome Brunet <jbrunet@baylibre.com>
+
+properties:
+ $nodename:
+ pattern: "^audio-controller@.*"
+
+ "#sound-dai-cells":
+ const: 0
+
+ compatible:
+ items:
+ - const: amlogic,t9015
+
+ clocks:
+ items:
+ - description: Peripheral clock
+
+ clock-names:
+ items:
+ - const: pclk
+
+ reg:
+ maxItems: 1
+
+ resets:
+ maxItems: 1
+
+required:
+ - "#sound-dai-cells"
+ - compatible
+ - reg
+ - clocks
+ - clock-names
+ - resets
+
+examples:
+ - |
+ #include <dt-bindings/clock/g12a-clkc.h>
+ #include <dt-bindings/reset/amlogic,meson-g12a-reset.h>
+
+ acodec: audio-controller@32000 {
+ compatible = "amlogic,t9015";
+ reg = <0x0 0x32000 0x0 0x14>;
+ #sound-dai-cells = <0>;
+ clocks = <&clkc CLKID_AUDIO_CODEC>;
+ clock-names = "pclk";
+ resets = <&reset RESET_AUDIO_CODEC>;
+ };
+
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: dt-bindings: Make RK3328 codec GPIO explicit" to the asoc tree
@ 2020-02-19 20:57 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 20:57 UTC (permalink / raw)
To: Robin Murphy; +Cc: alsa-devel, Mark Brown
The patch
ASoC: dt-bindings: Make RK3328 codec GPIO explicit
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From e14980976534d9d94f5cddd70033707965482ede Mon Sep 17 00:00:00 2001
From: Robin Murphy <robin.murphy@arm.com>
Date: Tue, 18 Feb 2020 21:31:58 +0000
Subject: [PATCH] ASoC: dt-bindings: Make RK3328 codec GPIO explicit
Existing RK3328 codec drivers have overloaded the GRF phandle to assume
implicit control of the limited-function GPIO_MUTE pin, which is usually
used to enable an external audio line driver IC. Since this pin has a
proper binding of its own (see gpio/rockchip,rk3328-grf-gpio.txt), make
a GPIO explicit in the codec binding too. This will help avoid ambiguity
on boards that use that pin for some other purpose.
(and while touching the example, enforce the "don't include status" rule)
Signed-off-by: Robin Murphy <robin.murphy@arm.com>
Link: https://lore.kernel.org/r/5f7a399dea8a9dedef57f6f99f0f6ab1c1fdc56a.1581376744.git.robin.murphy@arm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../devicetree/bindings/sound/rockchip,rk3328-codec.txt | 7 ++++++-
1 file changed, 6 insertions(+), 1 deletion(-)
diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt
index 2469588c7ccb..1ecd75d2032a 100644
--- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt
+++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt
@@ -10,6 +10,11 @@ Required properties:
- clock-names: should be "pclk".
- spk-depop-time-ms: speak depop time msec.
+Optional properties:
+
+- mute-gpios: GPIO specifier for external line driver control (typically the
+ dedicated GPIO_MUTE pin)
+
Example for rk3328 internal codec:
codec: codec@ff410000 {
@@ -18,6 +23,6 @@ codec: codec@ff410000 {
rockchip,grf = <&grf>;
clocks = <&cru PCLK_ACODEC>;
clock-names = "pclk";
+ mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>;
spk-depop-time-ms = 100;
- status = "disabled";
};
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro" to the asoc tree
@ 2020-02-19 20:57 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 20:57 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 56f1003f65830697bd68ae1e26b5561e4cbe6523 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 19 Feb 2020 15:56:25 +0900
Subject: [PATCH] ASoC: mediatek: mt8183-da7219-max98357: use
for_each_rtd_codec_dai() macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87h7znjcag.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
index 03d104fbe185..4a5ef07e956b 100644
--- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
+++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
@@ -40,6 +40,7 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai;
unsigned int rate = params_rate(params);
unsigned int mclk_fs_ratio = 256;
unsigned int mclk_fs = rate * mclk_fs_ratio;
@@ -51,8 +52,7 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream,
if (ret < 0)
dev_err(rtd->dev, "failed to set cpu dai sysclk\n");
- for (j = 0; j < rtd->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+ for_each_rtd_codec_dai(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name, "da7219.5-001a")) {
ret = snd_soc_dai_set_sysclk(codec_dai,
@@ -82,10 +82,10 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream,
static int mt8183_da7219_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai;
int ret = 0, j;
- for (j = 0; j < rtd->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+ for_each_rtd_codec_dai(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name, "da7219.5-001a")) {
ret = snd_soc_dai_set_pll(codec_dai,
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro" to the asoc tree
@ 2020-02-19 20:57 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 20:57 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 89a2870f6be6aa75de2df22f8baa982c2d7d86e8 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 19 Feb 2020 15:56:20 +0900
Subject: [PATCH] ASoC: intel: kbl_da7219_max98927: use
for_each_rtd_codec_dai() macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87imk3jcal.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
index 7a13e9b35187..88f69e3697d2 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98927.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -176,10 +176,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *runtime = substream->private_data;
+ struct snd_soc_dai *codec_dai;
int ret, j;
- for (j = 0; j < runtime->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
+ for_each_rtd_codec_dai(runtime, j, codec_dai) {
if (!strcmp(codec_dai->component->name, MAX98927_DEV0_NAME)) {
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16);
@@ -221,10 +221,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai;
int j, ret;
- for (j = 0; j < rtd->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+ for_each_rtd_codec_dai(rtd, j, codec_dai) {
const char *name = codec_dai->component->name;
struct snd_soc_component *component = codec_dai->component;
struct snd_soc_dapm_context *dapm =
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro" to the asoc tree
@ 2020-02-19 20:57 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 20:57 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 225c53a8cfb6fdd8defbbf72e8dcfb3801f7f51e Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 19 Feb 2020 15:56:15 +0900
Subject: [PATCH] ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai()
macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87k14jjcaq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/cml_rt1011_rt5682.c | 3 +--
1 file changed, 1 insertion(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c
index dd80d0186a6c..02aa18d24319 100644
--- a/sound/soc/intel/boards/cml_rt1011_rt5682.c
+++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c
@@ -164,8 +164,7 @@ static int cml_rt1011_hw_params(struct snd_pcm_substream *substream,
srate = params_rate(params);
- for (i = 0; i < rtd->num_codecs; i++) {
- codec_dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
/* 100 Fs to drive 24 bit data */
ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK,
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro" to the asoc tree
@ 2020-02-19 20:57 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 20:57 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From cf4dae032096f1299cf390fd55da489cb445dcbb Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 19 Feb 2020 15:56:09 +0900
Subject: [PATCH] ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87lfozjcaw.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/apq8016_sbc.c | 7 +++----
1 file changed, 3 insertions(+), 4 deletions(-)
diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index ac75838bbfab..2d064f3bc9b6 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -34,8 +34,8 @@ struct apq8016_sbc_data {
static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai;
struct snd_soc_component *component;
- struct snd_soc_dai_link *dai_link = rtd->dai_link;
struct snd_soc_card *card = rtd->card;
struct apq8016_sbc_data *pdata = snd_soc_card_get_drvdata(card);
int i, rval;
@@ -90,10 +90,9 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd)
pdata->jack_setup = true;
}
- for (i = 0 ; i < dai_link->num_codecs; i++) {
- struct snd_soc_dai *dai = rtd->codec_dais[i];
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
- component = dai->component;
+ component = codec_dai->component;
/* Set default mclk for internal codec */
rval = snd_soc_component_set_sysclk(component, 0, 0, DEFAULT_MCLK_RATE,
SND_SOC_CLOCK_IN);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro" to the asoc tree
@ 2020-02-19 20:57 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 20:57 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From a4eb41eef331d31b8593defa10b249e155e0314f Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 19 Feb 2020 15:56:01 +0900
Subject: [PATCH] ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87mu9fjcb4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/sdm845.c | 20 +++++++++-----------
1 file changed, 9 insertions(+), 11 deletions(-)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 3b5547a27aad..5a23597261ac 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -43,14 +43,14 @@ static int sdm845_slim_snd_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai_link *dai_link = rtd->dai_link;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai;
u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
u32 rx_ch_cnt = 0, tx_ch_cnt = 0;
int ret = 0, i;
- for (i = 0 ; i < dai_link->num_codecs; i++) {
- ret = snd_soc_dai_get_channel_map(rtd->codec_dais[i],
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ ret = snd_soc_dai_get_channel_map(codec_dai,
&tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch);
if (ret != 0 && ret != -ENOTSUPP) {
@@ -77,6 +77,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai;
int ret = 0, j;
int channels, slot_width;
@@ -125,8 +126,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
}
}
- for (j = 0; j < rtd->num_codecs; j++) {
- struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
+ for_each_rtd_codec_dai(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name_prefix, "Left")) {
ret = snd_soc_dai_set_tdm_slot(
@@ -214,7 +214,6 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card);
struct snd_jack *jack;
- struct snd_soc_dai_link *dai_link = rtd->dai_link;
/*
* Codec SLIMBUS configuration
* RX1, RX2, RX3, RX4, RX5, RX6, RX7, RX8, RX9, RX10, RX11, RX12, RX13
@@ -266,8 +265,8 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
}
break;
case SLIMBUS_0_RX...SLIMBUS_6_TX:
- for (i = 0 ; i < dai_link->num_codecs; i++) {
- rval = snd_soc_dai_set_channel_map(rtd->codec_dais[i],
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ rval = snd_soc_dai_set_channel_map(codec_dai,
ARRAY_SIZE(tx_ch),
tx_ch,
ARRAY_SIZE(rx_ch),
@@ -275,7 +274,7 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
if (rval != 0 && rval != -ENOTSUPP)
return rval;
- snd_soc_dai_set_sysclk(rtd->codec_dais[i], 0,
+ snd_soc_dai_set_sysclk(codec_dai, 0,
WCD934X_DEFAULT_MCLK_RATE,
SNDRV_PCM_STREAM_PLAYBACK);
}
@@ -345,8 +344,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream)
codec_dai_fmt |= SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_DSP_B;
- for (j = 0; j < rtd->num_codecs; j++) {
- codec_dai = rtd->codec_dais[j];
+ for_each_rtd_codec_dai(rtd, j, codec_dai) {
if (!strcmp(codec_dai->component->name_prefix,
"Left")) {
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro" to the asoc tree
@ 2020-02-19 20:57 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-19 20:57 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From ce83baca8526689c332135c0abca2667ba056009 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 19 Feb 2020 15:55:53 +0900
Subject: [PATCH] ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87o8tvjcbc.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
drivers/soundwire/qcom.c | 7 ++++---
1 file changed, 4 insertions(+), 3 deletions(-)
diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c
index 1c6c6a2e0def..fb30bbec999a 100644
--- a/drivers/soundwire/qcom.c
+++ b/drivers/soundwire/qcom.c
@@ -594,6 +594,7 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream,
struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(dai->dev);
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct sdw_stream_runtime *sruntime;
+ struct snd_soc_dai *codec_dai;
int ret, i;
sruntime = sdw_alloc_stream(dai->name);
@@ -602,12 +603,12 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream,
ctrl->sruntime[dai->id] = sruntime;
- for (i = 0; i < rtd->num_codecs; i++) {
- ret = snd_soc_dai_set_sdw_stream(rtd->codec_dais[i], sruntime,
+ for_each_rtd_codec_dai(rtd, i, codec_dai) {
+ ret = snd_soc_dai_set_sdw_stream(codec_dai, sruntime,
substream->stream);
if (ret < 0 && ret != -ENOTSUPP) {
dev_err(dai->dev, "Failed to set sdw stream on %s",
- rtd->codec_dais[i]->name);
+ codec_dai->name);
sdw_release_stream(sruntime);
return ret;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "dt-bindings: sound: Add TLV320ADCx140 dt bindings" to the asoc tree
@ 2020-02-20 22:18 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-20 22:18 UTC (permalink / raw)
To: Dan Murphy; +Cc: alsa-devel, Mark Brown
The patch
dt-bindings: sound: Add TLV320ADCx140 dt bindings
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 4ee67cbd97668ab1b17d86d85348302c0b7490cd Mon Sep 17 00:00:00 2001
From: Dan Murphy <dmurphy@ti.com>
Date: Thu, 20 Feb 2020 15:07:58 -0600
Subject: [PATCH] dt-bindings: sound: Add TLV320ADCx140 dt bindings
Add dt bindings for the TLV320ADCx140 Burr-Brown ADC.
The initial support is for the following:
TLV320ADC3140 - http://www.ti.com/lit/gpn/tlv320adc3140
TLV320ADC5140 - http://www.ti.com/lit/gpn/tlv320adc5140
TLV320ADC6140 - http://www.ti.com/lit/gpn/tlv320adc6140
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200220210759.31466-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../bindings/sound/tlv320adcx140.yaml | 83 +++++++++++++++++++
1 file changed, 83 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml
diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml
new file mode 100644
index 000000000000..1433ff62b14f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml
@@ -0,0 +1,83 @@
+# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause)
+# Copyright (C) 2019 Texas Instruments Incorporated
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/tlv320adcx140.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Texas Instruments TLV320ADCX140 Quad Channel Analog-to-Digital Converter
+
+maintainers:
+ - Dan Murphy <dmurphy@ti.com>
+
+description: |
+ The TLV320ADCX140 are multichannel (4-ch analog recording or 8-ch digital
+ PDM microphones recording), high-performance audio, analog-to-digital
+ converter (ADC) with analog inputs supporting up to 2V RMS. The TLV320ADCX140
+ family supports line and microphone Inputs, and offers a programmable
+ microphone bias or supply voltage generation.
+
+ Specifications can be found at:
+ http://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf
+ http://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf
+ http://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf
+
+properties:
+ compatible:
+ oneOf:
+ - const: ti,tlv320adc3140
+ - const: ti,tlv320adc5140
+ - const: ti,tlv320adc6140
+
+ reg:
+ maxItems: 1
+ description: |
+ I2C addresss of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f
+
+ reset-gpios:
+ description: |
+ GPIO used for hardware reset.
+
+ areg-supply:
+ description: |
+ Regulator with AVDD at 3.3V. If not defined then the internal regulator
+ is enabled.
+
+ ti,mic-bias-source:
+ description: |
+ Indicates the source for MIC Bias.
+ 0 - Mic bias is set to VREF
+ 1 - Mic bias is set to VREF × 1.096
+ 6 - Mic bias is set to AVDD
+ allOf:
+ - $ref: /schemas/types.yaml#/definitions/uint32
+ - enum: [0, 1, 6]
+
+ ti,vref-source:
+ description: |
+ Indicates the source for MIC Bias.
+ 0 - Set VREF to 2.75V
+ 1 - Set VREF to 2.5V
+ 2 - Set VREF to 1.375V
+ allOf:
+ - $ref: /schemas/types.yaml#/definitions/uint32
+ - enum: [0, 1, 2]
+
+required:
+ - compatible
+ - reg
+
+examples:
+ - |
+ #include <dt-bindings/gpio/gpio.h>
+ i2c0 {
+ #address-cells = <1>;
+ #size-cells = <0>;
+ codec: codec@4c {
+ compatible = "ti,tlv320adc5140";
+ reg = <0x4c>;
+ ti,use-internal-areg;
+ ti,mic-bias-source = <6>;
+ reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>;
+ };
+ };
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: tas2562: Add entries for the TAS2563 audio amplifier" to the asoc tree
@ 2020-02-26 18:47 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-26 18:47 UTC (permalink / raw)
To: Dan Murphy; +Cc: alsa-devel, Mark Brown
The patch
ASoC: tas2562: Add entries for the TAS2563 audio amplifier
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 14f8c8d8fd62207f081549d45099a90dd3717696 Mon Sep 17 00:00:00 2001
From: Dan Murphy <dmurphy@ti.com>
Date: Wed, 26 Feb 2020 07:03:04 -0600
Subject: [PATCH] ASoC: tas2562: Add entries for the TAS2563 audio amplifier
The TAS2563 is register compatible with the TAS2562. The main
difference is the TAS2563 has a programmable DSP to manage different
audio profiles.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200226130305.12043-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/tas2562.c | 9 ++++++++-
1 file changed, 8 insertions(+), 1 deletion(-)
diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c
index d5e04030a0c1..79c3c3d79766 100644
--- a/sound/soc/codecs/tas2562.c
+++ b/sound/soc/codecs/tas2562.c
@@ -55,6 +55,11 @@ struct tas2562_data {
int volume_lvl;
};
+enum tas256x_model {
+ TAS2562,
+ TAS2563,
+};
+
static int tas2562_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
@@ -664,13 +669,15 @@ static int tas2562_probe(struct i2c_client *client,
}
static const struct i2c_device_id tas2562_id[] = {
- { "tas2562", 0 },
+ { "tas2562", TAS2562 },
+ { "tas2563", TAS2563 },
{ }
};
MODULE_DEVICE_TABLE(i2c, tas2562_id);
static const struct of_device_id tas2562_of_match[] = {
{ .compatible = "ti,tas2562", },
+ { .compatible = "ti,tas2563", },
{ },
};
MODULE_DEVICE_TABLE(of, tas2562_of_match);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding" to the asoc tree
@ 2020-02-26 18:47 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-26 18:47 UTC (permalink / raw)
To: Dan Murphy; +Cc: alsa-devel, Mark Brown
The patch
ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From f87cdb1f9937e6f5234e3300804ac156e639bc00 Mon Sep 17 00:00:00 2001
From: Dan Murphy <dmurphy@ti.com>
Date: Wed, 26 Feb 2020 07:03:03 -0600
Subject: [PATCH] ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562
binding
Add the Texas Instruments TAS2563 audio amplifier to the TAS262
binding.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
CC: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200226130305.12043-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/tas2562.txt | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt
index 658e1fb18a99..94796b547184 100644
--- a/Documentation/devicetree/bindings/sound/tas2562.txt
+++ b/Documentation/devicetree/bindings/sound/tas2562.txt
@@ -8,7 +8,7 @@ real time monitoring of loudspeaker behavior.
Required properties:
- #address-cells - Should be <1>.
- #size-cells - Should be <0>.
- - compatible: - Should contain "ti,tas2562".
+ - compatible: - Should contain "ti,tas2562", "ti,tas2563".
- reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f.
- ti,imon-slot-no:- TDM TX current sense time slot.
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: Add multiple CPU DAI support in DAPM" to the asoc tree
@ 2020-02-26 18:47 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-26 18:47 UTC (permalink / raw)
To: Shreyas NC
Cc: alsa-devel, Kuninori Morimoto, Pierre-Louis Bossart, Vinod Koul,
Mark Brown, Bard Liao
The patch
ASoC: Add multiple CPU DAI support in DAPM
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From de6214a33633d8ce1c1490336f8e798e75ccd004 Mon Sep 17 00:00:00 2001
From: Shreyas NC <shreyas.nc@intel.com>
Date: Tue, 25 Feb 2020 21:39:15 +0800
Subject: [PATCH] ASoC: Add multiple CPU DAI support in DAPM
DAPM handles DAIs during soc_dapm_stream_event() and during addition
and creation of DAI widgets i.e., dapm_add_valid_dai_widget() and
dapm_connect_dai_link_widgets().
Extend these functions to handle multiple cpu dai.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Shreyas NC <shreyas.nc@intel.com>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200225133917.21314-5-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-dapm.c | 20 ++++++++++++++++----
1 file changed, 16 insertions(+), 4 deletions(-)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 539a1eaebeac..6ce024d52170 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -4361,9 +4361,19 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
struct snd_soc_dai *codec_dai;
int i;
- for_each_rtd_codec_dai(rtd, i, codec_dai)
- dapm_add_valid_dai_widget(card, rtd,
- codec_dai, rtd->cpu_dais[0]);
+ if (rtd->num_cpus == 1) {
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ dapm_add_valid_dai_widget(card, rtd, codec_dai,
+ rtd->cpu_dais[0]);
+ } else if (rtd->num_codecs == rtd->num_cpus) {
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ dapm_add_valid_dai_widget(card, rtd, codec_dai,
+ rtd->cpu_dais[i]);
+ } else {
+ dev_err(card->dev,
+ "N cpus to M codecs link is not supported yet\n");
+ }
+
}
static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream,
@@ -4424,9 +4434,11 @@ static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
int event)
{
struct snd_soc_dai *codec_dai;
+ struct snd_soc_dai *cpu_dai;
int i;
- soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event);
+ for_each_rtd_cpu_dai(rtd, i, cpu_dai)
+ soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event);
for_each_rtd_codec_dai(rtd, i, codec_dai)
soc_dapm_dai_stream_event(codec_dai, stream, event);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: Add dapm_add_valid_dai_widget helper" to the asoc tree
@ 2020-02-26 18:47 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-26 18:47 UTC (permalink / raw)
To: Shreyas NC
Cc: alsa-devel, Kuninori Morimoto, Pierre-Louis Bossart, Vinod Koul,
Mark Brown, Bard Liao
The patch
ASoC: Add dapm_add_valid_dai_widget helper
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 6c4b13b51aa36aab023dd0bf24bf5582c9ba091e Mon Sep 17 00:00:00 2001
From: Shreyas NC <shreyas.nc@intel.com>
Date: Tue, 25 Feb 2020 21:39:14 +0800
Subject: [PATCH] ASoC: Add dapm_add_valid_dai_widget helper
Adding a helper to connect widget for a specific cpu and codec dai
The helper will help dapm_connect_dai_link_widgets() to reduce indents.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Shreyas NC <shreyas.nc@intel.com>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200225133917.21314-4-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-dapm.c | 119 +++++++++++++++++++++++--------------------
1 file changed, 63 insertions(+), 56 deletions(-)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 58c318c9debb..539a1eaebeac 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -4277,16 +4277,15 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
return 0;
}
-static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
- struct snd_soc_pcm_runtime *rtd)
+static void dapm_add_valid_dai_widget(struct snd_soc_card *card,
+ struct snd_soc_pcm_runtime *rtd,
+ struct snd_soc_dai *codec_dai,
+ struct snd_soc_dai *cpu_dai)
{
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai;
struct snd_soc_dapm_widget *playback = NULL, *capture = NULL;
struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu;
struct snd_pcm_substream *substream;
struct snd_pcm_str *streams = rtd->pcm->streams;
- int i;
if (rtd->dai_link->params) {
playback_cpu = cpu_dai->capture_widget;
@@ -4298,67 +4297,75 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
capture_cpu = capture;
}
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- /* connect BE DAI playback if widgets are valid */
- codec = codec_dai->playback_widget;
-
- if (playback_cpu && codec) {
- if (!playback) {
- substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
- playback = snd_soc_dapm_new_dai(card, substream,
- "playback");
- if (IS_ERR(playback)) {
- dev_err(rtd->dev,
- "ASoC: Failed to create DAI %s: %ld\n",
- codec_dai->name,
- PTR_ERR(playback));
- continue;
- }
-
- snd_soc_dapm_add_path(&card->dapm, playback_cpu,
- playback, NULL, NULL);
+ /* connect BE DAI playback if widgets are valid */
+ codec = codec_dai->playback_widget;
+
+ if (playback_cpu && codec) {
+ if (!playback) {
+ substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ playback = snd_soc_dapm_new_dai(card, substream,
+ "playback");
+ if (IS_ERR(playback)) {
+ dev_err(rtd->dev,
+ "ASoC: Failed to create DAI %s: %ld\n",
+ codec_dai->name,
+ PTR_ERR(playback));
+ goto capture;
}
- dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
- cpu_dai->component->name, playback_cpu->name,
- codec_dai->component->name, codec->name);
-
- snd_soc_dapm_add_path(&card->dapm, playback, codec,
- NULL, NULL);
+ snd_soc_dapm_add_path(&card->dapm, playback_cpu,
+ playback, NULL, NULL);
}
- }
- for_each_rtd_codec_dai(rtd, i, codec_dai) {
- /* connect BE DAI capture if widgets are valid */
- codec = codec_dai->capture_widget;
-
- if (codec && capture_cpu) {
- if (!capture) {
- substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream;
- capture = snd_soc_dapm_new_dai(card, substream,
- "capture");
- if (IS_ERR(capture)) {
- dev_err(rtd->dev,
- "ASoC: Failed to create DAI %s: %ld\n",
- codec_dai->name,
- PTR_ERR(capture));
- continue;
- }
-
- snd_soc_dapm_add_path(&card->dapm, capture,
- capture_cpu, NULL, NULL);
+ dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
+ cpu_dai->component->name, playback_cpu->name,
+ codec_dai->component->name, codec->name);
+
+ snd_soc_dapm_add_path(&card->dapm, playback, codec,
+ NULL, NULL);
+ }
+
+capture:
+ /* connect BE DAI capture if widgets are valid */
+ codec = codec_dai->capture_widget;
+
+ if (codec && capture_cpu) {
+ if (!capture) {
+ substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ capture = snd_soc_dapm_new_dai(card, substream,
+ "capture");
+ if (IS_ERR(capture)) {
+ dev_err(rtd->dev,
+ "ASoC: Failed to create DAI %s: %ld\n",
+ codec_dai->name,
+ PTR_ERR(capture));
+ return;
}
- dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
- codec_dai->component->name, codec->name,
- cpu_dai->component->name, capture_cpu->name);
-
- snd_soc_dapm_add_path(&card->dapm, codec, capture,
- NULL, NULL);
+ snd_soc_dapm_add_path(&card->dapm, capture,
+ capture_cpu, NULL, NULL);
}
+
+ dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
+ codec_dai->component->name, codec->name,
+ cpu_dai->component->name, capture_cpu->name);
+
+ snd_soc_dapm_add_path(&card->dapm, codec, capture,
+ NULL, NULL);
}
}
+static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
+ struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dai *codec_dai;
+ int i;
+
+ for_each_rtd_codec_dai(rtd, i, codec_dai)
+ dapm_add_valid_dai_widget(card, rtd,
+ codec_dai, rtd->cpu_dais[0]);
+}
+
static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream,
int event)
{
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: tas2562: Fix sample rate error message" to the asoc tree
@ 2020-02-26 18:48 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-02-26 18:48 UTC (permalink / raw)
To: Dan Murphy; +Cc: alsa-devel, Mark Brown
The patch
ASoC: tas2562: Fix sample rate error message
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 38b6a714941a285319734b9ccc30d34e107d5d7e Mon Sep 17 00:00:00 2001
From: Dan Murphy <dmurphy@ti.com>
Date: Wed, 26 Feb 2020 07:03:05 -0600
Subject: [PATCH] ASoC: tas2562: Fix sample rate error message
Fix error message for setting the sample rate. It says bitwidth but
should say sample rate.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200226130305.12043-3-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/tas2562.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c
index 5b4803a76719..be52886a5edb 100644
--- a/sound/soc/codecs/tas2562.c
+++ b/sound/soc/codecs/tas2562.c
@@ -252,7 +252,7 @@ static int tas2562_hw_params(struct snd_pcm_substream *substream,
ret = tas2562_set_samplerate(tas2562, params_rate(params));
if (ret)
- dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret);
+ dev_err(tas2562->dev, "set sample rate failed, %d\n", ret);
return ret;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver" to the asoc tree
@ 2020-03-02 14:36 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-02 14:36 UTC (permalink / raw)
To: YueHaibing; +Cc: alsa-devel, Mark Brown
The patch
ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From a79ee2e095c0a60c32d5b1fce39d58e0fc4d9ec5 Mon Sep 17 00:00:00 2001
From: YueHaibing <yuehaibing@huawei.com>
Date: Mon, 2 Mar 2020 15:05:22 +0800
Subject: [PATCH] ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver
snd_soc_dai_driver should set ops in rt1015_dai driver.
Also make the two variable static to fix sparse warnings.
Fixes: df31007400c3 ("ASoC: rt1015: add rt1015 amplifier driver")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20200302070522.48104-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/rt1015.c | 1 +
1 file changed, 1 insertion(+)
diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c
index c118d030bd2d..100b8c89d537 100644
--- a/sound/soc/codecs/rt1015.c
+++ b/sound/soc/codecs/rt1015.c
@@ -857,6 +857,7 @@ static struct snd_soc_dai_driver rt1015_dai[] = {
.rates = RT1015_STEREO_RATES,
.formats = RT1015_FORMATS,
},
+ .ops = &rt1015_aif_dai_ops,
}
};
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: amd: Fix compile warning of argument type" to the asoc tree
@ 2020-03-03 17:07 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-03 17:07 UTC (permalink / raw)
To: Akshu Agrawal; +Cc: alsa-devel, Mark Brown, kbuild test robot
The patch
ASoC: amd: Fix compile warning of argument type
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From d7729c40b376ab7c1ad4b0498a8bbf44bed3cbf4 Mon Sep 17 00:00:00 2001
From: Akshu Agrawal <akshu.agrawal@amd.com>
Date: Tue, 3 Mar 2020 14:34:37 +0530
Subject: [PATCH] ASoC: amd: Fix compile warning of argument type
Fixes:
>> sound/soc//amd/acp3x-rt5682-max9836.c:341:23: warning: format '%d'
>> expects argument of type 'int', but argument 3 has type 'long int'
>> [-Wformat=]
dev_err(&pdev->dev, "DMIC gpio failed err=%d\n",
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Link: https://lore.kernel.org/r/20200303090444.95805-1-akshu.agrawal@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/amd/acp3x-rt5682-max9836.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c
index 511b8b1722aa..521c9ab4c29c 100644
--- a/sound/soc/amd/acp3x-rt5682-max9836.c
+++ b/sound/soc/amd/acp3x-rt5682-max9836.c
@@ -338,7 +338,7 @@ static int acp3x_probe(struct platform_device *pdev)
dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW);
if (IS_ERR(dmic_sel)) {
- dev_err(&pdev->dev, "DMIC gpio failed err=%d\n",
+ dev_err(&pdev->dev, "DMIC gpio failed err=%ld\n",
PTR_ERR(dmic_sel));
return PTR_ERR(dmic_sel);
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: stm32: sai: manage rebind issue" to the asoc tree
@ 2020-03-04 13:39 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-04 13:39 UTC (permalink / raw)
To: Olivier Moysan; +Cc: alsa-devel, Mark Brown
The patch
ASoC: stm32: sai: manage rebind issue
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 0d6defc7e0e437a9fd53622f7fd85740f38d5693 Mon Sep 17 00:00:00 2001
From: Olivier Moysan <olivier.moysan@st.com>
Date: Wed, 4 Mar 2020 11:24:06 +0100
Subject: [PATCH] ASoC: stm32: sai: manage rebind issue
The commit e894efef9ac7 ("ASoC: core: add support to card rebind")
allows to rebind the sound card after a rebind of one of its component.
With this commit, the sound card is actually rebound,
but may be no more functional. The following problems have been seen
with STM32 SAI driver.
1) DMA channel is not requested:
With the sound card rebind the simplified call sequence is:
stm32_sai_sub_probe
snd_soc_register_component
snd_soc_try_rebind_card
snd_soc_instantiate_card
devm_snd_dmaengine_pcm_register
The problem occurs because the pcm must be registered,
before snd_soc_instantiate_card() is called.
Modify SAI driver, to change the call sequence as follows:
stm32_sai_sub_probe
devm_snd_dmaengine_pcm_register
snd_soc_register_component
snd_soc_try_rebind_card
2) DMA channel is not released:
dma_release_channel() is not called when
devm_dmaengine_pcm_release() is executed.
This occurs because SND_DMAENGINE_PCM_DRV_NAME component,
has already been released through devm_component_release().
devm_dmaengine_pcm_release() should be called before
devm_component_release() to avoid this problem.
Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component()
explicitly from SAI driver, to have the right sequence.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Message-Id: <20200304102406.8093-1-olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/stm/stm32_sai_sub.c | 18 ++++++++++--------
1 file changed, 10 insertions(+), 8 deletions(-)
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index 30bcd5d3a32a..10eb4b8e8e7e 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -1543,20 +1543,20 @@ static int stm32_sai_sub_probe(struct platform_device *pdev)
return ret;
}
- ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component,
- &sai->cpu_dai_drv, 1);
+ ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0);
+ if (ret) {
+ dev_err(&pdev->dev, "Could not register pcm dma\n");
+ return ret;
+ }
+
+ ret = snd_soc_register_component(&pdev->dev, &stm32_component,
+ &sai->cpu_dai_drv, 1);
if (ret)
return ret;
if (STM_SAI_PROTOCOL_IS_SPDIF(sai))
conf = &stm32_sai_pcm_config_spdif;
- ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0);
- if (ret) {
- dev_err(&pdev->dev, "Could not register pcm dma\n");
- return ret;
- }
-
return 0;
}
@@ -1565,6 +1565,8 @@ static int stm32_sai_sub_remove(struct platform_device *pdev)
struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev);
clk_unprepare(sai->pdata->pclk);
+ snd_dmaengine_pcm_unregister(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths()" to the asoc tree
@ 2020-03-05 13:15 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-05 13:15 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 6e02feb0d2663c1b7caa5e271c2a60e219f0ca07 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 27 Feb 2020 10:54:48 +0900
Subject: [PATCH] ASoC: soc-pcm: remove duplicate be check from
dpcm_add_paths()
dpcm_add_paths() checks returned be from dpcm_get_be()
static int dpcm_add_paths(...)
{
...
for_each_dapm_widgets(list, i, widget) {
...
be = dpcm_get_be(...);
...
/* make sure BE is a real BE */
=> if (!be->dai_link->no_pcm)
continue;
...
}
...
}
But, dpcm_get_be() itself is checking it already.
dpcm_get_be(...)
{
...
for_each_card_rtds(card, be) {
=> if (!be->dai_link->no_pcm)
continue;
...
if (...)
=> return be;
}
return NULL
}
This patch removes duplicate check
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87lfoo7q1j.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 4 ----
1 file changed, 4 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 8c27eb4d5e4c..e3a2c4f7757b 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1690,10 +1690,6 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream,
continue;
}
- /* make sure BE is a real BE */
- if (!be->dai_link->no_pcm)
- continue;
-
/* don't connect if FE is not running */
if (!fe->dpcm[stream].runtime && !fe->fe_compr)
continue;
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: use defined stream" to the asoc tree
@ 2020-03-05 13:15 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-05 13:15 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: use defined stream
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 25c2f5156dd57f03aee2de079248c23a56222c92 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Thu, 27 Feb 2020 10:54:38 +0900
Subject: [PATCH] ASoC: soc-pcm: use defined stream
Many functions defines "stream = substream->stream", but
some of them is using "substream->stream" instead of "stream".
It is pointless. This patch uses defined stream.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87mu947q1t.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 15 +++++++--------
1 file changed, 7 insertions(+), 8 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 90857138c823..8c27eb4d5e4c 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -644,8 +644,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
* bailed out on a higher level, since there would be no
* CODEC to support the transfer direction in that case.
*/
- if (!snd_soc_dai_stream_valid(codec_dai,
- substream->stream))
+ if (!snd_soc_dai_stream_valid(codec_dai, stream))
continue;
codec_stream = snd_soc_dai_get_pcm_stream(codec_dai, stream);
@@ -2149,7 +2148,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream)
dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
- ret = dpcm_be_dai_startup(fe, fe_substream->stream);
+ ret = dpcm_be_dai_startup(fe, stream);
if (ret < 0) {
dev_err(fe->dev,"ASoC: failed to start some BEs %d\n", ret);
goto be_err;
@@ -2180,7 +2179,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream)
return 0;
unwind:
- dpcm_be_dai_startup_unwind(fe, fe_substream->stream);
+ dpcm_be_dai_startup_unwind(fe, stream);
be_err:
dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
return ret;
@@ -2234,7 +2233,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream)
dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
/* shutdown the BEs */
- dpcm_be_dai_shutdown(fe, substream->stream);
+ dpcm_be_dai_shutdown(fe, stream);
dev_dbg(fe->dev, "ASoC: close FE %s\n", fe->dai_link->name);
@@ -2412,9 +2411,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream,
mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
- memcpy(&fe->dpcm[substream->stream].hw_params, params,
+ memcpy(&fe->dpcm[stream].hw_params, params,
sizeof(struct snd_pcm_hw_params));
- ret = dpcm_be_dai_hw_params(fe, substream->stream);
+ ret = dpcm_be_dai_hw_params(fe, stream);
if (ret < 0) {
dev_err(fe->dev,"ASoC: hw_params BE failed %d\n", ret);
goto out;
@@ -2736,7 +2735,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream)
goto out;
}
- ret = dpcm_be_dai_prepare(fe, substream->stream);
+ ret = dpcm_be_dai_prepare(fe, stream);
if (ret < 0)
goto out;
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: tidyup dulicate handing at dpcm_fe_dai_startup()" to the asoc tree
@ 2020-03-06 15:03 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-06 15:03 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: tidyup dulicate handing at dpcm_fe_dai_startup()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 8a01fbf0ac115268293d8764850edc0628a58e4f Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 6 Mar 2020 10:09:59 +0900
Subject: [PATCH] ASoC: soc-pcm: tidyup dulicate handing at
dpcm_fe_dai_startup()
error handling at dpcm_fe_dai_startup() has duplicate code.
This patch tidyup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87lfoep9ug.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 10 +++-------
1 file changed, 3 insertions(+), 7 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 3a30776858bf..7d787e0966f3 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2162,17 +2162,13 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream)
snd_pcm_limit_hw_rates(runtime);
ret = dpcm_apply_symmetry(fe_substream, stream);
- if (ret < 0) {
+ if (ret < 0)
dev_err(fe->dev, "ASoC: failed to apply dpcm symmetry %d\n",
ret);
- goto unwind;
- }
-
- dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
- return 0;
unwind:
- dpcm_be_dai_startup_unwind(fe, stream);
+ if (ret < 0)
+ dpcm_be_dai_startup_unwind(fe, stream);
be_err:
dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
return ret;
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: use snd_soc_dai_get_pcm_stream() at dpcm_set_fe_runtime()" to the asoc tree
@ 2020-03-06 15:03 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-06 15:03 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: use snd_soc_dai_get_pcm_stream() at dpcm_set_fe_runtime()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 0c9ba720f0be457443ba89b09a5198616cd3e811 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 6 Mar 2020 10:09:54 +0900
Subject: [PATCH] ASoC: soc-pcm: use snd_soc_dai_get_pcm_stream() at
dpcm_set_fe_runtime()
We already have snd_soc_dai_get_pcm_stream(),
let's use it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87mu8up9ul.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 9 +++------
1 file changed, 3 insertions(+), 6 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b405fb3a181b..3a30776858bf 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2022,7 +2022,6 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai;
- struct snd_soc_dai_driver *cpu_dai_drv;
int i;
for_each_rtd_cpu_dai(rtd, i, cpu_dai) {
@@ -2033,11 +2032,9 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream)
if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream))
continue;
- cpu_dai_drv = cpu_dai->driver;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback);
- else
- dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture);
+ dpcm_init_runtime_hw(runtime,
+ snd_soc_dai_get_pcm_stream(cpu_dai,
+ substream->stream));
}
dpcm_runtime_merge_format(substream, &runtime->hw.formats);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: add dpcm_fe_dai_cleanup()" to the asoc tree
@ 2020-03-06 15:03 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-06 15:03 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: add dpcm_fe_dai_cleanup()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 265694b67c13f00384bd0b97549b4681cbcc85af Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 6 Mar 2020 10:09:49 +0900
Subject: [PATCH] ASoC: soc-pcm: add dpcm_fe_dai_cleanup()
dpcm_fe_dai_close() and error case of dpcm_fe_dai_open()
need to do same cleanup operation.
To avoid duplicate code, this patch adds dpcm_fe_dai_cleanup()
and use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87o8tap9uq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 30 ++++++++++++++++--------------
1 file changed, 16 insertions(+), 14 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 3686dda097e2..b405fb3a181b 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2978,14 +2978,11 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
return ret;
}
-static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
+static void dpcm_fe_dai_cleanup(struct snd_pcm_substream *fe_substream)
{
struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
struct snd_soc_dpcm *dpcm;
- int stream = fe_substream->stream, ret;
-
- mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
- ret = dpcm_fe_dai_shutdown(fe_substream);
+ int stream = fe_substream->stream;
/* mark FE's links ready to prune */
for_each_dpcm_be(fe, stream, dpcm)
@@ -2994,6 +2991,18 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
dpcm_be_disconnect(fe, stream);
fe->dpcm[stream].runtime = NULL;
+}
+
+static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
+{
+ struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
+ int ret;
+
+ mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+ ret = dpcm_fe_dai_shutdown(fe_substream);
+
+ dpcm_fe_dai_cleanup(fe_substream);
+
mutex_unlock(&fe->card->mutex);
return ret;
}
@@ -3001,7 +3010,6 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
{
struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
- struct snd_soc_dpcm *dpcm;
struct snd_soc_dapm_widget_list *list;
int ret;
int stream = fe_substream->stream;
@@ -3021,14 +3029,8 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
dpcm_process_paths(fe, stream, &list, 1);
ret = dpcm_fe_dai_startup(fe_substream);
- if (ret < 0) {
- /* clean up all links */
- for_each_dpcm_be(fe, stream, dpcm)
- dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
-
- dpcm_be_disconnect(fe, stream);
- fe->dpcm[stream].runtime = NULL;
- }
+ if (ret < 0)
+ dpcm_fe_dai_cleanup(fe_substream);
dpcm_clear_pending_state(fe, stream);
dpcm_path_put(&list);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-pcm: move dpcm_fe_dai_close()" to the asoc tree
@ 2020-03-06 15:03 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-06 15:03 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-pcm: move dpcm_fe_dai_close()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 30fca26f8e2277ccd14fe3277a330b4f21cadca7 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Fri, 6 Mar 2020 10:09:44 +0900
Subject: [PATCH] ASoC: soc-pcm: move dpcm_fe_dai_close()
move dpcm_fe_dai_close() next to dpcm_fe_dai_open().
This is prepare for dpcm_fe_dai_open() cleanup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87pndqp9uv.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-pcm.c | 40 ++++++++++++++++++++--------------------
1 file changed, 20 insertions(+), 20 deletions(-)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e3a2c4f7757b..3686dda097e2 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2978,6 +2978,26 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
return ret;
}
+static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
+{
+ struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
+ struct snd_soc_dpcm *dpcm;
+ int stream = fe_substream->stream, ret;
+
+ mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
+ ret = dpcm_fe_dai_shutdown(fe_substream);
+
+ /* mark FE's links ready to prune */
+ for_each_dpcm_be(fe, stream, dpcm)
+ dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
+
+ dpcm_be_disconnect(fe, stream);
+
+ fe->dpcm[stream].runtime = NULL;
+ mutex_unlock(&fe->card->mutex);
+ return ret;
+}
+
static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
{
struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
@@ -3017,26 +3037,6 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
return ret;
}
-static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
-{
- struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
- struct snd_soc_dpcm *dpcm;
- int stream = fe_substream->stream, ret;
-
- mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
- ret = dpcm_fe_dai_shutdown(fe_substream);
-
- /* mark FE's links ready to prune */
- for_each_dpcm_be(fe, stream, dpcm)
- dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE;
-
- dpcm_be_disconnect(fe, stream);
-
- fe->dpcm[stream].runtime = NULL;
- mutex_unlock(&fe->card->mutex);
- return ret;
-}
-
/* create a new pcm */
int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
{
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: pcm: Export parameter intersection logic" to the asoc tree
@ 2020-03-06 15:03 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-06 15:03 UTC (permalink / raw)
To: Samuel Holland; +Cc: alsa-devel, Mark Brown
The patch
ASoC: pcm: Export parameter intersection logic
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 5854a46486ad5b8d73766735fb0d77f05956b22c Mon Sep 17 00:00:00 2001
From: Samuel Holland <samuel@sholland.org>
Date: Wed, 4 Mar 2020 23:11:42 -0600
Subject: [PATCH] ASoC: pcm: Export parameter intersection logic
The logic to calculate the subset of stream parameters supported by all
DAIs associated with a PCM stream is nontrivial. Export a helper
function so it can be used to set up simple codec2codec DAI links.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Link: https://lore.kernel.org/r/20200305051143.60691-3-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc.h | 3 +++
sound/soc/soc-pcm.c | 56 ++++++++++++++++++++++++++++++---------------
2 files changed, 41 insertions(+), 18 deletions(-)
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 81e5d17be935..9543d9246ca4 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -471,6 +471,9 @@ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd);
void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream);
void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream);
+int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hardware *hw, int stream);
+
int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd,
unsigned int dai_fmt);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e3a2c4f7757b..de4226357e2b 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -587,11 +587,18 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream)
soc_pcm_set_msb(substream, cpu_bits);
}
-static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
+/**
+ * snd_soc_runtime_calc_hw() - Calculate hw limits for a PCM stream
+ * @rtd: ASoC PCM runtime
+ * @hw: PCM hardware parameters (output)
+ * @stream: Direction of the PCM stream
+ *
+ * Calculates the subset of stream parameters supported by all DAIs
+ * associated with the PCM stream.
+ */
+int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hardware *hw, int stream)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_pcm_hardware *hw = &runtime->hw;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai;
struct snd_soc_dai *cpu_dai;
struct snd_soc_pcm_stream *codec_stream;
@@ -602,7 +609,6 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
unsigned int cpu_rate_min = 0, cpu_rate_max = UINT_MAX;
unsigned int rates = UINT_MAX, cpu_rates = UINT_MAX;
u64 formats = ULLONG_MAX;
- int stream = substream->stream;
int i;
/* first calculate min/max only for CPUs in the DAI link */
@@ -613,12 +619,8 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
* Otherwise, since the rate, channel, and format values will
* zero in that case, we would have no usable settings left,
* causing the resulting setup to fail.
- * At least one CPU should match, otherwise we should have
- * bailed out on a higher level, since there would be no
- * CPU to support the transfer direction in that case.
*/
- if (!snd_soc_dai_stream_valid(cpu_dai,
- substream->stream))
+ if (!snd_soc_dai_stream_valid(cpu_dai, stream))
continue;
cpu_stream = snd_soc_dai_get_pcm_stream(cpu_dai, stream);
@@ -640,9 +642,6 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
* Otherwise, since the rate, channel, and format values will
* zero in that case, we would have no usable settings left,
* causing the resulting setup to fail.
- * At least one CODEC should match, otherwise we should have
- * bailed out on a higher level, since there would be no
- * CODEC to support the transfer direction in that case.
*/
if (!snd_soc_dai_stream_valid(codec_dai, stream))
continue;
@@ -657,6 +656,10 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates);
}
+ /* Verify both a valid CPU DAI and a valid CODEC DAI were found */
+ if (!chan_min || !cpu_chan_min)
+ return -EINVAL;
+
/*
* chan min/max cannot be enforced if there are multiple CODEC DAIs
* connected to CPU DAI(s), use CPU DAI's directly and let
@@ -670,18 +673,35 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
/* finally find a intersection between CODECs and CPUs */
hw->channels_min = max(chan_min, cpu_chan_min);
hw->channels_max = min(chan_max, cpu_chan_max);
- if (hw->formats)
- hw->formats &= formats;
- else
- hw->formats = formats;
+ hw->formats = formats;
hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_rates);
- snd_pcm_limit_hw_rates(runtime);
+ snd_pcm_hw_limit_rates(hw);
hw->rate_min = max(hw->rate_min, cpu_rate_min);
hw->rate_min = max(hw->rate_min, rate_min);
hw->rate_max = min_not_zero(hw->rate_max, cpu_rate_max);
hw->rate_max = min_not_zero(hw->rate_max, rate_max);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_runtime_calc_hw);
+
+static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_hardware *hw = &substream->runtime->hw;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ u64 formats = hw->formats;
+
+ /*
+ * At least one CPU and one CODEC should match. Otherwise, we should
+ * have bailed out on a higher level, since there would be no CPU or
+ * CODEC to support the transfer direction in that case.
+ */
+ snd_soc_runtime_calc_hw(rtd, hw, substream->stream);
+
+ if (formats)
+ hw->formats &= formats;
}
static int soc_pcm_components_open(struct snd_pcm_substream *substream)
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates" to the asoc tree
@ 2020-03-06 15:03 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-06 15:03 UTC (permalink / raw)
To: Samuel Holland; +Cc: alsa-devel, Mark Brown
The patch
ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 4769bfb9dada678b31a2ec275372624dbfeed9d1 Mon Sep 17 00:00:00 2001
From: Samuel Holland <samuel@sholland.org>
Date: Wed, 4 Mar 2020 23:11:41 -0600
Subject: [PATCH] ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates
It can be useful to derive min/max rates of a snd_pcm_hardware without
having a snd_pcm_runtime, such as before constructing an ASoC DAI link.
Create a new helper that takes a pointer to a snd_pcm_hardware directly,
and refactor the original function as a wrapper around it, to avoid
needing to update any call sites.
Signed-off-by: Samuel Holland <samuel@sholland.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200305051143.60691-2-samuel@sholland.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/pcm.h | 9 ++++++++-
sound/core/pcm_misc.c | 18 +++++++++---------
2 files changed, 17 insertions(+), 10 deletions(-)
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 2628246b76fa..f7a95b711100 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -1127,7 +1127,14 @@ snd_pcm_kernel_readv(struct snd_pcm_substream *substream,
return __snd_pcm_lib_xfer(substream, bufs, false, frames, true);
}
-int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime);
+int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw);
+
+static inline int
+snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
+{
+ return snd_pcm_hw_limit_rates(&runtime->hw);
+}
+
unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate);
unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit);
unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a,
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index a6a541511534..5dd2e5335900 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -474,32 +474,32 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int
EXPORT_SYMBOL(snd_pcm_format_set_silence);
/**
- * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields
- * @runtime: the runtime instance
+ * snd_pcm_hw_limit_rates - determine rate_min/rate_max fields
+ * @hw: the pcm hw instance
*
* Determines the rate_min and rate_max fields from the rates bits of
- * the given runtime->hw.
+ * the given hw.
*
* Return: Zero if successful.
*/
-int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
+int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw)
{
int i;
for (i = 0; i < (int)snd_pcm_known_rates.count; i++) {
- if (runtime->hw.rates & (1 << i)) {
- runtime->hw.rate_min = snd_pcm_known_rates.list[i];
+ if (hw->rates & (1 << i)) {
+ hw->rate_min = snd_pcm_known_rates.list[i];
break;
}
}
for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) {
- if (runtime->hw.rates & (1 << i)) {
- runtime->hw.rate_max = snd_pcm_known_rates.list[i];
+ if (hw->rates & (1 << i)) {
+ hw->rate_max = snd_pcm_known_rates.list[i];
break;
}
}
return 0;
}
-EXPORT_SYMBOL(snd_pcm_limit_hw_rates);
+EXPORT_SYMBOL(snd_pcm_hw_limit_rates);
/**
* snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: jz4740-i2s: Fix divider written at incorrect offset in register" to the asoc tree
@ 2020-03-09 14:03 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-09 14:03 UTC (permalink / raw)
To: Paul Cercueil; +Cc: alsa-devel, Mark Brown, stable
The patch
ASoC: jz4740-i2s: Fix divider written at incorrect offset in register
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 9401d5aa328e64617d87abd59af1c91cace4c3e4 Mon Sep 17 00:00:00 2001
From: Paul Cercueil <paul@crapouillou.net>
Date: Fri, 6 Mar 2020 23:29:27 +0100
Subject: [PATCH] ASoC: jz4740-i2s: Fix divider written at incorrect offset in
register
The 4-bit divider value was written at offset 8, while the jz4740
programming manual locates it at offset 0.
Fixes: 26b0aad80a86 ("ASoC: jz4740: Add dynamic sampling rate support to jz4740-i2s")
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200306222931.39664-2-paul@crapouillou.net
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/jz4740/jz4740-i2s.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index 9d5405881209..434737b2b2b2 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -83,7 +83,7 @@
#define JZ_AIC_I2S_STATUS_BUSY BIT(2)
#define JZ_AIC_CLK_DIV_MASK 0xf
-#define I2SDIV_DV_SHIFT 8
+#define I2SDIV_DV_SHIFT 0
#define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT)
#define I2SDIV_IDV_SHIFT 8
#define I2SDIV_IDV_MASK (0xf << I2SDIV_IDV_SHIFT)
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: zte: zx-tdm: remove redundant variables dev" to the asoc tree
@ 2020-03-11 18:00 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-11 18:00 UTC (permalink / raw)
To: tangbin; +Cc: alsa-devel, Mark Brown
The patch
ASoC: zte: zx-tdm: remove redundant variables dev
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 103ae95513803102d2a2c91458cfac5dfbaad124 Mon Sep 17 00:00:00 2001
From: tangbin <tangbin@cmss.chinamobile.com>
Date: Wed, 11 Mar 2020 22:46:46 +0800
Subject: [PATCH] ASoC: zte: zx-tdm: remove redundant variables dev
In this function, the variable 'dev' is assigned to '&pdev->dev',
but in the following code, all the assignments to 'struce device'
are used '&pdev->dev' instead of 'dev',except 'zx_tdm->dev'.
So,the variable 'dev' in this function is redundant and can be
replaced by '&pdev->dev' as elsewhere.
Signed-off-by: tangbin <tangbin@cmss.chinamobile.com>
Link: https://lore.kernel.org/r/20200311144646.11292-1-tangbin@cmss.chinamobile.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/zte/zx-tdm.c | 3 +--
1 file changed, 1 insertion(+), 2 deletions(-)
diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c
index 0e5a05b25a77..4f787185d630 100644
--- a/sound/soc/zte/zx-tdm.c
+++ b/sound/soc/zte/zx-tdm.c
@@ -371,7 +371,6 @@ static struct snd_soc_dai_driver zx_tdm_dai = {
static int zx_tdm_probe(struct platform_device *pdev)
{
- struct device *dev = &pdev->dev;
struct of_phandle_args out_args;
unsigned int dma_reg_offset;
struct zx_tdm_info *zx_tdm;
@@ -384,7 +383,7 @@ static int zx_tdm_probe(struct platform_device *pdev)
if (!zx_tdm)
return -ENOMEM;
- zx_tdm->dev = dev;
+ zx_tdm->dev = &pdev->dev;
zx_tdm->dai_wclk = devm_clk_get(&pdev->dev, "wclk");
if (IS_ERR(zx_tdm->dai_wclk)) {
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: Intel: Fix stream cleanup on hw free" to the asoc tree
@ 2020-03-12 13:12 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-12 13:12 UTC (permalink / raw)
To: Cezary Rojewski; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: SOF: Intel: Fix stream cleanup on hw free
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 16dcefc23eefda57ee705dcffba67f1bd42c01bc Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Thu, 12 Mar 2020 13:00:58 +0100
Subject: [PATCH] ASoC: SOF: Intel: Fix stream cleanup on hw free
Field "substream" gets assigned during stream setup in
hda_dsp_pcm_hw_params() but it is never cleared afterwards during
cleanup procedure. Now, any non-pcm operation e.g.: compress can
mistakenly make use of that pointer as it's bypassing all
"if (s->substream)" checks.
Nulling the pointer during hw_free operation ensures no wild pointers
are left behind.
Fixes: cdae3b9a47aa ("ASoC: SOF: Intel: Add Intel specific HDA PCM operations")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200312120058.15057-1-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-stream.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index c0ab9bb2a797..d2234f802788 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -547,6 +547,8 @@ int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev,
SOF_HDA_REG_PP_PPCTL, mask, 0);
spin_unlock_irq(&bus->reg_lock);
+ stream->substream = NULL;
+
return 0;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp" to the asoc tree
@ 2020-03-13 16:17 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-13 16:17 UTC (permalink / raw)
To: Yong Zhi; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 3f32e596b03ef50fd2c49f2e13d42e1931525f35 Mon Sep 17 00:00:00 2001
From: Yong Zhi <yong.zhi@intel.com>
Date: Thu, 12 Mar 2020 14:48:57 -0500
Subject: [PATCH] ASoC: Intel: sof_da7219_max98373: Add support for max98360a
speaker amp
Add Maxim MAX98360A plug-and-play Class-D amplifier support on SSP1,
new card ID is sofda7219max98360a, name sof-da7219max98360a.
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200312194859.4051-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 4 +-
sound/soc/intel/boards/sof_da7219_max98373.c | 74 +++++++++++++++++--
.../intel/common/soc-acpi-intel-jsl-match.c | 19 ++++-
3 files changed, 87 insertions(+), 10 deletions(-)
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 6833ef548710..ab4ce652cc1a 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -507,7 +507,7 @@ endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK
if SND_SOC_SOF_JASPERLAKE
config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH
- tristate "SOF with DA7219 and MAX98373 in I2S Mode"
+ tristate "SOF with DA7219 and MAX98373/MAX98360A in I2S Mode"
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
depends on SND_HDA_CODEC_HDMI
@@ -516,7 +516,7 @@ config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH
select SND_SOC_DMIC
help
This adds support for ASoC machine driver for SOF platforms
- with DA7219 + MAX98373 I2S audio codec.
+ with DA7219 + MAX98373/MAX98360A I2S audio codec.
Say Y if you have such a device.
If unsure select "N".
diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c
index 8c657da5fcf0..7847dd44f41b 100644
--- a/sound/soc/intel/boards/sof_da7219_max98373.c
+++ b/sound/soc/intel/boards/sof_da7219_max98373.c
@@ -2,7 +2,7 @@
// Copyright(c) 2019 Intel Corporation.
/*
- * Intel SOF Machine driver for DA7219 + MAX98373 codec
+ * Intel SOF Machine driver for DA7219 + MAX98373/MAX98360A codec
*/
#include <linux/input.h>
@@ -69,11 +69,14 @@ static const struct snd_kcontrol_new controls[] = {
SOC_DAPM_PIN_SWITCH("Right Spk"),
};
+static const struct snd_kcontrol_new m98360a_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+};
+
static const struct snd_soc_dapm_widget widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_SPK("Left Spk", NULL),
- SND_SOC_DAPM_SPK("Right Spk", NULL),
SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
platform_clock_control, SND_SOC_DAPM_POST_PMD |
SND_SOC_DAPM_PRE_PMU),
@@ -83,15 +86,23 @@ static const struct snd_soc_dapm_route audio_map[] = {
{ "Headphone Jack", NULL, "HPL" },
{ "Headphone Jack", NULL, "HPR" },
- { "Left Spk", NULL, "Left BE_OUT" },
- { "Right Spk", NULL, "Right BE_OUT" },
-
{ "MIC", NULL, "Headset Mic" },
{ "Headphone Jack", NULL, "Platform Clock" },
{ "Headset Mic", NULL, "Platform Clock" },
};
+/* For MAX98373 amp */
+static const struct snd_soc_dapm_widget max98373_widgets[] = {
+ SND_SOC_DAPM_SPK("Left Spk", NULL),
+ SND_SOC_DAPM_SPK("Right Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route max98373_map[] = {
+ { "Left Spk", NULL, "Left BE_OUT" },
+ { "Right Spk", NULL, "Right BE_OUT" },
+};
+
static struct snd_soc_jack headset;
static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
@@ -133,6 +144,21 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
return ret;
}
+static int speaker_amp_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int ret;
+
+ /* Add widgets */
+ ret = snd_soc_dapm_new_controls(&rtd->card->dapm, max98373_widgets,
+ ARRAY_SIZE(max98373_widgets));
+ if (ret)
+ return ret;
+
+ /* Add routes */
+ return snd_soc_dapm_add_routes(&rtd->card->dapm, max98373_map,
+ ARRAY_SIZE(max98373_map));
+}
+
static int ssp1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -223,6 +249,8 @@ SND_SOC_DAILINK_DEF(ssp1_amps,
DAILINK_COMP_ARRAY(
/* Left */ COMP_CODEC(MAXIM_DEV0_NAME, MAX98373_CODEC_DAI),
/* Right */ COMP_CODEC(MAXIM_DEV1_NAME, MAX98373_CODEC_DAI)));
+/* For the driver-less spk amp */
+SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY()));
SND_SOC_DAILINK_DEF(dmic_pin,
DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin")));
@@ -254,6 +282,7 @@ static struct snd_soc_dai_link dais[] = {
.id = 0,
.ignore_pmdown_time = 1,
.no_pcm = 1,
+ .init = speaker_amp_init,
.dpcm_playback = 1,
.dpcm_capture = 1, /* IV feedback */
.ops = &ssp1_ops,
@@ -320,6 +349,21 @@ static struct snd_soc_card card_da7219_m98373 = {
.late_probe = card_late_probe,
};
+static struct snd_soc_card card_da7219_m98360a = {
+ .name = "da7219max98360a",
+ .owner = THIS_MODULE,
+ .dai_link = dais,
+ .num_links = ARRAY_SIZE(dais),
+ .controls = m98360a_controls,
+ .num_controls = ARRAY_SIZE(m98360a_controls),
+ .dapm_widgets = widgets,
+ .num_dapm_widgets = ARRAY_SIZE(widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+ .fully_routed = true,
+ .late_probe = card_late_probe,
+};
+
static int audio_probe(struct platform_device *pdev)
{
static struct snd_soc_card *card;
@@ -331,6 +375,17 @@ static int audio_probe(struct platform_device *pdev)
if (!ctx)
return -ENOMEM;
+ /* By default dais[0] is configured for max98373 */
+ if (!strcmp(pdev->name, "sof_da7219_max98360a")) {
+ dais[0] = (struct snd_soc_dai_link) {
+ .name = "SSP1-Codec",
+ .id = 0,
+ .no_pcm = 1,
+ .dpcm_playback = 1,
+ .ignore_pmdown_time = 1,
+ SND_SOC_DAILINK_REG(ssp1_pin, dummy, platform) };
+ }
+
INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
card = (struct snd_soc_card *)pdev->id_entry->driver_data;
card->dev = &pdev->dev;
@@ -351,13 +406,17 @@ static const struct platform_device_id board_ids[] = {
.name = "sof_da7219_max98373",
.driver_data = (kernel_ulong_t)&card_da7219_m98373,
},
+ {
+ .name = "sof_da7219_max98360a",
+ .driver_data = (kernel_ulong_t)&card_da7219_m98360a,
+ },
{ }
};
static struct platform_driver audio = {
.probe = audio_probe,
.driver = {
- .name = "sof_da7219_max98373",
+ .name = "sof_da7219_max98_360a_373",
.pm = &snd_soc_pm_ops,
},
.id_table = board_ids,
@@ -368,4 +427,5 @@ module_platform_driver(audio)
MODULE_DESCRIPTION("ASoC Intel(R) SOF Machine driver");
MODULE_AUTHOR("Yong Zhi <yong.zhi@intel.com>");
MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:sof_da7219_max98360a");
MODULE_ALIAS("platform:sof_da7219_max98373");
diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c
index ed2b125f6a11..70f01495a166 100644
--- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c
@@ -9,13 +9,30 @@
#include <sound/soc-acpi.h>
#include <sound/soc-acpi-intel-match.h>
+static struct snd_soc_acpi_codecs jsl_7219_98373_codecs = {
+ .num_codecs = 1,
+ .codecs = {"MX98373"}
+};
+
+/*
+ * When adding new entry to the snd_soc_acpi_intel_jsl_machines array,
+ * use .quirk_data member to distinguish different machine driver,
+ * and keep ACPI .id field unchanged for the common codec.
+ */
struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = {
{
.id = "DLGS7219",
.drv_name = "sof_da7219_max98373",
- .machine_quirk = snd_soc_acpi_codec_list,
.sof_fw_filename = "sof-jsl.ri",
.sof_tplg_filename = "sof-jsl-da7219.tplg",
+ .machine_quirk = snd_soc_acpi_codec_list,
+ .quirk_data = &jsl_7219_98373_codecs,
+ },
+ {
+ .id = "DLGS7219",
+ .drv_name = "sof_da7219_max98360a",
+ .sof_fw_filename = "sof-jsl.ri",
+ .sof_tplg_filename = "sof-jsl-da7219-mx98360a.tplg",
},
{},
};
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: Intel: (cosmetic) simplify structure member access" to the asoc tree
@ 2020-03-13 16:17 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-13 16:17 UTC (permalink / raw)
To: Guennadi Liakhovetski; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: Intel: (cosmetic) simplify structure member access
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 42432196cfb01500ec058e8acc8dcfcf27eb76c9 Mon Sep 17 00:00:00 2001
From: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Date: Thu, 12 Mar 2020 14:48:56 -0500
Subject: [PATCH] ASoC: Intel: (cosmetic) simplify structure member access
Fix a clumsy structure member dereference in all machine drivers.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200312194859.4051-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/bdw-rt5650.c | 2 +-
sound/soc/intel/boards/bdw-rt5677.c | 2 +-
sound/soc/intel/boards/broadwell.c | 2 +-
sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +-
sound/soc/intel/boards/bxt_rt298.c | 2 +-
sound/soc/intel/boards/bytcht_da7213.c | 2 +-
sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +-
sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +-
sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +-
sound/soc/intel/boards/cml_rt1011_rt5682.c | 2 +-
sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +-
sound/soc/intel/boards/haswell.c | 2 +-
sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +-
sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +-
sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +-
sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +-
sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +-
sound/soc/intel/boards/sof_da7219_max98373.c | 2 +-
sound/soc/intel/boards/sof_rt5682.c | 2 +-
19 files changed, 19 insertions(+), 19 deletions(-)
diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c
index 1a302436d450..058abf3eec50 100644
--- a/sound/soc/intel/boards/bdw-rt5650.c
+++ b/sound/soc/intel/boards/bdw-rt5650.c
@@ -298,7 +298,7 @@ static int bdw_rt5650_probe(struct platform_device *pdev)
return -ENOMEM;
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5650_card,
mach->mach_params.platform);
diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index bb643c99069d..a94f498388e1 100644
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -412,7 +412,7 @@ static int bdw_rt5677_probe(struct platform_device *pdev)
}
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5677_card,
mach->mach_params.platform);
if (ret)
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index b9c12e24c70b..25178000c6a5 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -283,7 +283,7 @@ static int broadwell_audio_probe(struct platform_device *pdev)
broadwell_rt286.dev = &pdev->dev;
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
ret = snd_soc_fixup_dai_links_platform_name(&broadwell_rt286,
mach->mach_params.platform);
if (ret)
diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c
index 9177401c37a5..061462248bce 100644
--- a/sound/soc/intel/boards/bxt_da7219_max98357a.c
+++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c
@@ -721,7 +721,7 @@ static int broxton_audio_probe(struct platform_device *pdev)
}
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
platform_name = mach->mach_params.platform;
ret = snd_soc_fixup_dai_links_platform_name(&broxton_audio_card,
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index 4b67f261377c..4b5e7f6dbdf1 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -627,7 +627,7 @@ static int broxton_audio_probe(struct platform_device *pdev)
snd_soc_card_set_drvdata(card, ctx);
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
platform_name = mach->mach_params.platform;
ret = snd_soc_fixup_dai_links_platform_name(card,
diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c
index eda7a500cad6..d6b912c013fc 100644
--- a/sound/soc/intel/boards/bytcht_da7213.c
+++ b/sound/soc/intel/boards/bytcht_da7213.c
@@ -231,7 +231,7 @@ static int bytcht_da7213_probe(struct platform_device *pdev)
int ret_val = 0;
int i;
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
card = &bytcht_da7213_card;
card->dev = &pdev->dev;
diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index 70bb86f3342f..ea119d523926 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -553,7 +553,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
/* override plaform name, if required */
snd_soc_card_cht.dev = &pdev->dev;
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
platform_name = mach->mach_params.platform;
ret_val = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cht,
diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c
index 501bad3976fb..34d4e17e3295 100644
--- a/sound/soc/intel/boards/cht_bsw_nau8824.c
+++ b/sound/soc/intel/boards/cht_bsw_nau8824.c
@@ -259,7 +259,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
/* override plaform name, if required */
snd_soc_card_cht.dev = &pdev->dev;
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
platform_name = mach->mach_params.platform;
ret_val = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cht,
diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
index b5b016d493f1..452691db12cc 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5645.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -539,7 +539,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
if (!drv)
return -ENOMEM;
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) {
if (acpi_dev_found(snd_soc_cards[i].codec_id) &&
diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c
index 2a6e5b124099..30de502b4fbb 100644
--- a/sound/soc/intel/boards/cml_rt1011_rt5682.c
+++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c
@@ -451,7 +451,7 @@ static int snd_cml_rt1011_probe(struct platform_device *pdev)
return -ENOMEM;
INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
snd_soc_card_cml.dev = &pdev->dev;
platform_name = mach->mach_params.platform;
diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c
index 8e947bad143c..ea1de8b3f3cd 100644
--- a/sound/soc/intel/boards/glk_rt5682_max98357a.c
+++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c
@@ -604,7 +604,7 @@ static int geminilake_audio_probe(struct platform_device *pdev)
snd_soc_card_set_drvdata(card, ctx);
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
platform_name = mach->mach_params.platform;
ret = snd_soc_fixup_dai_links_platform_name(card, platform_name);
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index 3dadf9bff796..6589fa56873f 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -193,7 +193,7 @@ static int haswell_audio_probe(struct platform_device *pdev)
haswell_rt5640.dev = &pdev->dev;
/* override plaform name, if required */
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
ret = snd_soc_fixup_dai_links_platform_name(&haswell_rt5640,
mach->mach_params.platform);
if (ret)
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index f65feee1c166..20d566e9dd9d 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -962,7 +962,7 @@ static int kabylake_audio_probe(struct platform_device *pdev)
kabylake_audio_card->dev = &pdev->dev;
snd_soc_card_set_drvdata(kabylake_audio_card, ctx);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
if (mach)
dmic_constraints = mach->mach_params.dmic_num == 2 ?
&constraints_dmic_2ch : &constraints_dmic_channels;
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 341bb47311a6..6493ede89300 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -772,7 +772,7 @@ static int kabylake_audio_probe(struct platform_device *pdev)
kabylake_audio_card.dev = &pdev->dev;
snd_soc_card_set_drvdata(&kabylake_audio_card, ctx);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
if (mach)
dmic_constraints = mach->mach_params.dmic_num == 2 ?
&constraints_dmic_2ch : &constraints_dmic_channels;
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
index fe2d3a23a4ef..3be764299ab0 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_generic.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -182,7 +182,7 @@ static int skl_hda_audio_probe(struct platform_device *pdev)
INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
if (!mach)
return -EINVAL;
diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
index e6de3b28d840..8216c15fc8da 100644
--- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c
+++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
@@ -660,7 +660,7 @@ static int skylake_audio_probe(struct platform_device *pdev)
skylake_audio_card.dev = &pdev->dev;
snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
if (mach)
dmic_constraints = mach->mach_params.dmic_num == 2 ?
&constraints_dmic_2ch : &constraints_dmic_channels;
diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
index c99c8b23e509..6f68712ffce9 100644
--- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
+++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
@@ -703,7 +703,7 @@ static int skylake_audio_probe(struct platform_device *pdev)
skylake_audio_card.dev = &pdev->dev;
snd_soc_card_set_drvdata(&skylake_audio_card, ctx);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
if (mach)
dmic_constraints = mach->mach_params.dmic_num == 2 ?
&constraints_dmic_2ch : &constraints_dmic_channels;
diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c
index 8f44f13d2848..8c657da5fcf0 100644
--- a/sound/soc/intel/boards/sof_da7219_max98373.c
+++ b/sound/soc/intel/boards/sof_da7219_max98373.c
@@ -335,7 +335,7 @@ static int audio_probe(struct platform_device *pdev)
card = (struct snd_soc_card *)pdev->id_entry->driver_data;
card->dev = &pdev->dev;
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
ret = snd_soc_fixup_dai_links_platform_name(card,
mach->mach_params.platform);
if (ret)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 5d878873a8e0..99b5a5e01e38 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -604,7 +604,7 @@ static int sof_audio_probe(struct platform_device *pdev)
dmi_check_system(sof_rt5682_quirk_table);
- mach = (&pdev->dev)->platform_data;
+ mach = pdev->dev.platform_data;
/* A speaker amp might not be present when the quirk claims one is.
* Detect this via whether the machine driver match includes quirk_data.
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations" to the asoc tree
@ 2020-03-13 16:17 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-13 16:17 UTC (permalink / raw)
To: Guennadi Liakhovetski; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From ca841843a3a8038494e48968c2fd1c7ec5473ce3 Mon Sep 17 00:00:00 2001
From: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Date: Thu, 12 Mar 2020 14:48:55 -0500
Subject: [PATCH] ASoC: Intel: skylake: (cosmetic) remove redundant variable
initialisations
Variables, used as loop iterators, don't need to be initialised.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200312194859.4051-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/skylake/skl.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index f755ca2484cf..1aa8114a4f77 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -359,7 +359,7 @@ static int skl_resume(struct device *dev)
struct pci_dev *pci = to_pci_dev(dev);
struct hdac_bus *bus = pci_get_drvdata(pci);
struct skl_dev *skl = bus_to_skl(bus);
- struct hdac_ext_link *hlink = NULL;
+ struct hdac_ext_link *hlink;
int ret;
/*
@@ -794,7 +794,7 @@ static void skl_probe_work(struct work_struct *work)
{
struct skl_dev *skl = container_of(work, struct skl_dev, probe_work);
struct hdac_bus *bus = skl_to_bus(skl);
- struct hdac_ext_link *hlink = NULL;
+ struct hdac_ext_link *hlink;
int err;
if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) {
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations" to the asoc tree
@ 2020-03-13 16:17 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-13 16:17 UTC (permalink / raw)
To: Guennadi Liakhovetski; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From c4aafb337d311556d4448f8de857ca2683f5ca5b Mon Sep 17 00:00:00 2001
From: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Date: Thu, 12 Mar 2020 14:48:54 -0500
Subject: [PATCH] ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable
initialisations
Remove several redundant variable initialisations.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200312194859.4051-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/hdac_hdmi.c | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index e6558475e006..fba9b749839d 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -1998,11 +1998,11 @@ static struct hdac_hdmi_drv_data intel_drv_data = {
static int hdac_hdmi_dev_probe(struct hdac_device *hdev)
{
- struct hdac_hdmi_priv *hdmi_priv = NULL;
+ struct hdac_hdmi_priv *hdmi_priv;
struct snd_soc_dai_driver *hdmi_dais = NULL;
- struct hdac_ext_link *hlink = NULL;
+ struct hdac_ext_link *hlink;
int num_dais = 0;
- int ret = 0;
+ int ret;
struct hdac_driver *hdrv = drv_to_hdac_driver(hdev->dev.driver);
const struct hda_device_id *hdac_id = hdac_get_device_id(hdev, hdrv);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC" to the asoc tree
@ 2020-03-13 16:17 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-13 16:17 UTC (permalink / raw)
To: Kai Vehmanen; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 42c67753cae12f68b936901792ecd367e8dfca9b Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Thu, 12 Mar 2020 14:48:53 -0500
Subject: [PATCH] ASoC: SOF: Intel: hda: remove
SND_SOC_SOF_HDA_COMMON_HDMI_CODEC
To help user-space with HDMI codec driver transition, both
a kernel module parameter and a kernel option were initially
provided to configure default behaviour of SOF on Intel hardware
with commit 139c7febad1a ("ASoC: SOF: Intel: add support for
snd-hda-codec-hdmi").
As hdac-hdmi is already now lagging in features compared to
snd-hda-codec-hdmi, move ahead with the transition and remove
the build option to select between the two, and instead default
to snd-hda-codec-hdmi if it is enabled in kernel build.
The old behaviour of using hdac-hdmi driver can still be forced
via the kernel module parameter.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200312194859.4051-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/Kconfig | 11 -----------
sound/soc/sof/intel/hda.c | 3 +--
2 files changed, 1 insertion(+), 13 deletions(-)
diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index 3bc64dee7c39..c9a2bee4b55c 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -324,17 +324,6 @@ config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1
Say Y if you want to enable DMI Link L1
If unsure, select "N".
-config SND_SOC_SOF_HDA_COMMON_HDMI_CODEC
- bool "SOF common HDA HDMI codec driver"
- depends on SND_SOC_SOF_HDA_LINK
- depends on SND_HDA_CODEC_HDMI
- default SND_HDA_CODEC_HDMI
- help
- This adds support for HDMI audio by using the common HDA
- HDMI/DisplayPort codec driver.
- Say Y if you want to use the common codec driver with SOF.
- If unsure select "Y".
-
endif ## SND_SOC_SOF_HDA_COMMON
config SND_SOC_SOF_HDA_LINK_BASELINE
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 7ca887041a34..1de750a1dd19 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -54,8 +54,7 @@ static int hda_dmic_num = -1;
module_param_named(dmic_num, hda_dmic_num, int, 0444);
MODULE_PARM_DESC(dmic_num, "SOF HDA DMIC number");
-static bool hda_codec_use_common_hdmi =
- IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_COMMON_HDMI_CODEC);
+static bool hda_codec_use_common_hdmi = IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI);
module_param_named(use_common_hdmi, hda_codec_use_common_hdmi, bool, 0444);
MODULE_PARM_DESC(use_common_hdmi, "SOF HDA use common HDMI codec driver");
#endif
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms" to the asoc tree
@ 2020-03-13 16:17 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-13 16:17 UTC (permalink / raw)
To: Kai Vehmanen; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 15a5a89597e57e67d4dde1d57fa85105c5930bb3 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Thu, 12 Mar 2020 14:48:52 -0500
Subject: [PATCH] ASoC: Intel: sof_pcm512x: make HDMI optional for all
platforms
Make HDMI optional for APL and later platforms. If no HDMI codec
is found on the HDA bus, the graphics side driver is missing or
correct codec driver is not part of kernel build, codec_mask
reflects this and HDMI is disabled. The DSP topology will still
have the links for HDMI, so connect these to dummy codec to avoid
failures in topology loading.
This change also fixes a kernel oops that was triggered if
sof_pcm512x was used with SOF configured to use hdac-hdmi (can be
done via "use_common_hdmi=0" or by selecting
CONFIG_SND_SOC_SOF_HDA_COMMON_HDMI_CODEC=n). This is not a supported
configuration.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200312194859.4051-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/sof_pcm512x.c | 38 +++++++++++++++++++++-------
1 file changed, 29 insertions(+), 9 deletions(-)
diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c
index 626153bd71e7..4ce707b6eb79 100644
--- a/sound/soc/intel/boards/sof_pcm512x.c
+++ b/sound/soc/intel/boards/sof_pcm512x.c
@@ -27,6 +27,8 @@
#define SOF_PCM512X_SSP_CODEC(quirk) ((quirk) & GENMASK(3, 0))
#define SOF_PCM512X_SSP_CODEC_MASK (GENMASK(3, 0))
+#define IDISP_CODEC_MASK 0x4
+
/* Default: SSP5 */
static unsigned long sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(5);
@@ -40,6 +42,7 @@ struct sof_hdmi_pcm {
struct sof_card_private {
struct list_head hdmi_pcm_list;
+ bool idisp_codec;
};
static int sof_pcm512x_quirk_cb(const struct dmi_system_id *id)
@@ -136,6 +139,9 @@ static int sof_card_late_probe(struct snd_soc_card *card)
if (list_empty(&ctx->hdmi_pcm_list))
return -EINVAL;
+ if (!ctx->idisp_codec)
+ return 0;
+
pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head);
return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component);
@@ -214,7 +220,8 @@ SND_SOC_DAILINK_DEF(dmic_component,
static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
int ssp_codec,
int dmic_be_num,
- int hdmi_num)
+ int hdmi_num,
+ bool idisp_codec)
{
struct snd_soc_dai_link_component *idisp_components;
struct snd_soc_dai_link_component *cpus;
@@ -316,11 +323,19 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
if (!links[id].cpus->dai_name)
goto devm_err;
- idisp_components[i - 1].name = "ehdaudio0D2";
- idisp_components[i - 1].dai_name = devm_kasprintf(dev,
- GFP_KERNEL,
- "intel-hdmi-hifi%d",
- i);
+ /*
+ * topology cannot be loaded if codec is missing, so
+ * use the dummy codec if needed
+ */
+ if (idisp_codec) {
+ idisp_components[i - 1].name = "ehdaudio0D2";
+ idisp_components[i - 1].dai_name =
+ devm_kasprintf(dev, GFP_KERNEL,
+ "intel-hdmi-hifi%d", i);
+ } else {
+ idisp_components[i - 1].name = "snd-soc-dummy";
+ idisp_components[i - 1].dai_name = "snd-soc-dummy-dai";
+ }
if (!idisp_components[i - 1].dai_name)
goto devm_err;
@@ -341,8 +356,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
static int sof_audio_probe(struct platform_device *pdev)
{
+ struct snd_soc_acpi_mach *mach = pdev->dev.platform_data;
struct snd_soc_dai_link *dai_links;
- struct snd_soc_acpi_mach *mach;
struct sof_card_private *ctx;
int dmic_be_num, hdmi_num;
int ret, ssp_codec;
@@ -360,6 +375,11 @@ static int sof_audio_probe(struct platform_device *pdev)
} else {
dmic_be_num = 2;
#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI)
+ if (mach->mach_params.common_hdmi_codec_drv &&
+ (mach->mach_params.codec_mask & IDISP_CODEC_MASK))
+ ctx->idisp_codec = true;
+
+ /* links are always present in topology */
hdmi_num = 3;
#endif
}
@@ -374,7 +394,8 @@ static int sof_audio_probe(struct platform_device *pdev)
sof_audio_card_pcm512x.num_links = 1 + dmic_be_num + hdmi_num;
dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec,
- dmic_be_num, hdmi_num);
+ dmic_be_num, hdmi_num,
+ ctx->idisp_codec);
if (!dai_links)
return -ENOMEM;
@@ -383,7 +404,6 @@ static int sof_audio_probe(struct platform_device *pdev)
INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
sof_audio_card_pcm512x.dev = &pdev->dev;
- mach = (&pdev->dev)->platform_data;
/* set platform name for each dailink */
ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_pcm512x,
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI" to the asoc tree
@ 2020-03-13 16:17 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-13 16:17 UTC (permalink / raw)
To: Kai Vehmanen; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 4399afd21a0169a125a16c2dbba68f7d9a2cffdb Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Thu, 12 Mar 2020 14:48:51 -0500
Subject: [PATCH] ASoC: Intel: sof_pcm512x: drop reverse deps for
SND_HDA_CODEC_HDMI
Having a reverse dependency to a config that has its own additional
dependencies, is generally not recommended. And this applies to
select statements for SND_HDA_CODEC_HDMI, e.g. the case where SND_HDA
and SND_SOC_SOF_HDA are built as modules, but the machine driver is
built-in, leading to compile errors (reported as
i386-randconfig-e003-20200206).
Give up on trying to define different dependencies based on SOF/SST
selection, and simply add a "depends on" for SND_HDA_CODEC_HDMI. This
fixes the issue with randconfigs. Only downside is that SND_HDA_CODEC_HDMI
may be built unnecessarily in some cases, but this seems like the lesser
evil.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200312194859.4051-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 67d85a7be559..6833ef548710 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -464,8 +464,8 @@ config SND_SOC_INTEL_SOF_PCM512x_MACH
depends on I2C && ACPI
depends on (SND_SOC_SOF_HDA_AUDIO_CODEC && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\
(SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST))
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_PCM512x_I2C
- select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
help
This adds support for ASoC machine driver for SOF platforms
with TI PCM512x I2S audio codec.
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI" to the asoc tree
@ 2020-03-13 16:18 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-13 16:18 UTC (permalink / raw)
To: Kai Vehmanen
Cc: alsa-devel, Mark Brown, kbuild test robot, Pierre-Louis Bossart
The patch
ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 5c82813ce43e3a2017c43b32a57dc7a8802d9ad4 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Thu, 12 Mar 2020 14:48:50 -0500
Subject: [PATCH] ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI
Having a reverse dependency to a config that has its own additional
dependencies, is generally not recommended. And this applies to
select statements for SND_HDA_CODEC_HDMI, e.g. the case where SND_HDA
and SND_SOC_SOF_HDA are built as modules, but the machine driver is
built-in, leading to compile errors (reported as
i386-randconfig-e003-20200206).
Give up on trying to define different dependencies based on SOF/SST
selection, and simply add a "depends on" for SND_HDA_CODEC_HDMI. This
fixes the issue with randconfigs. Only downside is that SND_HDA_CODEC_HDMI
may be built unnecessarily in some cases, but this seems like the lesser
evil.
Fixes: aa2b4a5 ('ASoC: Intel: boards: fix incorrect HDMI Kconfig dependency')
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200312194859.4051-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/Kconfig | 13 +++++++------
1 file changed, 7 insertions(+), 6 deletions(-)
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig
index 755e1de19df9..67d85a7be559 100644
--- a/sound/soc/intel/boards/Kconfig
+++ b/sound/soc/intel/boards/Kconfig
@@ -289,7 +289,6 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC
select SND_SOC_DA7219
select SND_SOC_MAX98357A
select SND_SOC_DMIC
- select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
select SND_SOC_HDAC_HDMI
config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
@@ -302,6 +301,7 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH
tristate "Broxton with DA7219 and MAX98357A in I2S Mode"
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
help
This adds support for ASoC machine driver for Broxton-P platforms
@@ -402,6 +402,7 @@ config SND_SOC_INTEL_GLK_DA7219_MAX98357A_MACH
tristate "GLK with DA7219 and MAX98357A in I2S Mode"
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
help
This adds support for ASoC machine driver for Geminilake platforms
@@ -413,10 +414,10 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
tristate "GLK with RT5682 and MAX98357A in I2S Mode"
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_RT5682
select SND_SOC_MAX98357A
select SND_SOC_DMIC
- select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
select SND_SOC_HDAC_HDMI
help
This adds support for ASoC machine driver for Geminilake platforms
@@ -430,7 +431,7 @@ if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC
config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
tristate "SKL/KBL/BXT/APL with HDA Codecs"
- select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_HDAC_HDMI
select SND_SOC_DMIC
# SND_SOC_HDAC_HDA is already selected
@@ -448,9 +449,9 @@ config SND_SOC_INTEL_SOF_RT5682_MACH
depends on I2C && ACPI
depends on (SND_SOC_SOF_HDA_LINK && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\
(SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST))
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_RT5682
select SND_SOC_DMIC
- select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
select SND_SOC_HDAC_HDMI
help
This adds support for ASoC machine driver for SOF platforms
@@ -490,11 +491,11 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH
tristate "CML with RT1011 and RT5682 in I2S Mode"
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_RT1011
select SND_SOC_RT5682
select SND_SOC_DMIC
select SND_SOC_HDAC_HDMI
- select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
help
This adds support for ASoC machine driver for SOF platform with
RT1011 + RT5682 I2S codec.
@@ -509,10 +510,10 @@ config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH
tristate "SOF with DA7219 and MAX98373 in I2S Mode"
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
+ depends on SND_HDA_CODEC_HDMI
select SND_SOC_DA7219
select SND_SOC_MAX98373
select SND_SOC_DMIC
- select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
help
This adds support for ASoC machine driver for SOF platforms
with DA7219 + MAX98373 I2S audio codec.
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ALSA: compress: bump the version" to the asoc tree
@ 2020-03-16 18:07 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-16 18:07 UTC (permalink / raw)
To: Vinod Koul; +Cc: alsa-devel, Mark Brown
The patch
ALSA: compress: bump the version
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 54ce83a3080c3d06b14faa7533a9adb4e158dcea Mon Sep 17 00:00:00 2001
From: Vinod Koul <vkoul@kernel.org>
Date: Mon, 16 Mar 2020 11:22:21 +0530
Subject: [PATCH] ALSA: compress: bump the version
We have added support for bunch of new decoders and parameters for
decoders. To help users find the support bump the version up to 0,2,0.
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200316055221.1944464-10-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/uapi/sound/compress_offload.h | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h
index 56d95673ce0f..7184265c0b0d 100644
--- a/include/uapi/sound/compress_offload.h
+++ b/include/uapi/sound/compress_offload.h
@@ -31,7 +31,7 @@
#include <sound/compress_params.h>
-#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 2)
+#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 2, 0)
/**
* struct snd_compressed_buffer - compressed buffer
* @fragment_size: size of buffer fragment in bytes
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: qcom: q6asm: add support for alac and ape configs" to the asoc tree
@ 2020-03-16 18:08 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-16 18:08 UTC (permalink / raw)
To: Vinod Koul; +Cc: alsa-devel, Mark Brown
The patch
ASoC: qcom: q6asm: add support for alac and ape configs
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 7076bf4da0b1b53dd5d96dc930c9899a8ad6f217 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vkoul@kernel.org>
Date: Mon, 16 Mar 2020 11:22:19 +0530
Subject: [PATCH] ASoC: qcom: q6asm: add support for alac and ape configs
Qualcomm DSPs expect ALAC and APE configs to be send for decoders,
so add the API to program the respective config to the DSP.
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200316055221.1944464-8-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/qdsp6/q6asm.c | 118 +++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6asm.h | 32 ++++++++++
2 files changed, 150 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 4cec95c657ba..0e0e8f7a460a 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -48,6 +48,8 @@
#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4
#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
+#define ASM_MEDIA_FMT_ALAC 0x00012f31
+#define ASM_MEDIA_FMT_APE 0x00012f32
#define ASM_LEGACY_STREAM_SESSION 0
@@ -133,6 +135,36 @@ struct asm_wmaprov10_fmt_blk_v2 {
u32 advanced_enc_options2;
} __packed;
+struct asm_alac_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u32 frame_length;
+ u8 compatible_version;
+ u8 bit_depth;
+ u8 pb;
+ u8 mb;
+ u8 kb;
+ u8 num_channels;
+ u16 max_run;
+ u32 max_frame_bytes;
+ u32 avg_bit_rate;
+ u32 sample_rate;
+ u32 channel_layout_tag;
+} __packed;
+
+struct asm_ape_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u16 compatible_version;
+ u16 compression_level;
+ u32 format_flags;
+ u32 blocks_per_frame;
+ u32 final_frame_blocks;
+ u32 total_frames;
+ u16 bits_per_sample;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 seek_table_present;
+} __packed;
+
struct asm_stream_cmd_set_encdec_param {
u32 param_id;
u32 param_size;
@@ -941,6 +973,12 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
goto err;
}
break;
+ case SND_AUDIOCODEC_ALAC:
+ open->dec_fmt_id = ASM_MEDIA_FMT_ALAC;
+ break;
+ case SND_AUDIOCODEC_APE:
+ open->dec_fmt_id = ASM_MEDIA_FMT_APE;
+ break;
default:
dev_err(ac->dev, "Invalid format 0x%x\n", format);
rc = -EINVAL;
@@ -1198,6 +1236,86 @@ int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
}
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+ struct q6asm_alac_cfg *cfg)
+{
+ struct asm_alac_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+
+ fmt->frame_length = cfg->frame_length;
+ fmt->compatible_version = cfg->compatible_version;
+ fmt->bit_depth = cfg->bit_depth;
+ fmt->num_channels = cfg->num_channels;
+ fmt->max_run = cfg->max_run;
+ fmt->max_frame_bytes = cfg->max_frame_bytes;
+ fmt->avg_bit_rate = cfg->avg_bit_rate;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->channel_layout_tag = cfg->channel_layout_tag;
+ fmt->pb = cfg->pb;
+ fmt->mb = cfg->mb;
+ fmt->kb = cfg->kb;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
+
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+ struct q6asm_ape_cfg *cfg)
+{
+ struct asm_ape_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+
+ fmt->compatible_version = cfg->compatible_version;
+ fmt->compression_level = cfg->compression_level;
+ fmt->format_flags = cfg->format_flags;
+ fmt->blocks_per_frame = cfg->blocks_per_frame;
+ fmt->final_frame_blocks = cfg->final_frame_blocks;
+ fmt->total_frames = cfg->total_frames;
+ fmt->bits_per_sample = cfg->bits_per_sample;
+ fmt->num_channels = cfg->num_channels;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->seek_table_present = cfg->seek_table_present;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
+
/**
* q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
*
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 5d9fbc75688c..38a207d6cd95 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -58,6 +58,34 @@ struct q6asm_wma_cfg {
u32 adv_enc_options2;
};
+struct q6asm_alac_cfg {
+ u32 frame_length;
+ u8 compatible_version;
+ u8 bit_depth;
+ u8 pb;
+ u8 mb;
+ u8 kb;
+ u8 num_channels;
+ u16 max_run;
+ u32 max_frame_bytes;
+ u32 avg_bit_rate;
+ u32 sample_rate;
+ u32 channel_layout_tag;
+};
+
+struct q6asm_ape_cfg {
+ u16 compatible_version;
+ u16 compression_level;
+ u32 format_flags;
+ u32 blocks_per_frame;
+ u32 final_frame_blocks;
+ u32 total_frames;
+ u16 bits_per_sample;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 seek_table_present;
+};
+
typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
void *payload, void *priv);
struct audio_client;
@@ -86,6 +114,10 @@ int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
struct q6asm_wma_cfg *cfg);
int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
struct q6asm_wma_cfg *cfg);
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+ struct q6asm_alac_cfg *cfg);
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+ struct q6asm_ape_cfg *cfg);
int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
uint32_t lsw_ts);
int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ALSA: compress: add alac & ape decoder params" to the asoc tree
@ 2020-03-16 18:08 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-16 18:08 UTC (permalink / raw)
To: Vinod Koul; +Cc: alsa-devel, Mark Brown
The patch
ALSA: compress: add alac & ape decoder params
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 0f546d6f0292fb624b4cf0cc70096ddb9ea070e6 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vkoul@kernel.org>
Date: Mon, 16 Mar 2020 11:22:18 +0530
Subject: [PATCH] ALSA: compress: add alac & ape decoder params
Add ALAC (Apple Lossless Audio Codec) and APE (Monkey's Lossless Audio
Codec) defines and parameters required to configure these.
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200316055221.1944464-7-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/uapi/sound/compress_params.h | 26 +++++++++++++++++++++++++-
1 file changed, 25 insertions(+), 1 deletion(-)
diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index bf6f7155e775..79b14389ae41 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -75,7 +75,9 @@
#define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C)
#define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D)
#define SND_AUDIOCODEC_BESPOKE ((__u32) 0x0000000E)
-#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_BESPOKE
+#define SND_AUDIOCODEC_ALAC ((__u32) 0x0000000F)
+#define SND_AUDIOCODEC_APE ((__u32) 0x00000010)
+#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_APE
/*
* Profile and modes are listed with bit masks. This allows for a
@@ -336,6 +338,26 @@ struct snd_dec_wma {
__u32 reserved;
} __attribute__((packed, aligned(4)));
+struct snd_dec_alac {
+ __u32 frame_length;
+ __u8 compatible_version;
+ __u8 pb;
+ __u8 mb;
+ __u8 kb;
+ __u32 max_run;
+ __u32 max_frame_bytes;
+} __attribute__((packed, aligned(4)));
+
+struct snd_dec_ape {
+ __u16 compatible_version;
+ __u16 compression_level;
+ __u32 format_flags;
+ __u32 blocks_per_frame;
+ __u32 final_frame_blocks;
+ __u32 total_frames;
+ __u32 seek_table_present;
+} __attribute__((packed, aligned(4)));
+
union snd_codec_options {
struct snd_enc_wma wma;
struct snd_enc_vorbis vorbis;
@@ -344,6 +366,8 @@ union snd_codec_options {
struct snd_enc_generic generic;
struct snd_dec_flac flac_d;
struct snd_dec_wma wma_d;
+ struct snd_dec_alac alac_d;
+ struct snd_dec_ape ape_d;
} __attribute__((packed, aligned(4)));
/** struct snd_codec_desc - description of codec capabilities
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: qcom: q6asm-dai: add support to wma decoder" to the asoc tree
@ 2020-03-16 18:08 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-16 18:08 UTC (permalink / raw)
To: Vinod Koul; +Cc: alsa-devel, Mark Brown
The patch
ASoC: qcom: q6asm-dai: add support to wma decoder
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 40519a1c02303ebdf09e07fe57a52c2d1d188b01 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vkoul@kernel.org>
Date: Mon, 16 Mar 2020 11:22:17 +0530
Subject: [PATCH] ASoC: qcom: q6asm-dai: add support to wma decoder
Qualcomm DSPs also supports the wma decoder, so add support for wma
decoder and convert the snd_codec_params to qdsp format.
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200316055221.1944464-6-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/qdsp6/q6asm-dai.c | 67 +++++++++++++++++++++++++++++++-
1 file changed, 66 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index bc0e3f7cfd8e..fa685fe4a027 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -629,10 +629,13 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
int dir = stream->direction;
struct q6asm_dai_data *pdata;
struct q6asm_flac_cfg flac_cfg;
+ struct q6asm_wma_cfg wma_cfg;
+ unsigned int wma_v9 = 0;
struct device *dev = c->dev;
int ret;
union snd_codec_options *codec_options;
struct snd_dec_flac *flac;
+ struct snd_dec_wma *wma;
codec_options = &(prtd->codec_param.codec.options);
@@ -694,6 +697,67 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
return -EIO;
}
break;
+
+ case SND_AUDIOCODEC_WMA:
+ wma = &codec_options->wma_d;
+
+ memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
+
+ wma_cfg.sample_rate = params->codec.sample_rate;
+ wma_cfg.num_channels = params->codec.ch_in;
+ wma_cfg.bytes_per_sec = params->codec.bit_rate / 8;
+ wma_cfg.block_align = params->codec.align;
+ wma_cfg.bits_per_sample = prtd->bits_per_sample;
+ wma_cfg.enc_options = wma->encoder_option;
+ wma_cfg.adv_enc_options = wma->adv_encoder_option;
+ wma_cfg.adv_enc_options2 = wma->adv_encoder_option2;
+
+ if (wma_cfg.num_channels == 1)
+ wma_cfg.channel_mask = 4; /* Mono Center */
+ else if (wma_cfg.num_channels == 2)
+ wma_cfg.channel_mask = 3; /* Stereo FL/FR */
+ else
+ return -EINVAL;
+
+ /* check the codec profile */
+ switch (params->codec.profile) {
+ case SND_AUDIOPROFILE_WMA9:
+ wma_cfg.fmtag = 0x161;
+ wma_v9 = 1;
+ break;
+
+ case SND_AUDIOPROFILE_WMA10:
+ wma_cfg.fmtag = 0x166;
+ break;
+
+ case SND_AUDIOPROFILE_WMA9_PRO:
+ wma_cfg.fmtag = 0x162;
+ break;
+
+ case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+ wma_cfg.fmtag = 0x163;
+ break;
+
+ case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+ wma_cfg.fmtag = 0x167;
+ break;
+
+ default:
+ dev_err(dev, "Unknown WMA profile:%x\n",
+ params->codec.profile);
+ return -EIO;
+ }
+
+ if (wma_v9)
+ ret = q6asm_stream_media_format_block_wma_v9(
+ prtd->audio_client, &wma_cfg);
+ else
+ ret = q6asm_stream_media_format_block_wma_v10(
+ prtd->audio_client, &wma_cfg);
+ if (ret < 0) {
+ dev_err(dev, "WMA9 CMD failed:%d\n", ret);
+ return -EIO;
+ }
default:
break;
}
@@ -793,9 +857,10 @@ static int q6asm_dai_compr_get_caps(struct snd_compr_stream *stream,
caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
- caps->num_codecs = 2;
+ caps->num_codecs = 3;
caps->codecs[0] = SND_AUDIOCODEC_MP3;
caps->codecs[1] = SND_AUDIOCODEC_FLAC;
+ caps->codecs[2] = SND_AUDIOCODEC_WMA;
return 0;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: qcom: q6asm: add support to wma config" to the asoc tree
@ 2020-03-16 18:08 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-16 18:08 UTC (permalink / raw)
To: Vinod Koul; +Cc: alsa-devel, Mark Brown
The patch
ASoC: qcom: q6asm: add support to wma config
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 97163eadf18bff306b1cf8a8fa81938c5509899c Mon Sep 17 00:00:00 2001
From: Vinod Koul <vkoul@kernel.org>
Date: Mon, 16 Mar 2020 11:22:16 +0530
Subject: [PATCH] ASoC: qcom: q6asm: add support to wma config
Qualcomm DSPs expect wma v9 and wma v10 configs to be set for wma
decoders, so add the API to program the respective wma config to the DSP
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200316055221.1944464-5-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/qdsp6/q6asm.c | 123 +++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6asm.h | 17 +++++
2 files changed, 140 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 64eb7b6ba305..4cec95c657ba 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -39,6 +39,8 @@
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
#define ASM_MEDIA_FMT_MP3 0x00010BE9
#define ASM_MEDIA_FMT_FLAC 0x00010C16
+#define ASM_MEDIA_FMT_WMA_V9 0x00010DA8
+#define ASM_MEDIA_FMT_WMA_V10 0x00010DA7
#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
#define ASM_DATA_CMD_READ_V2 0x00010DAC
#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
@@ -104,6 +106,33 @@ struct asm_flac_fmt_blk_v2 {
u16 reserved;
} __packed;
+struct asm_wmastdv9_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u16 fmtag;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 bytes_per_sec;
+ u16 blk_align;
+ u16 bits_per_sample;
+ u32 channel_mask;
+ u16 enc_options;
+ u16 reserved;
+} __packed;
+
+struct asm_wmaprov10_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u16 fmtag;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 bytes_per_sec;
+ u16 blk_align;
+ u16 bits_per_sample;
+ u32 channel_mask;
+ u16 enc_options;
+ u16 advanced_enc_options1;
+ u32 advanced_enc_options2;
+} __packed;
+
struct asm_stream_cmd_set_encdec_param {
u32 param_id;
u32 param_size;
@@ -894,6 +923,24 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
case SND_AUDIOCODEC_FLAC:
open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
break;
+ case SND_AUDIOCODEC_WMA:
+ switch (codec_profile) {
+ case SND_AUDIOPROFILE_WMA9:
+ open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9;
+ break;
+ case SND_AUDIOPROFILE_WMA10:
+ case SND_AUDIOPROFILE_WMA9_PRO:
+ case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+ case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+ open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10;
+ break;
+ default:
+ dev_err(ac->dev, "Invalid codec profile 0x%x\n",
+ codec_profile);
+ rc = -EINVAL;
+ goto err;
+ }
+ break;
default:
dev_err(ac->dev, "Invalid format 0x%x\n", format);
rc = -EINVAL;
@@ -1075,6 +1122,82 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac,
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
+
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+ struct q6asm_wma_cfg *cfg)
+{
+ struct asm_wmastdv9_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+ fmt->fmtag = cfg->fmtag;
+ fmt->num_channels = cfg->num_channels;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->bytes_per_sec = cfg->bytes_per_sec;
+ fmt->blk_align = cfg->block_align;
+ fmt->bits_per_sample = cfg->bits_per_sample;
+ fmt->channel_mask = cfg->channel_mask;
+ fmt->enc_options = cfg->enc_options;
+ fmt->reserved = 0;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
+
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+ struct q6asm_wma_cfg *cfg)
+{
+ struct asm_wmaprov10_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+ fmt->fmtag = cfg->fmtag;
+ fmt->num_channels = cfg->num_channels;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->bytes_per_sec = cfg->bytes_per_sec;
+ fmt->blk_align = cfg->block_align;
+ fmt->bits_per_sample = cfg->bits_per_sample;
+ fmt->channel_mask = cfg->channel_mask;
+ fmt->enc_options = cfg->enc_options;
+ fmt->advanced_enc_options1 = cfg->adv_enc_options;
+ fmt->advanced_enc_options2 = cfg->adv_enc_options2;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
+
/**
* q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
*
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 1cff7f68b95d..5d9fbc75688c 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -45,6 +45,19 @@ struct q6asm_flac_cfg {
u16 md5_sum;
};
+struct q6asm_wma_cfg {
+ u32 fmtag;
+ u32 num_channels;
+ u32 sample_rate;
+ u32 bytes_per_sec;
+ u32 block_align;
+ u32 bits_per_sample;
+ u32 channel_mask;
+ u32 enc_options;
+ u32 adv_enc_options;
+ u32 adv_enc_options2;
+};
+
typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
void *payload, void *priv);
struct audio_client;
@@ -69,6 +82,10 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
uint16_t bits_per_sample);
int q6asm_stream_media_format_block_flac(struct audio_client *ac,
struct q6asm_flac_cfg *cfg);
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+ struct q6asm_wma_cfg *cfg);
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+ struct q6asm_wma_cfg *cfg);
int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
uint32_t lsw_ts);
int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: qcom: q6asm: pass codec profile to q6asm_open_write" to the asoc tree
@ 2020-03-16 18:08 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-16 18:08 UTC (permalink / raw)
To: Vinod Koul; +Cc: alsa-devel, Mark Brown
The patch
ASoC: qcom: q6asm: pass codec profile to q6asm_open_write
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 8504a72f7ce231989cf2a82f0cdbba5b72f65015 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vkoul@kernel.org>
Date: Mon, 16 Mar 2020 11:22:15 +0530
Subject: [PATCH] ASoC: qcom: q6asm: pass codec profile to q6asm_open_write
Codec profile is required to be passed for WMA codecs so that we know
the codec profile present and tell DSP accordingly, so update this API
to pass the codec profile as argument
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200316055221.1944464-4-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++--
sound/soc/qcom/qdsp6/q6asm.c | 2 +-
sound/soc/qcom/qdsp6/q6asm.h | 2 +-
3 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 8b48815ff918..bc0e3f7cfd8e 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -252,7 +252,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
- prtd->bits_per_sample);
+ 0, prtd->bits_per_sample);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
prtd->bits_per_sample);
@@ -654,7 +654,7 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream,
prtd->bits_per_sample = 16;
if (dir == SND_COMPRESS_PLAYBACK) {
ret = q6asm_open_write(prtd->audio_client, params->codec.id,
- prtd->bits_per_sample);
+ params->codec.profile, prtd->bits_per_sample);
if (ret < 0) {
dev_err(dev, "q6asm_open_write failed\n");
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 36e0eab13a98..64eb7b6ba305 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -858,7 +858,7 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
* Return: Will be an negative value on error or zero on success
*/
int q6asm_open_write(struct audio_client *ac, uint32_t format,
- uint16_t bits_per_sample)
+ u32 codec_profile, uint16_t bits_per_sample)
{
struct asm_stream_cmd_open_write_v3 *open;
struct apr_pkt *pkt;
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 6764f55f7078..1cff7f68b95d 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -55,7 +55,7 @@ void q6asm_audio_client_free(struct audio_client *ac);
int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
uint32_t lsw_ts, uint32_t flags);
int q6asm_open_write(struct audio_client *ac, uint32_t format,
- uint16_t bits_per_sample);
+ u32 codec_profile, uint16_t bits_per_sample);
int q6asm_open_read(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ALSA: compress: Add wma decoder params" to the asoc tree
@ 2020-03-16 18:08 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-16 18:08 UTC (permalink / raw)
To: Vinod Koul; +Cc: alsa-devel, Mark Brown
The patch
ALSA: compress: Add wma decoder params
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 20ff1456d26807aa5441be02f0f5459014664aad Mon Sep 17 00:00:00 2001
From: Vinod Koul <vkoul@kernel.org>
Date: Mon, 16 Mar 2020 11:22:14 +0530
Subject: [PATCH] ALSA: compress: Add wma decoder params
Some WMA decoders like WMAv10 etc need some additional encoder option
parameters, so add these as WMA decoder params.
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200316055221.1944464-3-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/uapi/sound/compress_params.h | 8 ++++++++
1 file changed, 8 insertions(+)
diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index a47d9df0fd7b..bf6f7155e775 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -329,6 +329,13 @@ struct snd_dec_flac {
__u16 reserved;
} __attribute__((packed, aligned(4)));
+struct snd_dec_wma {
+ __u32 encoder_option;
+ __u32 adv_encoder_option;
+ __u32 adv_encoder_option2;
+ __u32 reserved;
+} __attribute__((packed, aligned(4)));
+
union snd_codec_options {
struct snd_enc_wma wma;
struct snd_enc_vorbis vorbis;
@@ -336,6 +343,7 @@ union snd_codec_options {
struct snd_enc_flac flac;
struct snd_enc_generic generic;
struct snd_dec_flac flac_d;
+ struct snd_dec_wma wma_d;
} __attribute__((packed, aligned(4)));
/** struct snd_codec_desc - description of codec capabilities
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ALSA: compress: add wma codec profiles" to the asoc tree
@ 2020-03-16 18:08 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-16 18:08 UTC (permalink / raw)
To: Vinod Koul; +Cc: alsa-devel, Mark Brown
The patch
ALSA: compress: add wma codec profiles
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 3bd7ac41d85527126c31f336a75cb2275a6a8898 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vkoul@kernel.org>
Date: Mon, 16 Mar 2020 11:22:13 +0530
Subject: [PATCH] ALSA: compress: add wma codec profiles
Some codec profiles were missing for WMA, like WMA9/10 lossless and
wma10 pro, so add these profiles
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200316055221.1944464-2-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/uapi/sound/compress_params.h | 3 +++
1 file changed, 3 insertions(+)
diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index 9c96fb0e4d90..a47d9df0fd7b 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -142,6 +142,9 @@
#define SND_AUDIOPROFILE_WMA8 ((__u32) 0x00000002)
#define SND_AUDIOPROFILE_WMA9 ((__u32) 0x00000004)
#define SND_AUDIOPROFILE_WMA10 ((__u32) 0x00000008)
+#define SND_AUDIOPROFILE_WMA9_PRO ((__u32) 0x00000010)
+#define SND_AUDIOPROFILE_WMA9_LOSSLESS ((__u32) 0x00000020)
+#define SND_AUDIOPROFILE_WMA10_LOSSLESS ((__u32) 0x00000040)
#define SND_AUDIOMODE_WMA_LEVEL1 ((__u32) 0x00000001)
#define SND_AUDIOMODE_WMA_LEVEL2 ((__u32) 0x00000002)
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: codecs: wsa881x: remove soundwire stream handling" to the asoc tree
@ 2020-03-18 21:41 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-18 21:41 UTC (permalink / raw)
To: Srinivas Kandagatla; +Cc: alsa-devel, Mark Brown
The patch
ASoC: codecs: wsa881x: remove soundwire stream handling
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 16252a8f3af77f69c2193fdc7b2f595b30845a44 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Tue, 17 Mar 2020 15:12:33 +0000
Subject: [PATCH] ASoC: codecs: wsa881x: remove soundwire stream handling
There could be multiple instances of this codec on any platform,
so handling stream directly in this codec driver can lead to
multiple calls to prepare/enable/disable on the same SoundWire stream.
Move this stream handling to machine driver to fix this issue.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200317151233.8763-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wsa881x.c | 44 +-------------------------------------
1 file changed, 1 insertion(+), 43 deletions(-)
diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c
index b59f1d0e7f84..3d525297eac9 100644
--- a/sound/soc/codecs/wsa881x.c
+++ b/sound/soc/codecs/wsa881x.c
@@ -676,7 +676,6 @@ struct wsa881x_priv {
int active_ports;
bool port_prepared[WSA881X_MAX_SWR_PORTS];
bool port_enable[WSA881X_MAX_SWR_PORTS];
- bool stream_prepared;
};
static void wsa881x_init(struct wsa881x_priv *wsa881x)
@@ -954,41 +953,6 @@ static const struct snd_soc_dapm_widget wsa881x_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("SPKR"),
};
-static int wsa881x_prepare(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev);
- int ret;
-
- if (wsa881x->stream_prepared) {
- sdw_disable_stream(wsa881x->sruntime);
- sdw_deprepare_stream(wsa881x->sruntime);
- wsa881x->stream_prepared = false;
- }
-
-
- ret = sdw_prepare_stream(wsa881x->sruntime);
- if (ret)
- return ret;
-
- /**
- * NOTE: there is a strict hw requirement about the ordering of port
- * enables and actual PA enable. PA enable should only happen after
- * soundwire ports are enabled if not DC on the line is accumulated
- * resulting in Click/Pop Noise
- * PA enable/mute are handled as part of DAPM and digital mute.
- */
-
- ret = sdw_enable_stream(wsa881x->sruntime);
- if (ret) {
- sdw_deprepare_stream(wsa881x->sruntime);
- return ret;
- }
- wsa881x->stream_prepared = true;
-
- return ret;
-}
-
static int wsa881x_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -1016,12 +980,7 @@ static int wsa881x_hw_free(struct snd_pcm_substream *substream,
{
struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev);
- if (wsa881x->stream_prepared) {
- sdw_disable_stream(wsa881x->sruntime);
- sdw_deprepare_stream(wsa881x->sruntime);
- sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime);
- wsa881x->stream_prepared = false;
- }
+ sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime);
return 0;
}
@@ -1052,7 +1011,6 @@ static int wsa881x_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
static struct snd_soc_dai_ops wsa881x_dai_ops = {
.hw_params = wsa881x_hw_params,
- .prepare = wsa881x_prepare,
.hw_free = wsa881x_hw_free,
.mute_stream = wsa881x_digital_mute,
.set_sdw_stream = wsa881x_set_sdw_stream,
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-core: Merge CPU/Codec for soc_dai_pcm_new()" to the asoc tree
@ 2020-03-20 20:45 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-20 20:45 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-core: Merge CPU/Codec for soc_dai_pcm_new()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From d1eb6d116123b2bcebeefce8bcdc828c80b033b8 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 16 Mar 2020 15:37:20 +0900
Subject: [PATCH] ASoC: soc-core: Merge CPU/Codec for soc_dai_pcm_new()
Now CPU/Codec DAIs are alias for dais.
Thus, we can directly use for_each_rtd_dais() macro
for soc_dai_pcm_new().
This patch merge CPU/Codec for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87r1xsolen.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-core.c | 23 +++++++++--------------
1 file changed, 9 insertions(+), 14 deletions(-)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 246d59966795..843b8b1c89d4 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1037,20 +1037,20 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card,
}
EXPORT_SYMBOL_GPL(snd_soc_add_pcm_runtime);
-static int soc_dai_pcm_new(struct snd_soc_dai **dais, int num_dais,
- struct snd_soc_pcm_runtime *rtd)
+static int soc_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_soc_dai *dai;
int i, ret = 0;
- for (i = 0; i < num_dais; ++i) {
- struct snd_soc_dai_driver *drv = dais[i]->driver;
+ for_each_rtd_dais(rtd, i, dai) {
+ struct snd_soc_dai_driver *drv = dai->driver;
if (drv->pcm_new)
- ret = drv->pcm_new(rtd, dais[i]);
+ ret = drv->pcm_new(rtd, dai);
if (ret < 0) {
- dev_err(dais[i]->dev,
+ dev_err(dai->dev,
"ASoC: Failed to bind %s with pcm device\n",
- dais[i]->name);
+ dai->name);
return ret;
}
}
@@ -1121,13 +1121,8 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card,
dai_link->stream_name, ret);
return ret;
}
- ret = soc_dai_pcm_new(rtd->cpu_dais,
- rtd->num_cpus, rtd);
- if (ret < 0)
- return ret;
- ret = soc_dai_pcm_new(rtd->codec_dais,
- rtd->num_codecs, rtd);
- return ret;
+
+ return soc_dai_pcm_new(rtd);
}
static void soc_set_name_prefix(struct snd_soc_card *card,
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-dapm: Merge for_each_rtd_cpu/codec_dais()" to the asoc tree
@ 2020-03-20 20:45 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-20 20:45 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-dapm: Merge for_each_rtd_cpu/codec_dais()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From e3c3cf71013fd959abf455abc20386051d37c529 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 16 Mar 2020 15:37:09 +0900
Subject: [PATCH] ASoC: soc-dapm: Merge for_each_rtd_cpu/codec_dais()
Now we can use for_each_rtd_dais().
Let's use it instead of for_each_rtd_cpu/codec_dais().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87tv2ooley.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-dapm.c | 9 +++------
1 file changed, 3 insertions(+), 6 deletions(-)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d5eb52fe115b..04da7928c873 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -4433,14 +4433,11 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
int event)
{
- struct snd_soc_dai *codec_dai;
- struct snd_soc_dai *cpu_dai;
+ struct snd_soc_dai *dai;
int i;
- for_each_rtd_cpu_dais(rtd, i, cpu_dai)
- soc_dapm_dai_stream_event(cpu_dai, stream, event);
- for_each_rtd_codec_dais(rtd, i, codec_dai)
- soc_dapm_dai_stream_event(codec_dai, stream, event);
+ for_each_rtd_dais(rtd, i, dai)
+ soc_dapm_dai_stream_event(dai, stream, event);
dapm_power_widgets(rtd->card, event);
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-core: Merge for_each_rtd_cpu/codec_dais()" to the asoc tree
@ 2020-03-20 20:45 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-20 20:45 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-core: Merge for_each_rtd_cpu/codec_dais()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 3af6ff5035ad95ee02728aff271a9b543b912f15 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 16 Mar 2020 15:37:03 +0900
Subject: [PATCH] ASoC: soc-core: Merge for_each_rtd_cpu/codec_dais()
Now we can use for_each_rtd_dais().
Let's use it instead of for_each_rtd_cpu/codec_dais().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87v9n4olf4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-core.c | 25 +++++++------------------
1 file changed, 7 insertions(+), 18 deletions(-)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 0fd582c19c03..246d59966795 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1323,26 +1323,22 @@ static int soc_probe_dai(struct snd_soc_dai *dai, int order)
static void soc_remove_link_dais(struct snd_soc_card *card)
{
int i;
- struct snd_soc_dai *codec_dai;
- struct snd_soc_dai *cpu_dai;
+ struct snd_soc_dai *dai;
struct snd_soc_pcm_runtime *rtd;
int order;
for_each_comp_order(order) {
for_each_card_rtds(card, rtd) {
- /* remove the CODEC DAI */
- for_each_rtd_codec_dais(rtd, i, codec_dai)
- soc_remove_dai(codec_dai, order);
-
- for_each_rtd_cpu_dais(rtd, i, cpu_dai)
- soc_remove_dai(cpu_dai, order);
+ /* remove DAIs */
+ for_each_rtd_dais(rtd, i, dai)
+ soc_remove_dai(dai, order);
}
}
}
static int soc_probe_link_dais(struct snd_soc_card *card)
{
- struct snd_soc_dai *codec_dai, *cpu_dai;
+ struct snd_soc_dai *dai;
struct snd_soc_pcm_runtime *rtd;
int i, order, ret;
@@ -1354,15 +1350,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card)
card->name, rtd->num, order);
/* probe the CPU DAI */
- for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
- ret = soc_probe_dai(cpu_dai, order);
- if (ret)
- return ret;
- }
-
- /* probe the CODEC DAI */
- for_each_rtd_codec_dais(rtd, i, codec_dai) {
- ret = soc_probe_dai(codec_dai, order);
+ for_each_rtd_dais(rtd, i, dai) {
+ ret = soc_probe_dai(dai, order);
if (ret)
return ret;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-core: Merge CPU/Codec DAIs" to the asoc tree
@ 2020-03-20 20:45 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-20 20:45 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown
The patch
ASoC: soc-core: Merge CPU/Codec DAIs
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 22a2fc81658b3eebcfcc110de97bcbd32f5ee301 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 16 Mar 2020 15:36:58 +0900
Subject: [PATCH] ASoC: soc-core: Merge CPU/Codec DAIs
ALSA SoC is currently categorizing CPU/Codec DAIs,
and it works well.
But modern devices require more complex connections,
for example Codec to Codec, etc, and future devices will
enable to more complex connections.
Because of these background, CPU/Codec DAIs categorizing is
no longer good much to modern device.
Currently, rtd has both CPU/Codec DAIs pointer.
rtd->cpu_dais = [][][][][][][][][]
rtd->codec_dais = [][][][][][][][][]
This patch merges these into DAIs pointer.
rtd->dais = [][][][][][][][][][][][][][][][][][]
^cpu_dais ^codec_dais
|--- num_cpus ---|--- num_codecs --|
Then, we can merge for_each_rtd_cpu/codec_dais() from this patch.
- for_each_rtd_cpu_dais() {
- ...
- }
- for_each_rtd_codec_dais() {
- ...
- }
+ for_each_rtd_dais() {
+ ...
+ }
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87wo7kolfa.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc.h | 7 ++++++-
sound/soc/soc-core.c | 18 +++++++++---------
2 files changed, 15 insertions(+), 10 deletions(-)
diff --git a/include/sound/soc.h b/include/sound/soc.h
index d97c4aa779a2..539211bd0f94 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1145,6 +1145,7 @@ struct snd_soc_pcm_runtime {
struct snd_compr *compr;
struct snd_soc_dai *codec_dai;
struct snd_soc_dai *cpu_dai;
+ struct snd_soc_dai **dais;
struct snd_soc_dai **codec_dais;
unsigned int num_codecs;
@@ -1184,7 +1185,11 @@ struct snd_soc_pcm_runtime {
(i)++)
#define for_each_rtd_codec_dais_rollback(rtd, i, dai) \
for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);)
-
+#define for_each_rtd_dais(rtd, i, dai) \
+ for ((i) = 0; \
+ ((i) < (rtd)->num_cpus + (rtd)->num_codecs) && \
+ ((dai) = (rtd)->dais[i]); \
+ (i)++)
void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index e7e70b47590a..0fd582c19c03 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -475,22 +475,22 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
/*
- * for rtd->codec_dais
+ * for rtd->dais
*/
- rtd->codec_dais = devm_kcalloc(dev, dai_link->num_codecs,
+ rtd->dais = devm_kcalloc(dev, dai_link->num_cpus + dai_link->num_codecs,
sizeof(struct snd_soc_dai *),
GFP_KERNEL);
- if (!rtd->codec_dais)
+ if (!rtd->dais)
goto free_rtd;
/*
- * for rtd->cpu_dais
+ * dais = [][][][][][][][][][][][][][][][][][]
+ * ^cpu_dais ^codec_dais
+ * |--- num_cpus ---|--- num_codecs --|
*/
- rtd->cpu_dais = devm_kcalloc(dev, dai_link->num_cpus,
- sizeof(struct snd_soc_dai *),
- GFP_KERNEL);
- if (!rtd->cpu_dais)
- goto free_rtd;
+ rtd->cpu_dais = &rtd->dais[0];
+ rtd->codec_dais = &rtd->dais[dai_link->num_cpus];
+
/*
* rtd remaining settings
*/
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "dt-bindings: sound: rockchip-i2s: add #sound-dai-cells property" to the asoc tree
@ 2020-03-24 17:16 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-24 17:16 UTC (permalink / raw)
To: Johan Jonker; +Cc: alsa-devel, Mark Brown
The patch
dt-bindings: sound: rockchip-i2s: add #sound-dai-cells property
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 515d2757d31e1c9587f166dba5e828fb8569df84 Mon Sep 17 00:00:00 2001
From: Johan Jonker <jbx6244@gmail.com>
Date: Tue, 24 Mar 2020 10:41:48 +0100
Subject: [PATCH] dt-bindings: sound: rockchip-i2s: add #sound-dai-cells
property
'#sound-dai-cells' is required to properly interpret
the list of DAI specified in the 'sound-dai' property,
so add them to 'rockchip-i2s.yaml'
Signed-off-by: Johan Jonker <jbx6244@gmail.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200324094149.6904-2-jbx6244@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/rockchip-i2s.yaml | 5 +++++
1 file changed, 5 insertions(+)
diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml
index eff06b4b51db..7cd0e278ed85 100644
--- a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml
+++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml
@@ -77,6 +77,9 @@ properties:
Required property for controllers which support multi channel
playback/capture.
+ "#sound-dai-cells":
+ const: 0
+
required:
- compatible
- reg
@@ -85,6 +88,7 @@ required:
- clock-names
- dmas
- dma-names
+ - "#sound-dai-cells"
additionalProperties: false
@@ -103,4 +107,5 @@ examples:
dma-names = "tx", "rx";
rockchip,capture-channels = <2>;
rockchip,playback-channels = <8>;
+ #sound-dai-cells = <0>;
};
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "dt-bindings: sound: convert rockchip i2s bindings to yaml" to the asoc tree
@ 2020-03-24 17:16 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-24 17:16 UTC (permalink / raw)
To: Johan Jonker; +Cc: alsa-devel, Mark Brown
The patch
dt-bindings: sound: convert rockchip i2s bindings to yaml
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 472abb80fac66b07b6940a40c6392d3dea5dd2a3 Mon Sep 17 00:00:00 2001
From: Johan Jonker <jbx6244@gmail.com>
Date: Tue, 24 Mar 2020 10:41:47 +0100
Subject: [PATCH] dt-bindings: sound: convert rockchip i2s bindings to yaml
Current dts files with 'i2s' nodes are manually verified.
In order to automate this process rockchip-i2s.txt
has to be converted to yaml.
Signed-off-by: Johan Jonker <jbx6244@gmail.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200324094149.6904-1-jbx6244@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
.../bindings/sound/rockchip-i2s.txt | 49 --------
.../bindings/sound/rockchip-i2s.yaml | 106 ++++++++++++++++++
2 files changed, 106 insertions(+), 49 deletions(-)
delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.txt
create mode 100644 Documentation/devicetree/bindings/sound/rockchip-i2s.yaml
diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
deleted file mode 100644
index 54aefab71f2c..000000000000
--- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt
+++ /dev/null
@@ -1,49 +0,0 @@
-* Rockchip I2S controller
-
-The I2S bus (Inter-IC sound bus) is a serial link for digital
-audio data transfer between devices in the system.
-
-Required properties:
-
-- compatible: should be one of the following:
- - "rockchip,rk3066-i2s": for rk3066
- - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30
- - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036
- - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188
- - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228
- - "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288
- - "rockchip,rk3328-i2s", "rockchip,rk3066-i2s": for rk3328
- - "rockchip,rk3366-i2s", "rockchip,rk3066-i2s": for rk3366
- - "rockchip,rk3368-i2s", "rockchip,rk3066-i2s": for rk3368
- - "rockchip,rk3399-i2s", "rockchip,rk3066-i2s": for rk3399
-- reg: physical base address of the controller and length of memory mapped
- region.
-- interrupts: should contain the I2S interrupt.
-- dmas: DMA specifiers for tx and rx dma. See the DMA client binding,
- Documentation/devicetree/bindings/dma/dma.txt
-- dma-names: should include "tx" and "rx".
-- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
-- clock-names: should contain the following:
- - "i2s_hclk": clock for I2S BUS
- - "i2s_clk" : clock for I2S controller
-- rockchip,playback-channels: max playback channels, if not set, 8 channels default.
-- rockchip,capture-channels: max capture channels, if not set, 2 channels default.
-
-Required properties for controller which support multi channels
-playback/capture:
-
-- rockchip,grf: the phandle of the syscon node for GRF register.
-
-Example for rk3288 I2S controller:
-
-i2s@ff890000 {
- compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s";
- reg = <0xff890000 0x10000>;
- interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>;
- dmas = <&pdma1 0>, <&pdma1 1>;
- dma-names = "tx", "rx";
- clock-names = "i2s_hclk", "i2s_clk";
- clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>;
- rockchip,playback-channels = <8>;
- rockchip,capture-channels = <2>;
-};
diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml
new file mode 100644
index 000000000000..eff06b4b51db
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml
@@ -0,0 +1,106 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/rockchip-i2s.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Rockchip I2S controller
+
+description:
+ The I2S bus (Inter-IC sound bus) is a serial link for digital
+ audio data transfer between devices in the system.
+
+maintainers:
+ - Heiko Stuebner <heiko@sntech.de>
+
+properties:
+ compatible:
+ oneOf:
+ - const: rockchip,rk3066-i2s
+ - items:
+ - enum:
+ - rockchip,px30-i2s
+ - rockchip,rk3036-i2s
+ - rockchip,rk3188-i2s
+ - rockchip,rk3228-i2s
+ - rockchip,rk3288-i2s
+ - rockchip,rk3328-i2s
+ - rockchip,rk3366-i2s
+ - rockchip,rk3368-i2s
+ - rockchip,rk3399-i2s
+ - const: rockchip,rk3066-i2s
+
+ reg:
+ maxItems: 1
+
+ interrupts:
+ maxItems: 1
+
+ clocks:
+ items:
+ - description: clock for I2S controller
+ - description: clock for I2S BUS
+
+ clock-names:
+ items:
+ - const: i2s_clk
+ - const: i2s_hclk
+
+ dmas:
+ items:
+ - description: TX DMA Channel
+ - description: RX DMA Channel
+
+ dma-names:
+ items:
+ - const: tx
+ - const: rx
+
+ rockchip,capture-channels:
+ allOf:
+ - $ref: /schemas/types.yaml#/definitions/uint32
+ default: 2
+ description:
+ Max capture channels, if not set, 2 channels default.
+
+ rockchip,playback-channels:
+ allOf:
+ - $ref: /schemas/types.yaml#/definitions/uint32
+ default: 8
+ description:
+ Max playback channels, if not set, 8 channels default.
+
+ rockchip,grf:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description:
+ The phandle of the syscon node for the GRF register.
+ Required property for controllers which support multi channel
+ playback/capture.
+
+required:
+ - compatible
+ - reg
+ - interrupts
+ - clocks
+ - clock-names
+ - dmas
+ - dma-names
+
+additionalProperties: false
+
+examples:
+ - |
+ #include <dt-bindings/clock/rk3288-cru.h>
+ #include <dt-bindings/interrupt-controller/arm-gic.h>
+ #include <dt-bindings/interrupt-controller/irq.h>
+ i2s@ff890000 {
+ compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s";
+ reg = <0xff890000 0x10000>;
+ interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>;
+ clocks = <&cru SCLK_I2S0>, <&cru HCLK_I2S0>;
+ clock-names = "i2s_clk", "i2s_hclk";
+ dmas = <&pdma1 0>, <&pdma1 1>;
+ dma-names = "tx", "rx";
+ rockchip,capture-channels = <2>;
+ rockchip,playback-channels = <8>;
+ };
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: Intel: broadwell: Revert back SSP0 link to use dummy components" to the asoc tree
@ 2020-03-25 21:33 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-25 21:33 UTC (permalink / raw)
To: Cezary Rojewski
Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart, Dominik Brodowski
The patch
ASoC: Intel: broadwell: Revert back SSP0 link to use dummy components
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 68999d939dcfb430dd9834df56c71e43b7f37c6e Mon Sep 17 00:00:00 2001
From: Cezary Rojewski <cezary.rojewski@intel.com>
Date: Wed, 25 Mar 2020 14:16:09 +0100
Subject: [PATCH] ASoC: Intel: broadwell: Revert back SSP0 link to use dummy
components
Recent series of patches targeting broadwell boards, while enabling
SOF, changed behavior for non-SOF solutions. In essence replacing
platform 'dummy' with actual 'platform' causes redundant stream
initialization to occur during audio start. hw_params for haswell-pcm
destroys initial stream right after its creation - only to recreate it
again from proceed from there.
While harmless so far, this flow isn't correct and should be corrected.
The actual need for dummy components for SSP0 link is questionable but
that issue is subject for another series.
Link to first message in conversation:
https://lkml.org/lkml/2020/3/18/54
Fixes: 64df6afa0dab ("ASoC: Intel: broadwell: change cpu_dai and platform components for SOF")
Reported-by: Dominik Brodowski <linux@dominikbrodowski.net>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325131611.545-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/intel/boards/broadwell.c | 7 ++++---
1 file changed, 4 insertions(+), 3 deletions(-)
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index b9c12e24c70b..ffbb597052aa 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -167,9 +167,6 @@ SND_SOC_DAILINK_DEF(codec,
#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
SND_SOC_DAILINK_DEF(ssp0_port,
DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
-#else
-SND_SOC_DAILINK_DEF(ssp0_port,
- DAILINK_COMP_ARRAY(COMP_DUMMY()));
#endif
/* broadwell digital audio interface glue - connects codec <--> CPU */
@@ -226,7 +223,11 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
.ops = &broadwell_rt286_ops,
.dpcm_playback = 1,
.dpcm_capture = 1,
+#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
+ SND_SOC_DAILINK_REG(dummy, codec, dummy),
+#else
SND_SOC_DAILINK_REG(ssp0_port, codec, platform),
+#endif
},
};
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: fix uninitialised "work" with VirtIO" to the asoc tree
@ 2020-03-26 19:05 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-26 19:05 UTC (permalink / raw)
To: Guennadi Liakhovetski; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: SOF: fix uninitialised "work" with VirtIO
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 9ef91cad92ba75d17d5a5203230746c9d9009705 Mon Sep 17 00:00:00 2001
From: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Date: Wed, 25 Mar 2020 16:12:31 -0500
Subject: [PATCH] ASoC: SOF: fix uninitialised "work" with VirtIO
In the VirtIO case the sof_pcm_open() function isn't called on the
host during guest streaming, which then leaves "work" structures
uninitialised. However it is then used to handle position update
messages from the DSP. Move their initialisation to immediately after
allocation of the containing structure.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200325211233.27394-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/pcm.c | 4 +---
sound/soc/sof/sof-audio.h | 3 +++
sound/soc/sof/topology.c | 6 +++++-
3 files changed, 9 insertions(+), 4 deletions(-)
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index f4769e19965a..47cd741f2a8c 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -57,7 +57,7 @@ static int sof_pcm_dsp_params(struct snd_sof_pcm *spcm, struct snd_pcm_substream
/*
* sof pcm period elapse work
*/
-static void sof_pcm_period_elapsed_work(struct work_struct *work)
+void snd_sof_pcm_period_elapsed_work(struct work_struct *work)
{
struct snd_sof_pcm_stream *sps =
container_of(work, struct snd_sof_pcm_stream,
@@ -475,8 +475,6 @@ static int sof_pcm_open(struct snd_soc_component *component,
dev_dbg(component->dev, "pcm: open stream %d dir %d\n",
spcm->pcm.pcm_id, substream->stream);
- INIT_WORK(&spcm->stream[substream->stream].period_elapsed_work,
- sof_pcm_period_elapsed_work);
caps = &spcm->pcm.caps[substream->stream];
diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h
index eacd10e4da11..bf65f31af858 100644
--- a/sound/soc/sof/sof-audio.h
+++ b/sound/soc/sof/sof-audio.h
@@ -11,6 +11,8 @@
#ifndef __SOUND_SOC_SOF_AUDIO_H
#define __SOUND_SOC_SOF_AUDIO_H
+#include <linux/workqueue.h>
+
#include <sound/soc.h>
#include <sound/control.h>
#include <sound/sof/stream.h> /* needs to be included before control.h */
@@ -189,6 +191,7 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp,
struct snd_sof_pcm *snd_sof_find_spcm_pcm_id(struct snd_soc_component *scomp,
unsigned int pcm_id);
void snd_sof_pcm_period_elapsed(struct snd_pcm_substream *substream);
+void snd_sof_pcm_period_elapsed_work(struct work_struct *work);
/*
* Mixer IPC
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index 54437caf9488..fe8ba3e05e08 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -9,6 +9,7 @@
//
#include <linux/firmware.h>
+#include <linux/workqueue.h>
#include <sound/tlv.h>
#include <sound/pcm_params.h>
#include <uapi/sound/sof/tokens.h>
@@ -2461,8 +2462,11 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index,
spcm->scomp = scomp;
- for_each_pcm_streams(stream)
+ for_each_pcm_streams(stream) {
spcm->stream[stream].comp_id = COMP_ID_UNASSIGNED;
+ INIT_WORK(&spcm->stream[stream].period_elapsed_work,
+ snd_sof_pcm_period_elapsed_work);
+ }
spcm->pcm = *pcm;
dev_dbg(scomp->dev, "tplg: load pcm %s\n", pcm->dai_name);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: (cosmetic) use for_each_pcm_streams() in sof_dai_load()" to the asoc tree
@ 2020-03-26 19:05 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-26 19:05 UTC (permalink / raw)
To: Guennadi Liakhovetski; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: SOF: (cosmetic) use for_each_pcm_streams() in sof_dai_load()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From c688cf1d3a2cc1bca5737e7849325b3ac8e69a41 Mon Sep 17 00:00:00 2001
From: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Date: Wed, 25 Mar 2020 16:12:30 -0500
Subject: [PATCH] ASoC: SOF: (cosmetic) use for_each_pcm_streams() in
sof_dai_load()
Use for_each_pcm_streams() to enumerate streams in sof_dai_load()
instead of doing that manually.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200325211233.27394-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/topology.c | 13 ++++++++-----
1 file changed, 8 insertions(+), 5 deletions(-)
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index 058de94fb8cf..54437caf9488 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -2448,7 +2448,7 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index,
struct snd_soc_tplg_stream_caps *caps;
struct snd_soc_tplg_private *private = &pcm->priv;
struct snd_sof_pcm *spcm;
- int stream = SNDRV_PCM_STREAM_PLAYBACK;
+ int stream;
int ret = 0;
/* nothing to do for BEs atm */
@@ -2460,8 +2460,9 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index,
return -ENOMEM;
spcm->scomp = scomp;
- spcm->stream[SNDRV_PCM_STREAM_PLAYBACK].comp_id = COMP_ID_UNASSIGNED;
- spcm->stream[SNDRV_PCM_STREAM_CAPTURE].comp_id = COMP_ID_UNASSIGNED;
+
+ for_each_pcm_streams(stream)
+ spcm->stream[stream].comp_id = COMP_ID_UNASSIGNED;
spcm->pcm = *pcm;
dev_dbg(scomp->dev, "tplg: load pcm %s\n", pcm->dai_name);
@@ -2482,8 +2483,10 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index,
if (!spcm->pcm.playback)
goto capture;
+ stream = SNDRV_PCM_STREAM_PLAYBACK;
+
dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: playback d0i3:%d\n",
- spcm->pcm.pcm_name, spcm->stream[0].d0i3_compatible);
+ spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible);
caps = &spcm->pcm.caps[stream];
@@ -2513,7 +2516,7 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index,
return ret;
dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: capture d0i3:%d\n",
- spcm->pcm.pcm_name, spcm->stream[1].d0i3_compatible);
+ spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible);
caps = &spcm->pcm.caps[stream];
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread" to the asoc tree
@ 2020-03-27 15:33 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-27 15:33 UTC (permalink / raw)
To: Rander Wang; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 90de3281c86ae5378e951e84c76c4759390ff34d Mon Sep 17 00:00:00 2001
From: Rander Wang <rander.wang@intel.com>
Date: Wed, 25 Mar 2020 16:50:26 -0500
Subject: [PATCH] Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in
irq thread
If pci device is in D0, wakeen interrupt will be
aggregated at cAVS level as interrupt. This commit
check the wakeen status and process it in irq thread
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda.c | 16 ++++++++++++++++
sound/soc/sof/intel/hda.h | 6 ++++++
2 files changed, 22 insertions(+)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 7d1aa4c7d82c..211e91e79eae 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -241,6 +241,19 @@ static irqreturn_t hda_dsp_sdw_thread(int irq, void *context)
return sdw_intel_thread(irq, context);
}
+static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+
+ hdev = sdev->pdata->hw_pdata;
+ if (hdev->sdw &&
+ snd_sof_dsp_read(sdev, HDA_DSP_BAR,
+ HDA_DSP_REG_SNDW_WAKE_STS))
+ return true;
+
+ return false;
+}
+
void hda_sdw_process_wakeen(struct snd_sof_dev *sdev)
{
struct sof_intel_hda_dev *hdev;
@@ -684,6 +697,9 @@ static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context)
if (hda_dsp_check_sdw_irq(sdev))
hda_dsp_sdw_thread(irq, hdev->sdw);
+ if (hda_sdw_check_wakeen_irq(sdev))
+ hda_sdw_process_wakeen(sdev);
+
/* enable GIE interrupt */
snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR,
SOF_HDA_INTCTL,
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 6f1765b1ed1d..e9825798de77 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -233,6 +233,7 @@
#define HDA_DSP_REG_ADSPIS2 (HDA_DSP_GEN_BASE + 0x14)
#define HDA_DSP_REG_ADSPIS2_SNDW BIT(5)
+#define HDA_DSP_REG_SNDW_WAKE_STS 0x2C192
/* Intel HD Audio Inter-Processor Communication Registers */
#define HDA_DSP_IPC_BASE 0x40
@@ -709,6 +710,11 @@ static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context)
return IRQ_HANDLED;
}
+static inline bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev)
+{
+ return false;
+}
+
static inline void hda_sdw_process_wakeen(struct snd_sof_dev *sdev)
{
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire" to the asoc tree
@ 2020-03-27 15:33 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-27 15:33 UTC (permalink / raw)
To: Rander Wang; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From bbd19cdca8279cf244a301c6a13ae5ec9e4ef976 Mon Sep 17 00:00:00 2001
From: Rander Wang <rander.wang@intel.com>
Date: Wed, 25 Mar 2020 16:50:25 -0500
Subject: [PATCH] ASoC: SOF: Intel: hda: add WAKEEN interrupt support for
SoundWire
When a SoundWire link is in clock stop state, a Slave device may wake
up the Master for some events such as jack detection. The WAKEEN
interrupt will be triggered and processed by the audio pci device.
If audio device is in D3, the interrupt will be routed to PME, or
aggregated at cAVS level as interrupt when audio device is in D0. This
patch only supports D3 case, where the audio pci device will be
resumed by a PME event and the WAKEEN interrupt will be processed
after audio pci device is powered up and ROM is initialized
successfully.
The WAKEEN handling is only enabled after the first boot due to
dependencies on a shim_lock mutex being initialized.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-loader.c | 18 ++++++++++++++++++
sound/soc/sof/intel/hda.c | 11 +++++++++++
sound/soc/sof/intel/hda.h | 5 +++++
3 files changed, 34 insertions(+)
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index 2ae94ea53122..e1550ccd0a49 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -346,6 +346,24 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev)
goto cleanup;
}
+ /*
+ * When a SoundWire link is in clock stop state, a Slave
+ * device may trigger in-band wakes for events such as jack
+ * insertion or acoustic event detection. This event will lead
+ * to a WAKEEN interrupt, handled by the PCI device and routed
+ * to PME if the PCI device is in D3. The resume function in
+ * audio PCI driver will be invoked by ACPI for PME event and
+ * initialize the device and process WAKEEN interrupt.
+ *
+ * The WAKEEN interrupt should be processed ASAP to prevent an
+ * interrupt flood, otherwise other interrupts, such IPC,
+ * cannot work normally. The WAKEEN is handled after the ROM
+ * is initialized successfully, which ensures power rails are
+ * enabled before accessing the SoundWire SHIM registers
+ */
+ if (!sdev->first_boot)
+ hda_sdw_process_wakeen(sdev);
+
/*
* at this point DSP ROM has been initialized and
* should be ready for code loading and firmware boot
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 1e69cfcee8e0..7d1aa4c7d82c 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -241,6 +241,17 @@ static irqreturn_t hda_dsp_sdw_thread(int irq, void *context)
return sdw_intel_thread(irq, context);
}
+void hda_sdw_process_wakeen(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+
+ hdev = sdev->pdata->hw_pdata;
+ if (!hdev->sdw)
+ return;
+
+ sdw_intel_process_wakeen_event(hdev->sdw);
+}
+
#endif
/*
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index fc104c5ba006..6f1765b1ed1d 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -671,6 +671,7 @@ int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd);
int hda_sdw_startup(struct snd_sof_dev *sdev);
void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable);
+void hda_sdw_process_wakeen(struct snd_sof_dev *sdev);
#else
@@ -707,6 +708,10 @@ static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context)
{
return IRQ_HANDLED;
}
+
+static inline void hda_sdw_process_wakeen(struct snd_sof_dev *sdev)
+{
+}
#endif
/* common dai driver */
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks" to the asoc tree
@ 2020-03-27 15:33 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-27 15:33 UTC (permalink / raw)
To: Pierre-Louis Bossart; +Cc: alsa-devel, Mark Brown
The patch
ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 02df8f4364b070428af0e5b6c8739c884c8ad4e7 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 25 Mar 2020 16:50:24 -0500
Subject: [PATCH] ASoC: SOF: Intel: hda: add parameter to control SoundWire
clock stop quirks
Add module parameter so that the different modes can be quickly tested.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda.c | 11 +++++++++++
1 file changed, 11 insertions(+)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index ee4f1ceca883..1e69cfcee8e0 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -40,6 +40,16 @@
#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)
+/*
+ * The default for SoundWire clock stop quirks is to power gate the IP
+ * and do a Bus Reset, this will need to be modified when the DSP
+ * needs to remain in D0i3 so that the Master does not lose context
+ * and enumeration is not required on clock restart
+ */
+static int sdw_clock_stop_quirks = SDW_INTEL_CLK_STOP_BUS_RESET;
+module_param(sdw_clock_stop_quirks, int, 0444);
+MODULE_PARM_DESC(sdw_clock_stop_quirks, "SOF SoundWire clock stop quirks");
+
static int sdw_params_stream(struct device *dev,
struct sdw_intel_stream_params_data *params_data)
{
@@ -149,6 +159,7 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev)
res.parent = sdev->dev;
res.ops = &sdw_callback;
res.dev = sdev->dev;
+ res.clock_stop_quirks = sdw_clock_stop_quirks;
/*
* ops and arg fields are not populated for now,
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers" to the asoc tree
@ 2020-03-27 15:33 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-27 15:33 UTC (permalink / raw)
To: Bard Liao; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 722ba5f1f530a919d28a0dd9e8e0ec63af18270d Mon Sep 17 00:00:00 2001
From: Bard Liao <yung-chuan.liao@linux.intel.com>
Date: Wed, 25 Mar 2020 16:50:23 -0500
Subject: [PATCH] ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire
interrupt handlers
We have a single irq handler for SOF interrupts. We can further merge
SoundWire ones to completely remove MSI interrupts handling issues
leading to timeouts.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda.c | 36 ++++++++++++++++++++++++++++++++++++
sound/soc/sof/intel/hda.h | 11 +++++++++++
2 files changed, 47 insertions(+)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index dea3c385b664..ee4f1ceca883 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -198,6 +198,38 @@ static int hda_sdw_exit(struct snd_sof_dev *sdev)
return 0;
}
+
+static bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+ bool ret = false;
+ u32 irq_status;
+
+ hdev = sdev->pdata->hw_pdata;
+
+ if (!hdev->sdw)
+ return ret;
+
+ /* store status */
+ irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS2);
+
+ /* invalid message ? */
+ if (irq_status == 0xffffffff)
+ goto out;
+
+ /* SDW message ? */
+ if (irq_status & HDA_DSP_REG_ADSPIS2_SNDW)
+ ret = true;
+
+out:
+ return ret;
+}
+
+static irqreturn_t hda_dsp_sdw_thread(int irq, void *context)
+{
+ return sdw_intel_thread(irq, context);
+}
+
#endif
/*
@@ -618,6 +650,7 @@ static irqreturn_t hda_dsp_interrupt_handler(int irq, void *context)
static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context)
{
struct snd_sof_dev *sdev = context;
+ struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata;
/* deal with streams and controller first */
if (hda_dsp_check_stream_irq(sdev))
@@ -626,6 +659,9 @@ static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context)
if (hda_dsp_check_ipc_irq(sdev))
sof_ops(sdev)->irq_thread(irq, sdev);
+ if (hda_dsp_check_sdw_irq(sdev))
+ hda_dsp_sdw_thread(irq, hdev->sdw);
+
/* enable GIE interrupt */
snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR,
SOF_HDA_INTCTL,
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 928a3432e9e6..fc104c5ba006 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -232,6 +232,8 @@
#define HDA_DSP_REG_ADSPIC2 (HDA_DSP_GEN_BASE + 0x10)
#define HDA_DSP_REG_ADSPIS2 (HDA_DSP_GEN_BASE + 0x14)
+#define HDA_DSP_REG_ADSPIS2_SNDW BIT(5)
+
/* Intel HD Audio Inter-Processor Communication Registers */
#define HDA_DSP_IPC_BASE 0x40
#define HDA_DSP_REG_HIPCT (HDA_DSP_IPC_BASE + 0x00)
@@ -696,6 +698,15 @@ static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable)
{
}
+static inline bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev)
+{
+ return false;
+}
+
+static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context)
+{
+ return IRQ_HANDLED;
+}
#endif
/* common dai driver */
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend" to the asoc tree
@ 2020-03-27 15:33 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-27 15:33 UTC (permalink / raw)
To: Pierre-Louis Bossart; +Cc: alsa-devel, Mark Brown
The patch
ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 3eadff5639b01c17f5f2ffeb209d05cc19706687 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 25 Mar 2020 16:50:22 -0500
Subject: [PATCH] ASoC: SOF: Intel: hda: disable SoundWire interrupts on
suspend
Doing this avoid conflicts and errors reported on the bus.
The interrupts are only re-enabled on resume after the firmware is
downloaded, so the behavior is not fully symmetric
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-dsp.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index c396b7ef0328..1aff90042694 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -594,6 +594,8 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend)
#endif
int ret;
+ hda_sdw_int_enable(sdev, false);
+
/* disable IPC interrupts */
hda_dsp_ipc_int_disable(sdev);
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks" to the asoc tree
@ 2020-03-27 15:34 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-27 15:34 UTC (permalink / raw)
To: Pierre-Louis Bossart; +Cc: alsa-devel, Mark Brown
The patch
ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From d2c383aa49dce9c5b59930c999099ecc580857fe Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 25 Mar 2020 16:50:20 -0500
Subject: [PATCH] ASoC: SOF: Intel: hda: add SoundWire stream config/free
callbacks
These callbacks are invoked when a matching hw_params/hw_free() DAI
operation takes place, and will result in IPC operations with the SOF
firmware.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda.c | 71 +++++++++++++++++++++++++++++++++++++++
1 file changed, 71 insertions(+)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index c1fe94800da1..1f93124a63cd 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -24,6 +24,7 @@
#include <sound/intel-nhlt.h>
#include <sound/sof.h>
#include <sound/sof/xtensa.h>
+#include "../sof-audio.h"
#include "../ops.h"
#include "hda.h"
@@ -38,6 +39,74 @@
#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)
+static int sdw_params_stream(struct device *dev,
+ struct sdw_intel_stream_params_data *params_data)
+{
+ struct snd_sof_dev *sdev = dev_get_drvdata(dev);
+ struct snd_soc_dai *d = params_data->dai;
+ struct sof_ipc_dai_config config;
+ struct sof_ipc_reply reply;
+ int link_id = params_data->link_id;
+ int alh_stream_id = params_data->alh_stream_id;
+ int ret;
+ u32 size = sizeof(config);
+
+ memset(&config, 0, size);
+ config.hdr.size = size;
+ config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG;
+ config.type = SOF_DAI_INTEL_ALH;
+ config.dai_index = (link_id << 8) | (d->id);
+ config.alh.stream_id = alh_stream_id;
+
+ /* send message to DSP */
+ ret = sof_ipc_tx_message(sdev->ipc,
+ config.hdr.cmd, &config, size, &reply,
+ sizeof(reply));
+ if (ret < 0) {
+ dev_err(sdev->dev,
+ "error: failed to set DAI hw_params for link %d dai->id %d ALH %d\n",
+ link_id, d->id, alh_stream_id);
+ }
+
+ return ret;
+}
+
+static int sdw_free_stream(struct device *dev,
+ struct sdw_intel_stream_free_data *free_data)
+{
+ struct snd_sof_dev *sdev = dev_get_drvdata(dev);
+ struct snd_soc_dai *d = free_data->dai;
+ struct sof_ipc_dai_config config;
+ struct sof_ipc_reply reply;
+ int link_id = free_data->link_id;
+ int ret;
+ u32 size = sizeof(config);
+
+ memset(&config, 0, size);
+ config.hdr.size = size;
+ config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG;
+ config.type = SOF_DAI_INTEL_ALH;
+ config.dai_index = (link_id << 8) | d->id;
+ config.alh.stream_id = 0xFFFF; /* invalid value on purpose */
+
+ /* send message to DSP */
+ ret = sof_ipc_tx_message(sdev->ipc,
+ config.hdr.cmd, &config, size, &reply,
+ sizeof(reply));
+ if (ret < 0) {
+ dev_err(sdev->dev,
+ "error: failed to free stream for link %d dai->id %d\n",
+ link_id, d->id);
+ }
+
+ return ret;
+}
+
+static const struct sdw_intel_ops sdw_callback = {
+ .params_stream = sdw_params_stream,
+ .free_stream = sdw_free_stream,
+};
+
void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable)
{
sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable);
@@ -80,6 +149,8 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev)
res.irq = sdev->ipc_irq;
res.handle = hdev->info.handle;
res.parent = sdev->dev;
+ res.ops = &sdw_callback;
+ res.dev = sdev->dev;
/*
* ops and arg fields are not populated for now,
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: IPC: dai-intel: move ALH declarations in header file" to the asoc tree
@ 2020-03-27 15:34 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-27 15:34 UTC (permalink / raw)
To: Pierre-Louis Bossart; +Cc: alsa-devel, Mark Brown
The patch
ASoC: SOF: IPC: dai-intel: move ALH declarations in header file
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From f8e25018801548314b3d908315cbf271e8eceba8 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 25 Mar 2020 16:50:19 -0500
Subject: [PATCH] ASoC: SOF: IPC: dai-intel: move ALH declarations in header
file
ALH was inserted in the wrong place during integration, add after DMIC
to mirror the file used by SOF firmware.
No functional change, just text move in the same file to better track
changes, if any.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/sof/dai-intel.h | 18 +++++++++---------
1 file changed, 9 insertions(+), 9 deletions(-)
diff --git a/include/sound/sof/dai-intel.h b/include/sound/sof/dai-intel.h
index 5f1ef5565be6..04e48227f542 100644
--- a/include/sound/sof/dai-intel.h
+++ b/include/sound/sof/dai-intel.h
@@ -87,6 +87,15 @@ struct sof_ipc_dai_hda_params {
uint32_t link_dma_ch;
} __packed;
+/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */
+struct sof_ipc_dai_alh_params {
+ struct sof_ipc_hdr hdr;
+ uint32_t stream_id;
+
+ /* reserved for future use */
+ uint32_t reserved[15];
+} __packed;
+
/* DMIC Configuration Request - SOF_IPC_DAI_DMIC_CONFIG */
/* This struct is defined per 2ch PDM controller available in the platform.
@@ -179,13 +188,4 @@ struct sof_ipc_dai_dmic_params {
struct sof_ipc_dai_dmic_pdm_ctrl pdm[0];
} __packed;
-/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */
-struct sof_ipc_dai_alh_params {
- struct sof_ipc_hdr hdr;
- uint32_t stream_id;
-
- /* reserved for future use */
- uint32_t reserved[15];
-} __packed;
-
#endif
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: SOF: Intel: add SoundWire configuration interface" to the asoc tree
@ 2020-03-27 15:34 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-03-27 15:34 UTC (permalink / raw)
To: Pierre-Louis Bossart; +Cc: alsa-devel, Mark Brown
The patch
ASoC: SOF: Intel: add SoundWire configuration interface
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 51dfed1e178a38202960b98f6e29df009a06050f Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 25 Mar 2020 16:50:18 -0500
Subject: [PATCH] ASoC: SOF: Intel: add SoundWire configuration interface
Now that the SoundWire core supports the multi-step initialization,
call the relevant APIs.
The actual hardware enablement can be done in two places, ideally we'd
want to startup the SoundWire IP as soon as possible (while still
taking power rail dependencies into account)
However when suspend/resume is implemented, the DSP device will be
resumed first, and only when the DSP firmware is downloaded/booted
would the SoundWire child devices be resumed, so there are only
marginal benefits in starting the IP earlier for the first probe.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200325215027.28716-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/sof/intel/hda-loader.c | 13 ++++
sound/soc/sof/intel/hda.c | 120 +++++++++++++++++++++++++++++++
sound/soc/sof/intel/hda.h | 44 ++++++++++++
3 files changed, 177 insertions(+)
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index 0633b76dab49..2ae94ea53122 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -399,6 +399,19 @@ int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev)
/* post fw run operations */
int hda_dsp_post_fw_run(struct snd_sof_dev *sdev)
{
+ int ret;
+
+ if (sdev->first_boot) {
+ ret = hda_sdw_startup(sdev);
+ if (ret < 0) {
+ dev_err(sdev->dev,
+ "error: could not startup SoundWire links\n");
+ return ret;
+ }
+ }
+
+ hda_sdw_int_enable(sdev, true);
+
/* re-enable clock gating and power gating */
return hda_dsp_ctrl_clock_power_gating(sdev, true);
}
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index ee1edb53840c..c1fe94800da1 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -18,7 +18,9 @@
#include <sound/hdaudio_ext.h>
#include <sound/hda_register.h>
+#include <linux/acpi.h>
#include <linux/module.h>
+#include <linux/soundwire/sdw_intel.h>
#include <sound/intel-nhlt.h>
#include <sound/sof.h>
#include <sound/sof/xtensa.h>
@@ -34,6 +36,98 @@
#define EXCEPT_MAX_HDR_SIZE 0x400
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)
+
+void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable)
+{
+ sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable);
+}
+
+static int hda_sdw_acpi_scan(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+ acpi_handle handle;
+ int ret;
+
+ handle = ACPI_HANDLE(sdev->dev);
+
+ /* save ACPI info for the probe step */
+ hdev = sdev->pdata->hw_pdata;
+
+ ret = sdw_intel_acpi_scan(handle, &hdev->info);
+ if (ret < 0) {
+ dev_err(sdev->dev, "%s failed\n", __func__);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int hda_sdw_probe(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+ struct sdw_intel_res res;
+ acpi_handle handle;
+ void *sdw;
+
+ handle = ACPI_HANDLE(sdev->dev);
+
+ hdev = sdev->pdata->hw_pdata;
+
+ memset(&res, 0, sizeof(res));
+
+ res.mmio_base = sdev->bar[HDA_DSP_BAR];
+ res.irq = sdev->ipc_irq;
+ res.handle = hdev->info.handle;
+ res.parent = sdev->dev;
+
+ /*
+ * ops and arg fields are not populated for now,
+ * they will be needed when the DAI callbacks are
+ * provided
+ */
+
+ /* we could filter links here if needed, e.g for quirks */
+ res.count = hdev->info.count;
+ res.link_mask = hdev->info.link_mask;
+
+ sdw = sdw_intel_probe(&res);
+ if (!sdw) {
+ dev_err(sdev->dev, "error: SoundWire probe failed\n");
+ return -EINVAL;
+ }
+
+ /* save context */
+ hdev->sdw = sdw;
+
+ return 0;
+}
+
+int hda_sdw_startup(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+
+ hdev = sdev->pdata->hw_pdata;
+
+ return sdw_intel_startup(hdev->sdw);
+}
+
+static int hda_sdw_exit(struct snd_sof_dev *sdev)
+{
+ struct sof_intel_hda_dev *hdev;
+
+ hdev = sdev->pdata->hw_pdata;
+
+ hda_sdw_int_enable(sdev, false);
+
+ if (hdev->sdw)
+ sdw_intel_exit(hdev->sdw);
+ hdev->sdw = NULL;
+
+ return 0;
+}
+#endif
+
/*
* Debug
*/
@@ -346,9 +440,12 @@ static const char *fixup_tplg_name(struct snd_sof_dev *sdev,
static int hda_init_caps(struct snd_sof_dev *sdev)
{
struct hdac_bus *bus = sof_to_bus(sdev);
+ struct snd_sof_pdata *pdata = sdev->pdata;
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
struct hdac_ext_link *hlink;
#endif
+ struct sof_intel_hda_dev *hdev = pdata->hw_pdata;
+ u32 link_mask;
int ret = 0;
device_disable_async_suspend(bus->dev);
@@ -365,6 +462,27 @@ static int hda_init_caps(struct snd_sof_dev *sdev)
return ret;
}
+ /* scan SoundWire capabilities exposed by DSDT */
+ ret = hda_sdw_acpi_scan(sdev);
+ if (ret < 0) {
+ dev_err(sdev->dev, "error: SoundWire ACPI scan error\n");
+ return ret;
+ }
+
+ link_mask = hdev->info.link_mask;
+ if (!link_mask) {
+ /*
+ * probe/allocated SoundWire resources.
+ * The hardware configuration takes place in hda_sdw_startup
+ * after power rails are enabled.
+ */
+ ret = hda_sdw_probe(sdev);
+ if (ret < 0) {
+ dev_err(sdev->dev, "error: SoundWire probe error\n");
+ return ret;
+ }
+ }
+
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
if (bus->mlcap)
snd_hdac_ext_bus_get_ml_capabilities(bus);
@@ -622,6 +740,8 @@ int hda_dsp_remove(struct snd_sof_dev *sdev)
snd_hdac_ext_bus_device_remove(bus);
#endif
+ hda_sdw_exit(sdev);
+
if (!IS_ERR_OR_NULL(hda->dmic_dev))
platform_device_unregister(hda->dmic_dev);
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 2a57fc9f3e58..928a3432e9e6 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -11,6 +11,8 @@
#ifndef __SOF_INTEL_HDA_H
#define __SOF_INTEL_HDA_H
+#include <linux/soundwire/sdw.h>
+#include <linux/soundwire/sdw_intel.h>
#include <sound/compress_driver.h>
#include <sound/hda_codec.h>
#include <sound/hdaudio_ext.h>
@@ -436,6 +438,12 @@ struct sof_intel_hda_dev {
/* delayed work to enter D0I3 opportunistically */
struct delayed_work d0i3_work;
+
+ /* ACPI information stored between scan and probe steps */
+ struct sdw_intel_acpi_info info;
+
+ /* sdw context allocated by SoundWire driver */
+ struct sdw_intel_ctx *sdw;
};
static inline struct hdac_bus *sof_to_bus(struct snd_sof_dev *s)
@@ -654,6 +662,42 @@ int hda_dsp_trace_init(struct snd_sof_dev *sdev, u32 *stream_tag);
int hda_dsp_trace_release(struct snd_sof_dev *sdev);
int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd);
+/*
+ * SoundWire support
+ */
+#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE)
+
+int hda_sdw_startup(struct snd_sof_dev *sdev);
+void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable);
+
+#else
+
+static inline int hda_sdw_acpi_scan(struct snd_sof_dev *sdev)
+{
+ return 0;
+}
+
+static inline int hda_sdw_probe(struct snd_sof_dev *sdev)
+{
+ return 0;
+}
+
+static inline int hda_sdw_startup(struct snd_sof_dev *sdev)
+{
+ return 0;
+}
+
+static inline int hda_sdw_exit(struct snd_sof_dev *sdev)
+{
+ return 0;
+}
+
+static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable)
+{
+}
+
+#endif
+
/* common dai driver */
extern struct snd_soc_dai_driver skl_dai[];
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: tas571x: disable regulators on failed probe" to the asoc tree
@ 2020-04-14 11:59 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-04-14 11:59 UTC (permalink / raw)
To: Philipp Puschmann; +Cc: alsa-devel, Mark Brown
The patch
ASoC: tas571x: disable regulators on failed probe
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 9df8ba7c63073508e5aa677dade48fcab6a6773e Mon Sep 17 00:00:00 2001
From: Philipp Puschmann <p.puschmann@pironex.de>
Date: Tue, 14 Apr 2020 13:27:54 +0200
Subject: [PATCH] ASoC: tas571x: disable regulators on failed probe
If probe fails after enabling the regulators regulator_put is called for
each supply without having them disabled before. This produces some
warnings like
WARNING: CPU: 0 PID: 90 at drivers/regulator/core.c:2044 _regulator_put.part.0+0x154/0x15c
[<c010f7a8>] (unwind_backtrace) from [<c010c544>] (show_stack+0x10/0x14)
[<c010c544>] (show_stack) from [<c012b640>] (__warn+0xd0/0xf4)
[<c012b640>] (__warn) from [<c012b9b4>] (warn_slowpath_fmt+0x64/0xc4)
[<c012b9b4>] (warn_slowpath_fmt) from [<c04c4064>] (_regulator_put.part.0+0x154/0x15c)
[<c04c4064>] (_regulator_put.part.0) from [<c04c4094>] (regulator_put+0x28/0x38)
[<c04c4094>] (regulator_put) from [<c04c40cc>] (regulator_bulk_free+0x28/0x38)
[<c04c40cc>] (regulator_bulk_free) from [<c0579b2c>] (release_nodes+0x1d0/0x22c)
[<c0579b2c>] (release_nodes) from [<c05756dc>] (really_probe+0x108/0x34c)
[<c05756dc>] (really_probe) from [<c0575aec>] (driver_probe_device+0xb8/0x16c)
[<c0575aec>] (driver_probe_device) from [<c0575d40>] (device_driver_attach+0x58/0x60)
[<c0575d40>] (device_driver_attach) from [<c0575da0>] (__driver_attach+0x58/0xcc)
[<c0575da0>] (__driver_attach) from [<c0573978>] (bus_for_each_dev+0x78/0xc0)
[<c0573978>] (bus_for_each_dev) from [<c0574b5c>] (bus_add_driver+0x188/0x1e0)
[<c0574b5c>] (bus_add_driver) from [<c05768b0>] (driver_register+0x74/0x108)
[<c05768b0>] (driver_register) from [<c061ab7c>] (i2c_register_driver+0x3c/0x88)
[<c061ab7c>] (i2c_register_driver) from [<c0102df8>] (do_one_initcall+0x58/0x250)
[<c0102df8>] (do_one_initcall) from [<c01a91bc>] (do_init_module+0x60/0x244)
[<c01a91bc>] (do_init_module) from [<c01ab5a4>] (load_module+0x2180/0x2540)
[<c01ab5a4>] (load_module) from [<c01abbd4>] (sys_finit_module+0xd0/0xe8)
[<c01abbd4>] (sys_finit_module) from [<c01011e0>] (__sys_trace_return+0x0/0x20)
Fixes: 3fd6e7d9a146 (ASoC: tas571x: New driver for TI TAS571x power amplifiers)
Signed-off-by: Philipp Puschmann <p.puschmann@pironex.de>
Link: https://lore.kernel.org/r/20200414112754.3365406-1-p.puschmann@pironex.de
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/tas571x.c | 20 +++++++++++++++-----
1 file changed, 15 insertions(+), 5 deletions(-)
diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c
index 1554631cb397..5b7f9fcf6cbf 100644
--- a/sound/soc/codecs/tas571x.c
+++ b/sound/soc/codecs/tas571x.c
@@ -820,8 +820,10 @@ static int tas571x_i2c_probe(struct i2c_client *client,
priv->regmap = devm_regmap_init(dev, NULL, client,
priv->chip->regmap_config);
- if (IS_ERR(priv->regmap))
- return PTR_ERR(priv->regmap);
+ if (IS_ERR(priv->regmap)) {
+ ret = PTR_ERR(priv->regmap);
+ goto disable_regs;
+ }
priv->pdn_gpio = devm_gpiod_get_optional(dev, "pdn", GPIOD_OUT_LOW);
if (IS_ERR(priv->pdn_gpio)) {
@@ -845,7 +847,7 @@ static int tas571x_i2c_probe(struct i2c_client *client,
ret = regmap_write(priv->regmap, TAS571X_OSC_TRIM_REG, 0);
if (ret)
- return ret;
+ goto disable_regs;
usleep_range(50000, 60000);
@@ -861,12 +863,20 @@ static int tas571x_i2c_probe(struct i2c_client *client,
*/
ret = regmap_update_bits(priv->regmap, TAS571X_MVOL_REG, 1, 0);
if (ret)
- return ret;
+ goto disable_regs;
}
- return devm_snd_soc_register_component(&client->dev,
+ ret = devm_snd_soc_register_component(&client->dev,
&priv->component_driver,
&tas571x_dai, 1);
+ if (ret)
+ goto disable_regs;
+
+ return ret;
+
+disable_regs:
+ regulator_bulk_disable(priv->chip->num_supply_names, priv->supplies);
+ return ret;
}
static int tas571x_i2c_remove(struct i2c_client *client)
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: wsa881x: mark read_only_wordlength flag" to the asoc tree
@ 2020-04-14 14:46 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-04-14 14:46 UTC (permalink / raw)
To: Srinivas Kandagatla; +Cc: alsa-devel, Mark Brown
The patch
ASoC: wsa881x: mark read_only_wordlength flag
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 0f2a3b02274c02eb97697c4d89c019d1d21ac225 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Tue, 14 Apr 2020 12:03:47 +0100
Subject: [PATCH] ASoC: wsa881x: mark read_only_wordlength flag
WSA881x works in PDM mode so the wordlength is fixed, which also makes
the only field "WordLength" in DPN_BlockCtrl1 register a read-only.
Writing to this register will throw up errors with Qualcomm Controller.
So use ro_blockctrl1_reg flag to mark this field as read-only so that
core will not write to this register.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200414110347.23829-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/codecs/wsa881x.c | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c
index f2d6f2f81f14..d39d479e2378 100644
--- a/sound/soc/codecs/wsa881x.c
+++ b/sound/soc/codecs/wsa881x.c
@@ -394,6 +394,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = {
.min_ch = 1,
.max_ch = 1,
.simple_ch_prep_sm = true,
+ .read_only_wordlength = true,
}, {
/* COMP */
.num = 2,
@@ -401,6 +402,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = {
.min_ch = 1,
.max_ch = 1,
.simple_ch_prep_sm = true,
+ .read_only_wordlength = true,
}, {
/* BOOST */
.num = 3,
@@ -408,6 +410,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = {
.min_ch = 1,
.max_ch = 1,
.simple_ch_prep_sm = true,
+ .read_only_wordlength = true,
}, {
/* VISENSE */
.num = 4,
@@ -415,6 +418,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = {
.min_ch = 1,
.max_ch = 1,
.simple_ch_prep_sm = true,
+ .read_only_wordlength = true,
}
};
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: rockchip-spdif: add power-domains property" to the asoc tree
@ 2020-04-14 14:46 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-04-14 14:46 UTC (permalink / raw)
To: Johan Jonker; +Cc: alsa-devel, Mark Brown
The patch
ASoC: rockchip-spdif: add power-domains property
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 65f1d3d9cee7e10d105756b6ef678885a23a31d7 Mon Sep 17 00:00:00 2001
From: Johan Jonker <jbx6244@gmail.com>
Date: Sat, 4 Apr 2020 13:52:25 +0200
Subject: [PATCH] ASoC: rockchip-spdif: add power-domains property
In the old txt situation we add/describe only properties that are used
by the driver/hardware itself. With yaml it also filters things in a
node that are used by other drivers like 'power-domains' for rk3399,
so add it to 'rockchip-spdif.yaml'.
Signed-off-by: Johan Jonker <jbx6244@gmail.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200404115225.4314-3-jbx6244@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/rockchip-spdif.yaml | 3 +++
1 file changed, 3 insertions(+)
diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml
index f381dbbf5de6..c467152656f7 100644
--- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml
+++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml
@@ -51,6 +51,9 @@ properties:
dma-names:
const: tx
+ power-domains:
+ maxItems: 1
+
rockchip,grf:
$ref: /schemas/types.yaml#/definitions/phandle
description:
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: rockchip-spdif: add #sound-dai-cells property" to the asoc tree
@ 2020-04-14 14:46 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-04-14 14:46 UTC (permalink / raw)
To: Johan Jonker; +Cc: alsa-devel, Mark Brown
The patch
ASoC: rockchip-spdif: add #sound-dai-cells property
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From f0c0f5a4541cb5dee1f16cebcc4e11d952394c53 Mon Sep 17 00:00:00 2001
From: Johan Jonker <jbx6244@gmail.com>
Date: Sat, 4 Apr 2020 13:52:24 +0200
Subject: [PATCH] ASoC: rockchip-spdif: add #sound-dai-cells property
'#sound-dai-cells' is required to properly interpret
the list of DAI specified in the 'sound-dai' property,
so add them to 'rockchip-spdif.yaml'
Signed-off-by: Johan Jonker <jbx6244@gmail.com>
Reviewed-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20200404115225.4314-2-jbx6244@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
Documentation/devicetree/bindings/sound/rockchip-spdif.yaml | 5 +++++
1 file changed, 5 insertions(+)
diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml
index bfd44fd8ecbc..f381dbbf5de6 100644
--- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml
+++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml
@@ -57,6 +57,9 @@ properties:
The phandle of the syscon node for the GRF register.
Required property on RK3288.
+ "#sound-dai-cells":
+ const: 0
+
required:
- compatible
- reg
@@ -65,6 +68,7 @@ required:
- clock-names
- dmas
- dma-names
+ - "#sound-dai-cells"
if:
properties:
@@ -90,4 +94,5 @@ examples:
clock-names = "mclk", "hclk";
dmas = <&dmac1_s 8>;
dma-names = "tx";
+ #sound-dai-cells = <0>;
};
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais" to the asoc tree
@ 2020-04-14 14:48 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-04-14 14:48 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 1729025b04b9f242dca37f50dba8dd3705eb1ea1 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 30 Mar 2020 10:48:27 +0900
Subject: [PATCH] ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais
No-one is using cpu_dai/codec_dai/cpu_dais/codec_dais.
Let's remove these from snd_soc_pcm_runtime
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87eetabok4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc.h | 14 ++++++++------
sound/soc/soc-core.c | 11 +++--------
2 files changed, 11 insertions(+), 14 deletions(-)
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 4dff6745b067..a904086b8ef6 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1143,14 +1143,16 @@ struct snd_soc_pcm_runtime {
/* runtime devices */
struct snd_pcm *pcm;
struct snd_compr *compr;
- struct snd_soc_dai *codec_dai;
- struct snd_soc_dai *cpu_dai;
- struct snd_soc_dai **dais;
- struct snd_soc_dai **codec_dais;
+ /*
+ * dais = cpu_dai + codec_dai
+ * see
+ * soc_new_pcm_runtime()
+ * asoc_rtd_to_cpu()
+ * asoc_rtd_to_codec()
+ */
+ struct snd_soc_dai **dais;
unsigned int num_codecs;
-
- struct snd_soc_dai **cpu_dais;
unsigned int num_cpus;
struct delayed_work delayed_work;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4d617c495864..7ea9725e33b3 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -487,9 +487,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
* dais = [][][][][][][][][][][][][][][][][][]
* ^cpu_dais ^codec_dais
* |--- num_cpus ---|--- num_codecs --|
+ * see
+ * asoc_rtd_to_cpu()
+ * asoc_rtd_to_codec()
*/
- rtd->cpu_dais = &rtd->dais[0];
- rtd->codec_dais = &rtd->dais[dai_link->num_cpus];
rtd->num_cpus = dai_link->num_cpus;
rtd->num_codecs = dai_link->num_codecs;
rtd->card = card;
@@ -996,9 +997,6 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card,
snd_soc_rtd_add_component(rtd, asoc_rtd_to_cpu(rtd, i)->component);
}
- /* Single cpu links expect cpu and cpu_dai in runtime data */
- rtd->cpu_dai = rtd->cpu_dais[0];
-
/* Find CODEC from registered CODECs */
for_each_link_codecs(dai_link, i, codec) {
asoc_rtd_to_codec(rtd, i) = snd_soc_find_dai(codec);
@@ -1011,9 +1009,6 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card,
snd_soc_rtd_add_component(rtd, asoc_rtd_to_codec(rtd, i)->component);
}
- /* Single codec links expect codec and codec_dai in runtime data */
- rtd->codec_dai = rtd->codec_dais[0];
-
/* Find PLATFORM from registered PLATFORMs */
for_each_link_platforms(dai_link, i, platform) {
for_each_component(component) {
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups" to the asoc tree
@ 2020-04-14 14:48 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-04-14 14:48 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 01faf501eefa86fa6f334bb9c71a48498de653e6 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 30 Mar 2020 10:48:14 +0900
Subject: [PATCH] ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups
soc_new_pcm_runtime() setups rtd, but code is very random.
This patch tidyup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87ftdqbokh.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-core.c | 11 +++--------
1 file changed, 3 insertions(+), 8 deletions(-)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 8cef4fd6df36..4d617c495864 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -492,17 +492,12 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
rtd->codec_dais = &rtd->dais[dai_link->num_cpus];
rtd->num_cpus = dai_link->num_cpus;
rtd->num_codecs = dai_link->num_codecs;
-
- /*
- * rtd remaining settings
- */
- rtd->card = card;
- rtd->dai_link = dai_link;
+ rtd->card = card;
+ rtd->dai_link = dai_link;
+ rtd->num = card->num_rtd++;
/* see for_each_card_rtds */
list_add_tail(&rtd->list, &card->rtd_list);
- rtd->num = card->num_rtd;
- card->num_rtd++;
return rtd;
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime()" to the asoc tree
@ 2020-04-14 14:48 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-04-14 14:48 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime()
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From 49648d02008dd75ddd84b16f9abf29f24c383519 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 30 Mar 2020 10:47:54 +0900
Subject: [PATCH] ASoC: soc-core: set rtd->num_cpu/codec at
soc_new_pcm_runtime()
rtd->cpu_dais/codec_dais are set at soc_new_pcm_runtime().
rtd->num_cpus/codecs should be set there.
This patch do it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87h7y6bol1.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/soc-core.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4ad64b96d893..8cef4fd6df36 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -490,6 +490,8 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
*/
rtd->cpu_dais = &rtd->dais[0];
rtd->codec_dais = &rtd->dais[dai_link->num_cpus];
+ rtd->num_cpus = dai_link->num_cpus;
+ rtd->num_codecs = dai_link->num_codecs;
/*
* rtd remaining settings
@@ -989,7 +991,6 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card,
if (!rtd)
return -ENOMEM;
- rtd->num_cpus = dai_link->num_cpus;
for_each_link_cpus(dai_link, i, cpu) {
asoc_rtd_to_cpu(rtd, i) = snd_soc_find_dai(cpu);
if (!asoc_rtd_to_cpu(rtd, i)) {
@@ -1004,7 +1005,6 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card,
rtd->cpu_dai = rtd->cpu_dais[0];
/* Find CODEC from registered CODECs */
- rtd->num_codecs = dai_link->num_codecs;
for_each_link_codecs(dai_link, i, codec) {
asoc_rtd_to_codec(rtd, i) = snd_soc_find_dai(codec);
if (!asoc_rtd_to_codec(rtd, i)) {
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
* Applied "ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer" to the asoc tree
@ 2020-04-14 14:48 Mark Brown
0 siblings, 0 replies; 116+ messages in thread
From: Mark Brown @ 2020-04-14 14:48 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: alsa-devel, Mark Brown, Pierre-Louis Bossart
The patch
ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
From c2233a266178f8937cc26a84cd7672334b5424b7 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Mon, 30 Mar 2020 10:47:37 +0900
Subject: [PATCH] ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro
for DAI pointer
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87imimboli.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
---
include/sound/soc.h | 12 ++++-----
sound/soc/soc-compress.c | 36 +++++++++++++--------------
sound/soc/soc-core.c | 16 ++++++------
sound/soc/soc-dapm.c | 4 +--
sound/soc/soc-generic-dmaengine-pcm.c | 6 ++---
sound/soc/soc-pcm.c | 30 +++++++++++-----------
6 files changed, 52 insertions(+), 52 deletions(-)
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 13458e4fbb13..4dff6745b067 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1179,16 +1179,16 @@ struct snd_soc_pcm_runtime {
(i)++)
#define for_each_rtd_cpu_dais(rtd, i, dai) \
for ((i) = 0; \
- ((i) < rtd->num_cpus) && ((dai) = rtd->cpu_dais[i]); \
+ ((i) < rtd->num_cpus) && ((dai) = asoc_rtd_to_cpu(rtd, i)); \
(i)++)
#define for_each_rtd_cpu_dais_rollback(rtd, i, dai) \
- for (; (--(i) >= 0) && ((dai) = rtd->cpu_dais[i]);)
+ for (; (--(i) >= 0) && ((dai) = asoc_rtd_to_cpu(rtd, i));)
#define for_each_rtd_codec_dais(rtd, i, dai) \
for ((i) = 0; \
- ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \
+ ((i) < rtd->num_codecs) && ((dai) = asoc_rtd_to_codec(rtd, i)); \
(i)++)
#define for_each_rtd_codec_dais_rollback(rtd, i, dai) \
- for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);)
+ for (; (--(i) >= 0) && ((dai) = asoc_rtd_to_codec(rtd, i));)
#define for_each_rtd_dais(rtd, i, dai) \
for ((i) = 0; \
((i) < (rtd)->num_cpus + (rtd)->num_codecs) && \
@@ -1381,8 +1381,8 @@ struct snd_soc_dai *snd_soc_card_get_codec_dai(struct snd_soc_card *card,
struct snd_soc_pcm_runtime *rtd;
list_for_each_entry(rtd, &card->rtd_list, list) {
- if (!strcmp(rtd->codec_dai->name, dai_name))
- return rtd->codec_dai;
+ if (!strcmp(asoc_rtd_to_codec(rtd, 0)->name, dai_name))
+ return asoc_rtd_to_codec(rtd, 0);
}
return NULL;
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 50062eb79adb..3f77c0208e09 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -73,7 +73,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_soc_component *component, *save = NULL;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int ret, i;
for_each_rtd_components(rtd, i, component) {
@@ -141,7 +141,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
struct snd_pcm_substream *fe_substream =
fe->pcm->streams[cstream->direction].substream;
struct snd_soc_component *component;
- struct snd_soc_dai *cpu_dai = fe->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0);
struct snd_soc_dpcm *dpcm;
struct snd_soc_dapm_widget_list *list;
int stream;
@@ -230,8 +230,8 @@ static int soc_compr_free(struct snd_compr_stream *cstream)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_soc_component *component;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int stream, i;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
@@ -274,7 +274,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream)
static int soc_compr_free_fe(struct snd_compr_stream *cstream)
{
struct snd_soc_pcm_runtime *fe = cstream->private_data;
- struct snd_soc_dai *cpu_dai = fe->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0);
struct snd_soc_dpcm *dpcm;
int stream, ret;
@@ -343,8 +343,8 @@ static int soc_compr_components_trigger(struct snd_compr_stream *cstream,
static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int ret;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
@@ -373,7 +373,7 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd)
static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd)
{
struct snd_soc_pcm_runtime *fe = cstream->private_data;
- struct snd_soc_dai *cpu_dai = fe->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0);
int ret, stream;
if (cmd == SND_COMPR_TRIGGER_PARTIAL_DRAIN ||
@@ -446,7 +446,7 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
struct snd_compr_params *params)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int ret;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
@@ -500,7 +500,7 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream,
struct snd_soc_pcm_runtime *fe = cstream->private_data;
struct snd_pcm_substream *fe_substream =
fe->pcm->streams[cstream->direction].substream;
- struct snd_soc_dai *cpu_dai = fe->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0);
int ret, stream;
if (cstream->direction == SND_COMPRESS_PLAYBACK)
@@ -558,7 +558,7 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream,
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_soc_component *component;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int i, ret = 0;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
@@ -632,7 +632,7 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes)
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_soc_component *component;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int i, ret = 0;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
@@ -664,7 +664,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream,
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_soc_component *component;
int i, ret = 0;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
@@ -711,7 +711,7 @@ static int soc_compr_set_metadata(struct snd_compr_stream *cstream,
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_soc_component *component;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int i, ret;
if (cpu_dai->driver->cops && cpu_dai->driver->cops->set_metadata) {
@@ -739,7 +739,7 @@ static int soc_compr_get_metadata(struct snd_compr_stream *cstream,
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_soc_component *component;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
int i, ret;
if (cpu_dai->driver->cops && cpu_dai->driver->cops->get_metadata) {
@@ -801,8 +801,8 @@ static struct snd_compr_ops soc_compr_dyn_ops = {
int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
{
struct snd_soc_component *component;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct snd_compr *compr;
struct snd_pcm *be_pcm;
char new_name[64];
@@ -891,7 +891,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
ret = snd_compress_new(rtd->card->snd_card, num, direction,
new_name, compr);
if (ret < 0) {
- component = rtd->codec_dai->component;
+ component = asoc_rtd_to_codec(rtd, 0)->component;
dev_err(component->dev,
"Compress ASoC: can't create compress for codec %s: %d\n",
component->name, ret);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 843b8b1c89d4..4ad64b96d893 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -364,7 +364,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime);
*/
void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
int playback = SNDRV_PCM_STREAM_PLAYBACK;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
@@ -991,13 +991,13 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card,
rtd->num_cpus = dai_link->num_cpus;
for_each_link_cpus(dai_link, i, cpu) {
- rtd->cpu_dais[i] = snd_soc_find_dai(cpu);
- if (!rtd->cpu_dais[i]) {
+ asoc_rtd_to_cpu(rtd, i) = snd_soc_find_dai(cpu);
+ if (!asoc_rtd_to_cpu(rtd, i)) {
dev_info(card->dev, "ASoC: CPU DAI %s not registered\n",
cpu->dai_name);
goto _err_defer;
}
- snd_soc_rtd_add_component(rtd, rtd->cpu_dais[i]->component);
+ snd_soc_rtd_add_component(rtd, asoc_rtd_to_cpu(rtd, i)->component);
}
/* Single cpu links expect cpu and cpu_dai in runtime data */
@@ -1006,14 +1006,14 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card,
/* Find CODEC from registered CODECs */
rtd->num_codecs = dai_link->num_codecs;
for_each_link_codecs(dai_link, i, codec) {
- rtd->codec_dais[i] = snd_soc_find_dai(codec);
- if (!rtd->codec_dais[i]) {
+ asoc_rtd_to_codec(rtd, i) = snd_soc_find_dai(codec);
+ if (!asoc_rtd_to_codec(rtd, i)) {
dev_info(card->dev, "ASoC: CODEC DAI %s not registered\n",
codec->dai_name);
goto _err_defer;
}
- snd_soc_rtd_add_component(rtd, rtd->codec_dais[i]->component);
+ snd_soc_rtd_add_component(rtd, asoc_rtd_to_codec(rtd, i)->component);
}
/* Single codec links expect codec and codec_dai in runtime data */
@@ -1062,7 +1062,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card,
struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai_link *dai_link = rtd->dai_link;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct snd_soc_component *component;
int ret, num, i;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 679ed60d850e..8f260994b149 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -4370,11 +4370,11 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
if (rtd->num_cpus == 1) {
for_each_rtd_codec_dais(rtd, i, codec_dai)
dapm_add_valid_dai_widget(card, rtd, codec_dai,
- rtd->cpu_dais[0]);
+ asoc_rtd_to_cpu(rtd, 0));
} else if (rtd->num_codecs == rtd->num_cpus) {
for_each_rtd_codec_dais(rtd, i, codec_dai)
dapm_add_valid_dai_widget(card, rtd, codec_dai,
- rtd->cpu_dais[i]);
+ asoc_rtd_to_cpu(rtd, i));
} else {
dev_err(card->dev,
"N cpus to M codecs link is not supported yet\n");
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index facf1922a714..f728309a0833 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -68,7 +68,7 @@ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
return -EINVAL;
}
- dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config);
if (ret)
@@ -134,7 +134,7 @@ dmaengine_pcm_set_runtime_hwparams(struct snd_soc_component *component,
return snd_soc_set_runtime_hwparams(substream,
pcm->config->pcm_hardware);
- dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
memset(&hw, 0, sizeof(hw));
hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
@@ -203,7 +203,7 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel(
return NULL;
}
- dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream);
if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0])
return pcm->chan[0];
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 289aebc15529..454735f8fa92 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -136,7 +136,7 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf,
return -ENOMEM;
for_each_pcm_streams(stream)
- if (snd_soc_dai_stream_valid(fe->cpu_dai, stream))
+ if (snd_soc_dai_stream_valid(asoc_rtd_to_cpu(fe, 0), stream))
offset += dpcm_show_state(fe, stream,
buf + offset,
out_count - offset);
@@ -836,10 +836,10 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
soc_pcm_init_runtime_hw(substream);
if (rtd->num_codecs == 1)
- codec_dai_name = rtd->codec_dai->name;
+ codec_dai_name = asoc_rtd_to_codec(rtd, 0)->name;
if (rtd->num_cpus == 1)
- cpu_dai_name = rtd->cpu_dai->name;
+ cpu_dai_name = asoc_rtd_to_cpu(rtd, 0)->name;
if (soc_pcm_has_symmetry(substream))
runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX;
@@ -1483,7 +1483,7 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget,
int dpcm_path_get(struct snd_soc_pcm_runtime *fe,
int stream, struct snd_soc_dapm_widget_list **list)
{
- struct snd_soc_dai *cpu_dai = fe->cpu_dai;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(fe, 0);
int paths;
if (fe->num_cpus > 1) {
@@ -1842,7 +1842,7 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream,
* DAIs connected to a single CPU DAI, use CPU DAI's directly
*/
if (be->num_codecs == 1) {
- codec_stream = snd_soc_dai_get_pcm_stream(be->codec_dais[0], stream);
+ codec_stream = snd_soc_dai_get_pcm_stream(asoc_rtd_to_codec(be, 0), stream);
*channels_min = max(*channels_min,
codec_stream->channels_min);
@@ -2759,7 +2759,7 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
return 0;
/* only check active links */
- if (!fe->cpu_dai->active)
+ if (!asoc_rtd_to_cpu(fe, 0)->active)
return 0;
/* DAPM sync will call this to update DSP paths */
@@ -2769,13 +2769,13 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
for_each_pcm_streams(stream) {
/* skip if FE doesn't have playback/capture capability */
- if (!snd_soc_dai_stream_valid(fe->cpu_dai, stream) ||
- !snd_soc_dai_stream_valid(fe->codec_dai, stream))
+ if (!snd_soc_dai_stream_valid(asoc_rtd_to_cpu(fe, 0), stream) ||
+ !snd_soc_dai_stream_valid(asoc_rtd_to_codec(fe, 0), stream))
continue;
/* skip if FE isn't currently playing/capturing */
- if (!fe->cpu_dai->stream_active[stream] ||
- !fe->codec_dai->stream_active[stream])
+ if (!asoc_rtd_to_cpu(fe, 0)->stream_active[stream] ||
+ !asoc_rtd_to_codec(fe, 0)->stream_active[stream])
continue;
paths = dpcm_path_get(fe, stream, &list);
@@ -2922,9 +2922,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
for_each_rtd_codec_dais(rtd, i, codec_dai) {
if (rtd->num_cpus == 1) {
- cpu_dai = rtd->cpu_dais[0];
+ cpu_dai = asoc_rtd_to_cpu(rtd, 0);
} else if (rtd->num_cpus == rtd->num_codecs) {
- cpu_dai = rtd->cpu_dais[i];
+ cpu_dai = asoc_rtd_to_cpu(rtd, i);
} else {
dev_err(rtd->card->dev,
"N cpus to M codecs link is not supported yet\n");
@@ -2971,7 +2971,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
snprintf(new_name, sizeof(new_name), "%s %s-%d",
rtd->dai_link->stream_name,
(rtd->num_codecs > 1) ?
- "multicodec" : rtd->codec_dai->name, num);
+ "multicodec" : asoc_rtd_to_codec(rtd, 0)->name, num);
ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback,
capture, &pcm);
@@ -3050,8 +3050,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
pcm->no_device_suspend = true;
out:
dev_info(rtd->card->dev, "%s <-> %s mapping ok\n",
- (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name,
- (rtd->num_cpus > 1) ? "multicpu" : rtd->cpu_dai->name);
+ (rtd->num_codecs > 1) ? "multicodec" : asoc_rtd_to_codec(rtd, 0)->name,
+ (rtd->num_cpus > 1) ? "multicpu" : asoc_rtd_to_cpu(rtd, 0)->name);
return ret;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 116+ messages in thread
end of thread, other threads:[~2020-04-14 15:13 UTC | newest]
Thread overview: 116+ messages (download: mbox.gz / follow: Atom feed)
-- links below jump to the message on this page --
2020-01-13 15:13 [alsa-devel] Applied "ASoC: sti: fix possible sleep-in-atomic" to the asoc tree Mark Brown
-- strict thread matches above, loose matches on Subject: below --
2020-04-14 14:48 Applied "ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer" " Mark Brown
2020-04-14 14:48 Applied "ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime()" " Mark Brown
2020-04-14 14:48 Applied "ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups" " Mark Brown
2020-04-14 14:48 Applied "ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais" " Mark Brown
2020-04-14 14:46 Applied "ASoC: rockchip-spdif: add #sound-dai-cells property" " Mark Brown
2020-04-14 14:46 Applied "ASoC: rockchip-spdif: add power-domains " Mark Brown
2020-04-14 14:46 Applied "ASoC: wsa881x: mark read_only_wordlength flag" " Mark Brown
2020-04-14 11:59 Applied "ASoC: tas571x: disable regulators on failed probe" " Mark Brown
2020-03-27 15:34 Applied "ASoC: SOF: Intel: add SoundWire configuration interface" " Mark Brown
2020-03-27 15:34 Applied "ASoC: SOF: IPC: dai-intel: move ALH declarations in header file" " Mark Brown
2020-03-27 15:34 Applied "ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks" " Mark Brown
2020-03-27 15:33 Applied "ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend" " Mark Brown
2020-03-27 15:33 Applied "ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers" " Mark Brown
2020-03-27 15:33 Applied "ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks" " Mark Brown
2020-03-27 15:33 Applied "ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire" " Mark Brown
2020-03-27 15:33 Applied "Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread" " Mark Brown
2020-03-26 19:05 Applied "ASoC: SOF: (cosmetic) use for_each_pcm_streams() in sof_dai_load()" " Mark Brown
2020-03-26 19:05 Applied "ASoC: SOF: fix uninitialised "work" with VirtIO" " Mark Brown
2020-03-25 21:33 Applied "ASoC: Intel: broadwell: Revert back SSP0 link to use dummy components" " Mark Brown
2020-03-24 17:16 Applied "dt-bindings: sound: convert rockchip i2s bindings to yaml" " Mark Brown
2020-03-24 17:16 Applied "dt-bindings: sound: rockchip-i2s: add #sound-dai-cells property" " Mark Brown
2020-03-20 20:45 Applied "ASoC: soc-core: Merge CPU/Codec DAIs" " Mark Brown
2020-03-20 20:45 Applied "ASoC: soc-core: Merge for_each_rtd_cpu/codec_dais()" " Mark Brown
2020-03-20 20:45 Applied "ASoC: soc-dapm: " Mark Brown
2020-03-20 20:45 Applied "ASoC: soc-core: Merge CPU/Codec for soc_dai_pcm_new()" " Mark Brown
2020-03-18 21:41 Applied "ASoC: codecs: wsa881x: remove soundwire stream handling" " Mark Brown
2020-03-16 18:08 Applied "ALSA: compress: add wma codec profiles" " Mark Brown
2020-03-16 18:08 Applied "ALSA: compress: Add wma decoder params" " Mark Brown
2020-03-16 18:08 Applied "ASoC: qcom: q6asm: pass codec profile to q6asm_open_write" " Mark Brown
2020-03-16 18:08 Applied "ASoC: qcom: q6asm: add support to wma config" " Mark Brown
2020-03-16 18:08 Applied "ASoC: qcom: q6asm-dai: add support to wma decoder" " Mark Brown
2020-03-16 18:08 Applied "ALSA: compress: add alac & ape decoder params" " Mark Brown
2020-03-16 18:08 Applied "ASoC: qcom: q6asm: add support for alac and ape configs" " Mark Brown
2020-03-16 18:07 Applied "ALSA: compress: bump the version" " Mark Brown
2020-03-13 16:18 Applied "ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI" " Mark Brown
2020-03-13 16:17 Applied "ASoC: Intel: sof_pcm512x: " Mark Brown
2020-03-13 16:17 Applied "ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms" " Mark Brown
2020-03-13 16:17 Applied "ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC" " Mark Brown
2020-03-13 16:17 Applied "ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations" " Mark Brown
2020-03-13 16:17 Applied "ASoC: Intel: skylake: " Mark Brown
2020-03-13 16:17 Applied "ASoC: Intel: (cosmetic) simplify structure member access" " Mark Brown
2020-03-13 16:17 Applied "ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp" " Mark Brown
2020-03-12 13:12 Applied "ASoC: SOF: Intel: Fix stream cleanup on hw free" " Mark Brown
2020-03-11 18:00 Applied "ASoC: zte: zx-tdm: remove redundant variables dev" " Mark Brown
2020-03-09 14:03 Applied "ASoC: jz4740-i2s: Fix divider written at incorrect offset in register" " Mark Brown
2020-03-06 15:03 Applied "ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates" " Mark Brown
2020-03-06 15:03 Applied "ASoC: pcm: Export parameter intersection logic" " Mark Brown
2020-03-06 15:03 Applied "ASoC: soc-pcm: move dpcm_fe_dai_close()" " Mark Brown
2020-03-06 15:03 Applied "ASoC: soc-pcm: add dpcm_fe_dai_cleanup()" " Mark Brown
2020-03-06 15:03 Applied "ASoC: soc-pcm: use snd_soc_dai_get_pcm_stream() at dpcm_set_fe_runtime()" " Mark Brown
2020-03-06 15:03 Applied "ASoC: soc-pcm: tidyup dulicate handing at dpcm_fe_dai_startup()" " Mark Brown
2020-03-05 13:15 Applied "ASoC: soc-pcm: use defined stream" " Mark Brown
2020-03-05 13:15 Applied "ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths()" " Mark Brown
2020-03-04 13:39 Applied "ASoC: stm32: sai: manage rebind issue" " Mark Brown
2020-03-03 17:07 Applied "ASoC: amd: Fix compile warning of argument type" " Mark Brown
2020-03-02 14:36 Applied "ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver" " Mark Brown
2020-02-26 18:48 Applied "ASoC: tas2562: Fix sample rate error message" " Mark Brown
2020-02-26 18:47 Applied "ASoC: Add dapm_add_valid_dai_widget helper" " Mark Brown
2020-02-26 18:47 Applied "ASoC: Add multiple CPU DAI support in DAPM" " Mark Brown
2020-02-26 18:47 Applied "ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding" " Mark Brown
2020-02-26 18:47 Applied "ASoC: tas2562: Add entries for the TAS2563 audio amplifier" " Mark Brown
2020-02-20 22:18 Applied "dt-bindings: sound: Add TLV320ADCx140 dt bindings" " Mark Brown
2020-02-19 20:57 Applied "ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro" " Mark Brown
2020-02-19 20:57 Applied "ASoC: qcom: sdm845: " Mark Brown
2020-02-19 20:57 Applied "ASoC: qcom: apq8016_sbc: " Mark Brown
2020-02-19 20:57 Applied "ASoC: intel: cml_rt1011_rt5682: " Mark Brown
2020-02-19 20:57 Applied "ASoC: intel: kbl_da7219_max98927: " Mark Brown
2020-02-19 20:57 Applied "ASoC: mediatek: mt8183-da7219-max98357: " Mark Brown
2020-02-19 20:57 Applied "ASoC: dt-bindings: Make RK3328 codec GPIO explicit" " Mark Brown
2020-02-19 20:56 Applied "ASoC: meson: add t9015 internal codec binding documentation" " Mark Brown
2020-02-19 0:11 Applied "ASoC: tas2562: Return invalid for when bitwidth is invalid" " Mark Brown
2020-02-19 0:11 Applied "ALSA: core: Expand DMA buffer information" " Mark Brown
2020-02-19 0:11 Applied "ALSA: core: Implement compress page allocation and free routines" " Mark Brown
2020-02-19 0:10 Applied "ASoC: SOF: Implement Probe IPC API" " Mark Brown
2020-02-19 0:10 Applied "ASoC: SOF: Generic probe compress operations" " Mark Brown
2020-02-19 0:10 Applied "ASoC: SOF: Provide probe debugfs support" " Mark Brown
2020-02-19 0:10 Applied "ASoC: SOF: Intel: Add Probe compress CPU DAIs" " Mark Brown
2020-02-19 0:10 Applied "ASoC: rt5682: Add DAI clock binding info for WCLK/BCLK CCF usage" " Mark Brown
2020-02-19 0:10 Applied "ASoC: soc-pcm: move dai_get_widget()" " Mark Brown
2020-02-19 0:10 Applied "ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be()" " Mark Brown
2020-02-19 0:10 Applied "ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be()" " Mark Brown
2020-02-19 0:10 Applied "ASoC: soc-pcm: use dpcm_get_be() " Mark Brown
2020-02-19 0:10 Applied "ASoC: soc-pcm: remove soc_dpcm_be_digital_mute()" " Mark Brown
2020-02-19 0:09 Applied "ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state()" " Mark Brown
2020-02-19 0:09 Applied "ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code" " Mark Brown
2020-02-19 0:09 Applied "ASoC: soc-pcm: use goto and remove multi return" " Mark Brown
2020-02-19 0:09 Applied "ASoC: soc-pcm: merge playback/cature_active into stream_active" " Mark Brown
2020-02-19 0:09 Applied "ALSA: pcm.h: add for_each_pcm_streams()" " Mark Brown
2020-02-19 0:09 Applied "ASoC: soc-core: use for_each_pcm_streams() macro" " Mark Brown
2020-02-19 0:09 Applied "ASoC: soc-pcm: " Mark Brown
2020-02-19 0:09 Applied "ASoC: soc-generic-dmaengine-pcm: " Mark Brown
2020-02-19 0:09 Applied "ASoC: dwc: dwc-i2s: " Mark Brown
2020-02-19 0:09 Applied "ASoC: fsl: fsl_asrc_dma: " Mark Brown
2020-02-19 0:09 Applied "ASoC: qcom: lpass-platform: " Mark Brown
2020-02-19 0:09 Applied "ASoC: sof: sof-audio: " Mark Brown
2020-02-19 0:09 Applied "ALSA: usx2y: " Mark Brown
2020-02-13 13:32 [alsa-devel] Applied "ASoC: mediatek: mt8183-da7219: change supported formats of DL2 and UL1" " Mark Brown
2020-02-13 13:32 [alsa-devel] Applied "ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed" " Mark Brown
2020-02-13 13:32 [alsa-devel] Applied "ASoC: mediatek: mt8183-da7219: support TDM out and 8ch I2S out" " Mark Brown
2020-02-13 13:32 [alsa-devel] Applied "ASoC: mediatek: mt8183-da7219: apply some refactors" " Mark Brown
2020-01-15 14:12 [alsa-devel] Applied "regulator fix for "regulator: core: Add regulator_is_equal() helper"" " Mark Brown
2020-01-13 16:11 [alsa-devel] Applied "ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m" " Mark Brown
2020-01-13 16:11 [alsa-devel] Applied "ASoC: rt1308-sdw: add rt1308 SdW amplifier driver" " Mark Brown
2020-01-13 15:13 [alsa-devel] Applied "ASoC: SOF: Intel: fix HDA codec driver probe with multiple controllers" " Mark Brown
2020-01-10 13:38 [alsa-devel] Applied "ASoC: max98090: Drop incorrectly applied duplicate commit" " Mark Brown
2019-12-27 1:08 [alsa-devel] Applied "ASoC: amd: Create multiple I2S platform device endpoint" " Mark Brown
2019-12-25 18:21 [alsa-devel] Applied "ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec" " Mark Brown
2019-12-25 18:21 [alsa-devel] Applied "ASoC: wcd934x: add basic controls" " Mark Brown
2019-12-25 18:21 [alsa-devel] Applied "ASoC: wcd934x: add playback dapm widgets" " Mark Brown
2019-12-25 18:21 [alsa-devel] Applied "ASoC: wcd934x: add capture " Mark Brown
2019-12-25 18:21 [alsa-devel] Applied "ASoC: wcd934x: add audio routings" " Mark Brown
2019-12-25 18:21 [alsa-devel] Applied "ASoC: qcom: dt-bindings: Add compatible for DB845c and Lenovo Yoga" " Mark Brown
2019-12-25 18:21 [alsa-devel] Applied "ASoC: qcom: sdm845: add support to " Mark Brown
2019-12-25 0:09 [alsa-devel] Applied "ASoC: fsl_audmix: add missed pm_runtime_disable" " Mark Brown
2019-11-18 17:48 [alsa-devel] Applied "ASoC: SOF: Intel: Fix build break" " Mark Brown
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