[PATCH 0/6] ASoC: qdsp6: Add missing slimbus capture support

[PATCH 0/6] ASoC: qdsp6: Add missing slimbus capture support

[Srinivas Kandagatla]

This patchset adds support to slimbus tx ports plus 3 trival fixes
to do with closing port when its not really open.
slim tx support invoving add dais and routing, rest of the dsp
configuration commands are already available in mainline.

slim tx is verified with wcd9335 codec on DB820c.

Thanks,
srini

Srinivas Kandagatla (6):
  ASoC: qdsp6: q6afe: Add missing slimbus capture ports
  ASoC: qdsp6: q6afe-dai: add support to slim tx dais
  ASoC: qdsp6: q6routing: add slim rx routings
  ASoC: qdsp6: q6afe-dai: Do not overwrite slim dai num_channels
  ASoC: qdsp6: q6afe-dai: do not close port if its not opened
  ASoC: qdsp6: q6asm-dai: do not close port if its not opened

 sound/soc/qcom/qdsp6/q6afe-dai.c | 179 +++++++++++++++++++++++++++++++++++----
 sound/soc/qcom/qdsp6/q6afe.c     |  21 +++++
 sound/soc/qcom/qdsp6/q6asm-dai.c |   4 +-
 sound/soc/qcom/qdsp6/q6routing.c |  28 ++++++
 4 files changed, 216 insertions(+), 16 deletions(-)

-- 
2.16.2

[PATCH 1/6] ASoC: qdsp6: q6afe: Add missing slimbus capture ports

[Srinivas Kandagatla]

Existing code already has support for SLIMbus TX and RX, only thing
that was missing from TX side was mapping between virtual to actual
DSP port ids.

This patch adds those mappings.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6afe.c | 21 +++++++++++++++++++++
 1 file changed, 21 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 621b67b34db9..90e2d8e658bf 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -514,6 +514,20 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = {
 				SLIMBUS_5_RX, 1, 1},
 	[SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX,
 				SLIMBUS_6_RX, 1, 1},
+	[SLIMBUS_0_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX,
+				SLIMBUS_0_TX, 0, 1},
+	[SLIMBUS_1_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX,
+				SLIMBUS_1_TX, 0, 1},
+	[SLIMBUS_2_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX,
+				SLIMBUS_2_TX, 0, 1},
+	[SLIMBUS_3_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX,
+				SLIMBUS_3_TX, 0, 1},
+	[SLIMBUS_4_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX,
+				SLIMBUS_4_TX, 0, 1},
+	[SLIMBUS_5_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX,
+				SLIMBUS_5_TX, 0, 1},
+	[SLIMBUS_6_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX,
+				SLIMBUS_6_TX, 0, 1},
 	[PRIMARY_MI2S_RX] = { AFE_PORT_ID_PRIMARY_MI2S_RX,
 				PRIMARY_MI2S_RX, 1, 1},
 	[PRIMARY_MI2S_TX] = { AFE_PORT_ID_PRIMARY_MI2S_TX,
@@ -1372,6 +1386,13 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
 	case AFE_PORT_ID_MULTICHAN_HDMI_RX:
 		cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
 		break;
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX:
 	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX:
 	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX:
 	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX:
-- 
2.16.2

Applied "ASoC: qdsp6: q6afe: Add missing slimbus capture ports" to the asoc tree

[Mark Brown]

The patch

   ASoC: qdsp6: q6afe: Add missing slimbus capture ports

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 25090bc3f36cc3c171ec020dcc89c71db6bd0a67 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Wed, 4 Jul 2018 10:49:39 +0100
Subject: [PATCH] ASoC: qdsp6: q6afe: Add missing slimbus capture ports

Existing code already has support for SLIMbus TX and RX, only thing
that was missing from TX side was mapping between virtual to actual
DSP port ids.

This patch adds those mappings.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/qdsp6/q6afe.c | 21 +++++++++++++++++++++
 1 file changed, 21 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 671743453fbb..000775b4bba8 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -514,6 +514,20 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = {
 				SLIMBUS_5_RX, 1, 1},
 	[SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX,
 				SLIMBUS_6_RX, 1, 1},
+	[SLIMBUS_0_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX,
+				SLIMBUS_0_TX, 0, 1},
+	[SLIMBUS_1_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX,
+				SLIMBUS_1_TX, 0, 1},
+	[SLIMBUS_2_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX,
+				SLIMBUS_2_TX, 0, 1},
+	[SLIMBUS_3_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX,
+				SLIMBUS_3_TX, 0, 1},
+	[SLIMBUS_4_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX,
+				SLIMBUS_4_TX, 0, 1},
+	[SLIMBUS_5_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX,
+				SLIMBUS_5_TX, 0, 1},
+	[SLIMBUS_6_TX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX,
+				SLIMBUS_6_TX, 0, 1},
 	[PRIMARY_MI2S_RX] = { AFE_PORT_ID_PRIMARY_MI2S_RX,
 				PRIMARY_MI2S_RX, 1, 1},
 	[PRIMARY_MI2S_TX] = { AFE_PORT_ID_PRIMARY_MI2S_TX,
@@ -1372,6 +1386,13 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
 	case AFE_PORT_ID_MULTICHAN_HDMI_RX:
 		cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
 		break;
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX:
+	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX:
 	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX:
 	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX:
 	case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX:
-- 
2.18.0.rc2

[PATCH 2/6] ASoC: qdsp6: q6afe-dai: add support to slim tx dais

[Srinivas Kandagatla]

This patch adds support to SLIMbus TX dais in AFE module.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6afe-dai.c | 175 +++++++++++++++++++++++++++++++++++----
 1 file changed, 161 insertions(+), 14 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 1d2e5013c121..8dd3683eb367 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -382,23 +382,31 @@ static int q6slim_set_channel_map(struct snd_soc_dai *dai,
 	struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id];
 	int i;
 
-	if (!rx_slot) {
-		pr_err("%s: rx slot not found\n", __func__);
-		return -EINVAL;
-	}
+	if (dai->id & 0x1) {
+		/* TX */
+		if (!tx_slot) {
+			pr_err("%s: tx slot not found\n", __func__);
+			return -EINVAL;
+		}
 
-	for (i = 0; i < rx_num; i++) {
-		pcfg->slim.ch_mapping[i] =   rx_slot[i];
-		pr_debug("%s: find number of channels[%d] ch[%d]\n",
-		       __func__, i, rx_slot[i]);
-	}
+		for (i = 0; i < tx_num; i++)
+			pcfg->slim.ch_mapping[i] = tx_slot[i];
+
+		pcfg->slim.num_channels = tx_num;
+
+
+	} else {
+		if (!rx_slot) {
+			pr_err("%s: rx slot not found\n", __func__);
+			return -EINVAL;
+		}
 
-	pcfg->slim.num_channels = rx_num;
+		for (i = 0; i < rx_num; i++)
+			pcfg->slim.ch_mapping[i] =   rx_slot[i];
 
-	pr_debug("%s: SLIMBUS_%d_RX cnt[%d] ch[%d %d]\n", __func__,
-		(dai->id - SLIMBUS_0_RX) / 2, rx_num,
-		pcfg->slim.ch_mapping[0],
-		pcfg->slim.ch_mapping[1]);
+		pcfg->slim.num_channels = rx_num;
+
+	}
 
 	return 0;
 }
@@ -443,6 +451,14 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
 	{"Slimbus5 Playback", NULL, "SLIMBUS_5_RX"},
 	{"Slimbus6 Playback", NULL, "SLIMBUS_6_RX"},
 
+	{"SLIMBUS_0_TX", NULL, "Slimbus Capture"},
+	{"SLIMBUS_1_TX", NULL, "Slimbus1 Capture"},
+	{"SLIMBUS_2_TX", NULL, "Slimbus2 Capture"},
+	{"SLIMBUS_3_TX", NULL, "Slimbus3 Capture"},
+	{"SLIMBUS_4_TX", NULL, "Slimbus4 Capture"},
+	{"SLIMBUS_5_TX", NULL, "Slimbus5 Capture"},
+	{"SLIMBUS_6_TX", NULL, "Slimbus6 Capture"},
+
 	{"Primary MI2S Playback", NULL, "PRI_MI2S_RX"},
 	{"Secondary MI2S Playback", NULL, "SEC_MI2S_RX"},
 	{"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"},
@@ -636,6 +652,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 			.rate_min = 8000,
 			.rate_max = 192000,
 		},
+	}, {
+		.name = "SLIMBUS_0_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_0_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus1 Playback",
@@ -654,6 +688,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_1_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.name = "SLIMBUS_1_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_1_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus1 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus2 Playback",
@@ -672,6 +724,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_2_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_2_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_2_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus2 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus3 Playback",
@@ -690,6 +761,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_3_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_3_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_3_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus3 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus4 Playback",
@@ -708,6 +798,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_4_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_4_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_4_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus4 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus5 Playback",
@@ -726,6 +835,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_5_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_5_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_5_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus5 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus6 Playback",
@@ -744,6 +872,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_6_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_6_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_6_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus6 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Primary MI2S Playback",
-- 
2.16.2

Applied "ASoC: qdsp6: q6afe-dai: add support to slim tx dais" to the asoc tree

[Mark Brown]

The patch

   ASoC: qdsp6: q6afe-dai: add support to slim tx 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 f03d6b1b4d2460a749fb2826aa71e15a66104a88 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Wed, 4 Jul 2018 10:49:40 +0100
Subject: [PATCH] ASoC: qdsp6: q6afe-dai: add support to slim tx dais

This patch adds support to SLIMbus TX dais in AFE module.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/qdsp6/q6afe-dai.c | 175 ++++++++++++++++++++++++++++---
 1 file changed, 161 insertions(+), 14 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 1d2e5013c121..8dd3683eb367 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -382,23 +382,31 @@ static int q6slim_set_channel_map(struct snd_soc_dai *dai,
 	struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id];
 	int i;
 
-	if (!rx_slot) {
-		pr_err("%s: rx slot not found\n", __func__);
-		return -EINVAL;
-	}
+	if (dai->id & 0x1) {
+		/* TX */
+		if (!tx_slot) {
+			pr_err("%s: tx slot not found\n", __func__);
+			return -EINVAL;
+		}
 
-	for (i = 0; i < rx_num; i++) {
-		pcfg->slim.ch_mapping[i] =   rx_slot[i];
-		pr_debug("%s: find number of channels[%d] ch[%d]\n",
-		       __func__, i, rx_slot[i]);
-	}
+		for (i = 0; i < tx_num; i++)
+			pcfg->slim.ch_mapping[i] = tx_slot[i];
+
+		pcfg->slim.num_channels = tx_num;
+
+
+	} else {
+		if (!rx_slot) {
+			pr_err("%s: rx slot not found\n", __func__);
+			return -EINVAL;
+		}
 
-	pcfg->slim.num_channels = rx_num;
+		for (i = 0; i < rx_num; i++)
+			pcfg->slim.ch_mapping[i] =   rx_slot[i];
 
-	pr_debug("%s: SLIMBUS_%d_RX cnt[%d] ch[%d %d]\n", __func__,
-		(dai->id - SLIMBUS_0_RX) / 2, rx_num,
-		pcfg->slim.ch_mapping[0],
-		pcfg->slim.ch_mapping[1]);
+		pcfg->slim.num_channels = rx_num;
+
+	}
 
 	return 0;
 }
@@ -443,6 +451,14 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
 	{"Slimbus5 Playback", NULL, "SLIMBUS_5_RX"},
 	{"Slimbus6 Playback", NULL, "SLIMBUS_6_RX"},
 
+	{"SLIMBUS_0_TX", NULL, "Slimbus Capture"},
+	{"SLIMBUS_1_TX", NULL, "Slimbus1 Capture"},
+	{"SLIMBUS_2_TX", NULL, "Slimbus2 Capture"},
+	{"SLIMBUS_3_TX", NULL, "Slimbus3 Capture"},
+	{"SLIMBUS_4_TX", NULL, "Slimbus4 Capture"},
+	{"SLIMBUS_5_TX", NULL, "Slimbus5 Capture"},
+	{"SLIMBUS_6_TX", NULL, "Slimbus6 Capture"},
+
 	{"Primary MI2S Playback", NULL, "PRI_MI2S_RX"},
 	{"Secondary MI2S Playback", NULL, "SEC_MI2S_RX"},
 	{"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"},
@@ -636,6 +652,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 			.rate_min = 8000,
 			.rate_max = 192000,
 		},
+	}, {
+		.name = "SLIMBUS_0_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_0_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus1 Playback",
@@ -654,6 +688,24 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_1_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+	}, {
+		.name = "SLIMBUS_1_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_1_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus1 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus2 Playback",
@@ -672,6 +724,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_2_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_2_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_2_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus2 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus3 Playback",
@@ -690,6 +761,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_3_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_3_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_3_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus3 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus4 Playback",
@@ -708,6 +798,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_4_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_4_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_4_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus4 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus5 Playback",
@@ -726,6 +835,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_5_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_5_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_5_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus5 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Slimbus6 Playback",
@@ -744,6 +872,25 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 		.id = SLIMBUS_6_RX,
 		.probe = msm_dai_q6_dai_probe,
 		.remove = msm_dai_q6_dai_remove,
+
+	}, {
+		.name = "SLIMBUS_6_TX",
+		.ops = &q6slim_ops,
+		.id = SLIMBUS_6_TX,
+		.probe = msm_dai_q6_dai_probe,
+		.remove = msm_dai_q6_dai_remove,
+		.capture = {
+			.stream_name = "Slimbus6 Capture",
+			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+				 SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE |
+				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
+			.rate_min = 8000,
+			.rate_max = 192000,
+		},
 	}, {
 		.playback = {
 			.stream_name = "Primary MI2S Playback",
-- 
2.18.0.rc2

[PATCH 3/6] ASoC: qdsp6: q6routing: add slim rx routings

[Srinivas Kandagatla]

This patch add routings mixer controls for slim rx ports.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6routing.c | 28 ++++++++++++++++++++++++++++
 1 file changed, 28 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index c80fdbc2442e..35269b492761 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -68,6 +68,13 @@
 	{ mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" },	\
 	{ mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" },	\
 	{ mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" },		\
+	{ mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" },		\
+	{ mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" },		\
+	{ mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" },		\
+	{ mix_name, "SLIMBUS_3_TX", "SLIMBUS_3_TX" },		\
+	{ mix_name, "SLIMBUS_4_TX", "SLIMBUS_4_TX" },		\
+	{ mix_name, "SLIMBUS_5_TX", "SLIMBUS_5_TX" },		\
+	{ mix_name, "SLIMBUS_6_TX", "SLIMBUS_6_TX" },		\
 	{ mix_name, "PRIMARY_TDM_TX_0", "PRIMARY_TDM_TX_0"},	\
 	{ mix_name, "PRIMARY_TDM_TX_1", "PRIMARY_TDM_TX_1"},	\
 	{ mix_name, "PRIMARY_TDM_TX_2", "PRIMARY_TDM_TX_2"},	\
@@ -122,6 +129,27 @@
 	SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,		\
 		id, 1, 0, msm_routing_get_audio_mixer,			\
 		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_1_TX", SLIMBUS_1_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_2_TX", SLIMBUS_2_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_3_TX", SLIMBUS_3_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_4_TX", SLIMBUS_4_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_5_TX", SLIMBUS_5_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_6_TX", SLIMBUS_6_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
 	SOC_SINGLE_EXT("PRIMARY_TDM_TX_0", PRIMARY_TDM_TX_0,		\
 		id, 1, 0, msm_routing_get_audio_mixer,			\
 		msm_routing_put_audio_mixer),				\
-- 
2.16.2

Applied "ASoC: qdsp6: q6routing: add slim rx routings" to the asoc tree

[Mark Brown]

The patch

   ASoC: qdsp6: q6routing: add slim rx 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 9191ffe2d212f64aa2ec311f4294ba7066d1f8a1 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Wed, 4 Jul 2018 10:49:41 +0100
Subject: [PATCH] ASoC: qdsp6: q6routing: add slim rx routings

This patch add routings mixer controls for slim rx ports.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/qdsp6/q6routing.c | 28 ++++++++++++++++++++++++++++
 1 file changed, 28 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index c80fdbc2442e..35269b492761 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -68,6 +68,13 @@
 	{ mix_name, "SEC_MI2S_TX", "SEC_MI2S_TX" },	\
 	{ mix_name, "QUAT_MI2S_TX", "QUAT_MI2S_TX" },	\
 	{ mix_name, "TERT_MI2S_TX", "TERT_MI2S_TX" },		\
+	{ mix_name, "SLIMBUS_0_TX", "SLIMBUS_0_TX" },		\
+	{ mix_name, "SLIMBUS_1_TX", "SLIMBUS_1_TX" },		\
+	{ mix_name, "SLIMBUS_2_TX", "SLIMBUS_2_TX" },		\
+	{ mix_name, "SLIMBUS_3_TX", "SLIMBUS_3_TX" },		\
+	{ mix_name, "SLIMBUS_4_TX", "SLIMBUS_4_TX" },		\
+	{ mix_name, "SLIMBUS_5_TX", "SLIMBUS_5_TX" },		\
+	{ mix_name, "SLIMBUS_6_TX", "SLIMBUS_6_TX" },		\
 	{ mix_name, "PRIMARY_TDM_TX_0", "PRIMARY_TDM_TX_0"},	\
 	{ mix_name, "PRIMARY_TDM_TX_1", "PRIMARY_TDM_TX_1"},	\
 	{ mix_name, "PRIMARY_TDM_TX_2", "PRIMARY_TDM_TX_2"},	\
@@ -122,6 +129,27 @@
 	SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,		\
 		id, 1, 0, msm_routing_get_audio_mixer,			\
 		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_0_TX", SLIMBUS_0_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_1_TX", SLIMBUS_1_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_2_TX", SLIMBUS_2_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_3_TX", SLIMBUS_3_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_4_TX", SLIMBUS_4_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_5_TX", SLIMBUS_5_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
+	SOC_SINGLE_EXT("SLIMBUS_6_TX", SLIMBUS_6_TX,			\
+		id, 1, 0, msm_routing_get_audio_mixer,			\
+		msm_routing_put_audio_mixer),				\
 	SOC_SINGLE_EXT("PRIMARY_TDM_TX_0", PRIMARY_TDM_TX_0,		\
 		id, 1, 0, msm_routing_get_audio_mixer,			\
 		msm_routing_put_audio_mixer),				\
-- 
2.18.0.rc2

[PATCH 4/6] ASoC: qdsp6: q6afe-dai: Do not overwrite slim dai num_channels

[Srinivas Kandagatla]

num_channels for slim dais are aready set int set_channel_map,
do not overwrite them in hw_params.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6afe-dai.c | 1 -
 1 file changed, 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 8dd3683eb367..074582afda85 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -81,7 +81,6 @@ static int q6slim_hw_params(struct snd_pcm_substream *substream,
 	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
 	struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim;
 
-	slim->num_channels = params_channels(params);
 	slim->sample_rate = params_rate(params);
 
 	switch (params_format(params)) {
-- 
2.16.2

Applied "ASoC: qdsp6: q6afe-dai: Do not overwrite slim dai num_channels" to the asoc tree

[Mark Brown]

The patch

   ASoC: qdsp6: q6afe-dai: Do not overwrite slim dai num_channels

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 f1478a1476d45c5d7b070c726f46a7b581d3103c Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Wed, 4 Jul 2018 10:49:42 +0100
Subject: [PATCH] ASoC: qdsp6: q6afe-dai: Do not overwrite slim dai
 num_channels

num_channels for slim dais are aready set int set_channel_map,
do not overwrite them in hw_params.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/qdsp6/q6afe-dai.c | 1 -
 1 file changed, 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 8dd3683eb367..074582afda85 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -81,7 +81,6 @@ static int q6slim_hw_params(struct snd_pcm_substream *substream,
 	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
 	struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim;
 
-	slim->num_channels = params_channels(params);
 	slim->sample_rate = params_rate(params);
 
 	switch (params_format(params)) {
-- 
2.18.0.rc2

[PATCH 5/6] ASoC: qdsp6: q6afe-dai: do not close port if its not opened

[Srinivas Kandagatla]

afe ports are open as part of prepare, so for use cases like
"aplay sample.wav" were sample.wav is not present. This would
call port close eventhough port was never opened. DSP would
return errors for such use cases.

Avoid doing this by checking the port state.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6afe-dai.c | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 074582afda85..e7f809e3ee1f 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -314,6 +314,9 @@ static void q6afe_dai_shutdown(struct snd_pcm_substream *substream,
 	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
 	int rc;
 
+	if (!dai_data->is_port_started[dai->id])
+		return;
+
 	rc = q6afe_port_stop(dai_data->port[dai->id]);
 	if (rc < 0)
 		dev_err(dai->dev, "fail to close AFE port (%d)\n", rc);
-- 
2.16.2

Applied "ASoC: qdsp6: q6afe-dai: do not close port if its not opened" to the asoc tree

[Mark Brown]

The patch

   ASoC: qdsp6: q6afe-dai: do not close port if its not opened

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 110743189c863e96dc08a581d56c50b965870a3f Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Wed, 4 Jul 2018 10:49:43 +0100
Subject: [PATCH] ASoC: qdsp6: q6afe-dai: do not close port if its not opened

afe ports are open as part of prepare, so for use cases like
"aplay sample.wav" were sample.wav is not present. This would
call port close eventhough port was never opened. DSP would
return errors for such use cases.

Avoid doing this by checking the port state.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/qdsp6/q6afe-dai.c | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index 5002dd05bf27..a373ca5523ff 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -315,6 +315,9 @@ static void q6afe_dai_shutdown(struct snd_pcm_substream *substream,
 	struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
 	int rc;
 
+	if (!dai_data->is_port_started[dai->id])
+		return;
+
 	rc = q6afe_port_stop(dai_data->port[dai->id]);
 	if (rc < 0)
 		dev_err(dai->dev, "fail to close AFE port (%d)\n", rc);
-- 
2.18.0.rc2

[PATCH 6/6] ASoC: qdsp6: q6asm-dai: do not close port if its not opened

[Srinivas Kandagatla]

asm ports are open as part of prepare, so for use cases like
"aplay sample.wav" were sample.wav is not present. This would
call port close eventhough port was never opened. DSP would
return errors for such use cases.

Avoid doing this by checking the port state.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 1196dc7483d2..199d0fb6f90a 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -390,7 +390,9 @@ static int q6asm_dai_close(struct snd_pcm_substream *substream)
 	struct q6asm_dai_rtd *prtd = runtime->private_data;
 
 	if (prtd->audio_client) {
-		q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+		if (prtd->state)
+			q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+
 		q6asm_unmap_memory_regions(substream->stream,
 					   prtd->audio_client);
 		q6asm_audio_client_free(prtd->audio_client);
-- 
2.16.2

Applied "ASoC: qdsp6: q6asm-dai: do not close port if its not opened" to the asoc tree

[Mark Brown]

The patch

   ASoC: qdsp6: q6asm-dai: do not close port if its not opened

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 5dffc1752cabde6396fca28ff8343febfa524512 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Wed, 4 Jul 2018 10:49:44 +0100
Subject: [PATCH] ASoC: qdsp6: q6asm-dai: do not close port if its not opened

asm ports are open as part of prepare, so for use cases like
"aplay sample.wav" were sample.wav is not present. This would
call port close eventhough port was never opened. DSP would
return errors for such use cases.

Avoid doing this by checking the port state.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 349c6a883c63..360936703b3d 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -390,7 +390,9 @@ static int q6asm_dai_close(struct snd_pcm_substream *substream)
 	struct q6asm_dai_rtd *prtd = runtime->private_data;
 
 	if (prtd->audio_client) {
-		q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+		if (prtd->state)
+			q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+
 		q6asm_unmap_memory_regions(substream->stream,
 					   prtd->audio_client);
 		q6asm_audio_client_free(prtd->audio_client);
-- 
2.18.0.rc2