* [PATCH] ASoC: qcom: add sdm845 sound card support
@ 2018-06-18 11:16 Rohit kumar
2018-06-19 5:05 ` [alsa-devel] " Vinod
2018-06-19 8:46 ` Srinivas Kandagatla
0 siblings, 2 replies; 9+ messages in thread
From: Rohit kumar @ 2018-06-18 11:16 UTC (permalink / raw)
To: lgirdwood, broonie, robh+dt, mark.rutland, plai, bgoswami, perex,
srinivas.kandagatla, tiwai, alsa-devel, devicetree, linux-kernel
Cc: Rohit kumar
This patch adds sdm845 audio machine driver support.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
---
.../devicetree/bindings/sound/qcom,sdm845.txt | 87 ++++
sound/soc/qcom/Kconfig | 9 +
sound/soc/qcom/Makefile | 2 +
sound/soc/qcom/sdm845.c | 539 +++++++++++++++++++++
4 files changed, 637 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/qcom,sdm845.txt
create mode 100644 sound/soc/qcom/sdm845.c
diff --git a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
new file mode 100644
index 0000000..fc0e98c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
@@ -0,0 +1,87 @@
+* Qualcomm Technologies Inc. SDM845 ASoC sound card driver
+
+This binding describes the SDM845 sound card, which uses qdsp for audio.
+
+- compatible:
+ Usage: required
+ Value type: <stringlist>
+ Definition: must be "qcom,sdm845-sndcard"
+
+- qcom,audio-routing:
+ Usage: Optional
+ Value type: <stringlist>
+ Definition: A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the
+ connection's sink, the second being the connection's
+ source. Valid names could be power supplies, MicBias
+ of codec and the jacks on the board.
+
+- cdc-vdd-supply:
+ Usage: Optional
+ Value type: <phandle>
+ Definition: phandle of regulator supply required for codec vdd.
+
+= dailinks
+Each subnode of sndcard represents either a dailink, and subnodes of each
+dailinks would be cpu/codec/platform dais.
+
+- link-name:
+ Usage: required
+ Value type: <string>
+ Definition: User friendly name for dai link
+
+= CPU, PLATFORM, CODEC dais subnodes
+- cpu:
+ Usage: required
+ Value type: <subnode>
+ Definition: cpu dai sub-node
+
+- codec:
+ Usage: required
+ Value type: <subnode>
+ Definition: codec dai sub-node
+
+- platform:
+ Usage: opional
+ Value type: <subnode>
+ Definition: platform dai sub-node
+
+- sound-dai:
+ Usage: required
+ Value type: <phandle>
+ Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node.
+
+Example:
+
+audio {
+ compatible = "qcom,sdm845-sndcard";
+ qcom,model = "sdm845-snd-card";
+ pinctrl-names = "pri_active", "pri_sleep";
+ pinctrl-0 = <&pri_mi2s_active &pri_mi2s_ws_active>;
+ pinctrl-1 = <&pri_mi2s_sleep &pri_mi2s_ws_sleep>;
+
+ qcom,audio-routing = "PRI_MI2S_RX Audio Mixer", "Pri-mi2s-gpio";
+
+ cdc-vdd-supply = <&pm8998_l14>;
+
+ fe@1 {
+ link-name = "MultiMedia1";
+ cpu {
+ sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>;
+ };
+ platform {
+ sound-dai = <&q6asmdai>;
+ };
+ };
+
+ be@1 {
+ link-name = "PRI MI2S Playback";
+ cpu {
+ sound-dai = <&q6afedai PRIMARY_MI2S_RX>;
+ };
+
+ platform {
+ sound-dai = <&q6routing>;
+ };
+ };
+};
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 87838fa..09de50d 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -90,3 +90,12 @@ config SND_SOC_MSM8996
Support for Qualcomm Technologies LPASS audio block in
APQ8096 SoC-based systems.
Say Y if you want to use audio device on this SoCs
+
+config SND_SOC_SDM845
+ tristate "SoC Machine driver for SDM845 boards"
+ depends on QCOM_APR
+ select SND_SOC_QDSP6
+ help
+ To add support for audio on Qualcomm Technologies Inc.
+ SDM845 SoC-based systems.
+ Say Y if you want to use audio device on this SoCs
diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile
index 206945b..ac9609e 100644
--- a/sound/soc/qcom/Makefile
+++ b/sound/soc/qcom/Makefile
@@ -14,10 +14,12 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o
snd-soc-storm-objs := storm.o
snd-soc-apq8016-sbc-objs := apq8016_sbc.o
snd-soc-apq8096-objs := apq8096.o
+snd-soc-sdm845-objs := sdm845.o
obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o
obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o
obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o
++obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o
#DSP lib
obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
new file mode 100644
index 0000000..70d2611
--- /dev/null
+++ b/sound/soc/qcom/sdm845.c
@@ -0,0 +1,539 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Copyright (c) 2018, The Linux Foundation. All rights reserved.
+ */
+
+#include <linux/module.h>
+#include <linux/component.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+#include <linux/atomic.h>
+#include <linux/of_device.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <linux/soc/qcom/apr.h>
+#include "qdsp6/q6afe.h"
+
+#define DEFAULT_SAMPLE_RATE_48K 48000
+#define DEFAULT_MCLK_RATE 24576000
+#define DEFAULT_BCLK_RATE 1536000
+
+struct sdm845_snd_data {
+ struct snd_soc_card *card;
+ struct regulator *vdd_supply;
+ struct snd_soc_dai_link dai_link[];
+};
+
+static struct mutex pri_mi2s_res_lock;
+static struct mutex quat_tdm_res_lock;
+static atomic_t pri_mi2s_clk_count;
+static atomic_t quat_tdm_clk_count;
+
+static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
+
+static int sdm845_tdm_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 *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+ int channels, slot_width;
+
+ channels = params_channels(params);
+ if (channels < 1 || channels > 8) {
+ pr_err("%s: invalid param channels %d\n",
+ __func__, channels);
+ return -EINVAL;
+ }
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S32_LE:
+ case SNDRV_PCM_FORMAT_S24_LE:
+ case SNDRV_PCM_FORMAT_S16_LE:
+ slot_width = 32;
+ break;
+ default:
+ pr_err("%s: invalid param format 0x%x\n",
+ __func__, params_format(params));
+ return -EINVAL;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0x3,
+ channels, slot_width);
+ if (ret < 0) {
+ pr_err("%s: failed to set tdm slot, err:%d\n",
+ __func__, ret);
+ goto end;
+ }
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL,
+ channels, tdm_slot_offset);
+ if (ret < 0) {
+ pr_err("%s: failed to set channel map, err:%d\n",
+ __func__, ret);
+ goto end;
+ }
+ } else {
+ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xf, 0,
+ channels, slot_width);
+ if (ret < 0) {
+ pr_err("%s: failed to set tdm slot, err:%d\n",
+ __func__, ret);
+ goto end;
+ }
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai, channels,
+ tdm_slot_offset, 0, NULL);
+ if (ret < 0) {
+ pr_err("%s: failed to set channel map, err:%d\n",
+ __func__, ret);
+ goto end;
+ }
+ }
+end:
+ return ret;
+}
+
+static int sdm845_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 *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+
+ switch (cpu_dai->id) {
+ case QUATERNARY_TDM_RX_0:
+ case QUATERNARY_TDM_TX_0:
+ ret = sdm845_tdm_snd_hw_params(substream, params);
+ break;
+ default:
+ pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
+ break;
+ }
+ return ret;
+}
+
+static int sdm845_snd_startup(struct snd_pcm_substream *substream)
+{
+ unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+ pr_debug("%s: dai_id: 0x%x\n", __func__, cpu_dai->id);
+ switch (cpu_dai->id) {
+ case PRIMARY_MI2S_RX:
+ case PRIMARY_MI2S_TX:
+ mutex_lock(&pri_mi2s_res_lock);
+ if (atomic_inc_return(&pri_mi2s_clk_count) == 1) {
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_MCLK_1,
+ DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
+ DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+ }
+ mutex_unlock(&pri_mi2s_res_lock);
+ snd_soc_dai_set_fmt(cpu_dai, fmt);
+ break;
+ case QUATERNARY_TDM_RX_0:
+ case QUATERNARY_TDM_TX_0:
+ mutex_lock(&quat_tdm_res_lock);
+ if (atomic_inc_return(&quat_tdm_clk_count) == 1) {
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT,
+ DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
+ }
+ mutex_unlock(&quat_tdm_res_lock);
+ break;
+ default:
+ pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
+ break;
+ }
+ return 0;
+}
+
+static void sdm845_snd_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+ pr_debug("%s: dai_id: 0x%x\n", __func__, cpu_dai->id);
+ switch (cpu_dai->id) {
+ case PRIMARY_MI2S_RX:
+ case PRIMARY_MI2S_TX:
+ mutex_lock(&pri_mi2s_res_lock);
+ if (!atomic_dec_return(&pri_mi2s_clk_count)) {
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_MCLK_1,
+ 0, SNDRV_PCM_STREAM_PLAYBACK);
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
+ 0, SNDRV_PCM_STREAM_PLAYBACK);
+ };
+ mutex_unlock(&pri_mi2s_res_lock);
+ break;
+ case QUATERNARY_TDM_RX_0:
+ case QUATERNARY_TDM_TX_0:
+ mutex_lock(&quat_tdm_res_lock);
+ if (!atomic_dec_return(&quat_tdm_clk_count)) {
+ snd_soc_dai_set_sysclk(cpu_dai,
+ Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT,
+ 0, SNDRV_PCM_STREAM_PLAYBACK);
+ }
+ mutex_unlock(&quat_tdm_res_lock);
+ break;
+ default:
+ pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
+ break;
+ }
+}
+
+static struct snd_soc_ops sdm845_be_ops = {
+ .hw_params = sdm845_snd_hw_params,
+ .startup = sdm845_snd_startup,
+ .shutdown = sdm845_snd_shutdown,
+};
+
+static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ rate->min = rate->max = DEFAULT_SAMPLE_RATE_48K;
+ channels->min = channels->max = 2;
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget sdm845_widgets[] = {
+ SND_SOC_DAPM_PINCTRL("Pri-mi2s-gpio", "pri_active", "pri_sleep"),
+ SND_SOC_DAPM_PINCTRL("Quat-tdm-gpio", "quat_active", "quat_sleep"),
+};
+
+static int sdm845_sbc_parse_of(struct snd_soc_card *card)
+{
+ struct device *dev = card->dev;
+ struct snd_soc_dai_link *link;
+ struct device_node *np, *codec, *platform, *cpu, *node;
+ int ret, num_links;
+ struct sdm845_snd_data *data;
+
+ ret = snd_soc_of_parse_card_name(card, "qcom,model");
+ if (ret) {
+ dev_err(dev, "Error parsing card name: %d\n", ret);
+ return ret;
+ }
+
+ node = dev->of_node;
+
+ /* DAPM routes */
+ if (of_property_read_bool(node, "qcom,audio-routing")) {
+ ret = snd_soc_of_parse_audio_routing(card,
+ "qcom,audio-routing");
+ if (ret)
+ return ret;
+ }
+
+ /* Populate links */
+ num_links = of_get_child_count(node);
+
+ dev_info(dev, "Found %d child audio dai links..\n", num_links);
+ /* Allocate the private data and the DAI link array */
+ data = kzalloc(sizeof(*data) + sizeof(*link) * num_links,
+ GFP_KERNEL);
+ if (!data)
+ return -ENOMEM;
+
+ card->dai_link = &data->dai_link[0];
+ card->num_links = num_links;
+ card->dapm_widgets = sdm845_widgets;
+ card->num_dapm_widgets = ARRAY_SIZE(sdm845_widgets);
+
+ link = data->dai_link;
+ data->card = card;
+
+ for_each_child_of_node(node, np) {
+ cpu = of_get_child_by_name(np, "cpu");
+ platform = of_get_child_by_name(np, "platform");
+ codec = of_get_child_by_name(np, "codec");
+
+ if (!cpu) {
+ dev_err(dev, "Can't find cpu DT node\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
+ if (!link->cpu_of_node) {
+ dev_err(card->dev, "error getting cpu phandle\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ link->platform_of_node = of_parse_phandle(platform,
+ "sound-dai", 0);
+ if (!link->platform_of_node) {
+ dev_err(card->dev, "error getting platform phandle\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
+ if (ret) {
+ dev_err(card->dev, "error getting cpu dai name\n");
+ goto fail;
+ }
+
+ if (codec) {
+ ret = snd_soc_of_get_dai_link_codecs(dev, codec, link);
+ if (ret < 0) {
+ dev_err(card->dev, "error getting codec dai name\n");
+ goto fail;
+ }
+ link->no_pcm = 1;
+ link->ignore_suspend = 1;
+ link->ignore_pmdown_time = 1;
+ link->ops = &sdm845_be_ops;
+ link->be_hw_params_fixup = sdm845_be_hw_params_fixup;
+ } else {
+ link->dynamic = 1;
+ link->codec_dai_name = "snd-soc-dummy-dai";
+ link->codec_name = "snd-soc-dummy";
+ }
+
+ ret = of_property_read_string(np, "link-name", &link->name);
+ if (ret) {
+ dev_err(card->dev,
+ "error getting codec dai_link name\n");
+ goto fail;
+ }
+
+ link->dpcm_playback = 1;
+ link->dpcm_capture = 1;
+ link->stream_name = link->name;
+ link++;
+ }
+ dev_set_drvdata(dev, card);
+ snd_soc_card_set_drvdata(card, data);
+
+ return ret;
+fail:
+ kfree(data);
+ return ret;
+}
+
+static void sdm845_init_supplies(struct device *dev)
+{
+ struct snd_soc_card *card = dev_get_drvdata(dev);
+ struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+
+ data->vdd_supply = regulator_get(dev, "cdc-vdd");
+ if (IS_ERR(data->vdd_supply)) {
+ dev_err(dev, "Unable to get regulator supplies\n");
+ data->vdd_supply = NULL;
+ return;
+ }
+
+ if (regulator_enable(data->vdd_supply))
+ dev_err(dev, "Unable to enable vdd supply\n");
+}
+
+static void sdm845_deinit_supplies(struct device *dev)
+{
+ struct snd_soc_card *card = dev_get_drvdata(dev);
+ struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+
+ if (!data->vdd_supply)
+ return;
+
+ regulator_disable(data->vdd_supply);
+ regulator_put(data->vdd_supply);
+}
+
+static int sdm845_bind(struct device *dev)
+{
+ struct snd_soc_card *card;
+ int ret;
+
+ card = kzalloc(sizeof(*card), GFP_KERNEL);
+ if (!card)
+ return -ENOMEM;
+
+ ret = component_bind_all(dev, card);
+ if (ret) {
+ dev_err(dev, "Audio components bind failed: %d\n", ret);
+ goto bind_fail;
+ }
+
+ card->dev = dev;
+ ret = sdm845_sbc_parse_of(card);
+ if (ret) {
+ dev_err(dev, "Error parsing OF data\n");
+ goto parse_dt_fail;
+ }
+ sdm845_init_supplies(dev);
+
+ mutex_init(&pri_mi2s_res_lock);
+ mutex_init(&quat_tdm_res_lock);
+ atomic_set(&pri_mi2s_clk_count, 0);
+ atomic_set(&quat_tdm_clk_count, 0);
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(dev, "Sound card registration failed\n");
+ goto register_card_fail;
+ }
+ return ret;
+
+register_card_fail:
+ mutex_destroy(&pri_mi2s_res_lock);
+ mutex_destroy(&quat_tdm_res_lock);
+ sdm845_deinit_supplies(dev);
+ kfree(snd_soc_card_get_drvdata(card));
+parse_dt_fail:
+ component_unbind_all(dev, card);
+bind_fail:
+ kfree(card);
+ return ret;
+}
+
+static void sdm845_unbind(struct device *dev)
+{
+ struct snd_soc_card *card = dev_get_drvdata(dev);
+ struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+
+ mutex_destroy(&pri_mi2s_res_lock);
+ mutex_destroy(&quat_tdm_res_lock);
+ if (data->vdd_supply)
+ regulator_put(data->vdd_supply);
+ component_unbind_all(dev, card);
+ snd_soc_unregister_card(card);
+ kfree(data);
+ kfree(card);
+}
+
+static const struct component_master_ops sdm845_ops = {
+ .bind = sdm845_bind,
+ .unbind = sdm845_unbind,
+};
+
+static int sdm845_runtime_resume(struct device *dev)
+{
+ struct snd_soc_card *card = dev_get_drvdata(dev);
+ struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+
+ if (!data->vdd_supply) {
+ dev_dbg(dev, "no supplies defined\n");
+ return 0;
+ }
+
+ if (regulator_enable(data->vdd_supply))
+ dev_err(dev, "Enable regulator supply failed\n");
+
+ return 0;
+}
+
+static int sdm845_runtime_suspend(struct device *dev)
+{
+ struct snd_soc_card *card = dev_get_drvdata(dev);
+ struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
+
+ if (!data->vdd_supply) {
+ dev_dbg(dev, "no supplies defined\n");
+ return 0;
+ }
+
+ if (regulator_disable(data->vdd_supply))
+ dev_err(dev, "Disable regulator supply failed\n");
+
+ return 0;
+}
+
+static const struct dev_pm_ops sdm845_pm_ops = {
+ SET_RUNTIME_PM_OPS(sdm845_runtime_suspend,
+ sdm845_runtime_resume, NULL)
+};
+
+static int sdm845_compare_of(struct device *dev, void *data)
+{
+ return dev->of_node == data;
+}
+
+static void sdm845_release_of(struct device *dev, void *data)
+{
+ of_node_put(data);
+}
+
+static int add_audio_components(struct device *dev,
+ struct component_match **matchptr)
+{
+ struct device_node *np, *platform, *cpu, *node, *dai_node;
+
+ node = dev->of_node;
+
+ for_each_child_of_node(node, np) {
+ cpu = of_get_child_by_name(np, "cpu");
+ if (cpu) {
+ dai_node = of_parse_phandle(cpu, "sound-dai", 0);
+ of_node_get(dai_node);
+ component_match_add_release(dev, matchptr,
+ sdm845_release_of,
+ sdm845_compare_of,
+ dai_node);
+ }
+
+ platform = of_get_child_by_name(np, "platform");
+ if (platform) {
+ dai_node = of_parse_phandle(platform, "sound-dai", 0);
+ component_match_add_release(dev, matchptr,
+ sdm845_release_of,
+ sdm845_compare_of,
+ dai_node);
+ }
+ }
+
+ return 0;
+}
+
+static int sdm845_snd_platform_probe(struct platform_device *pdev)
+{
+ struct component_match *match = NULL;
+ int ret;
+
+ ret = add_audio_components(&pdev->dev, &match);
+ if (ret)
+ return ret;
+
+ return component_master_add_with_match(&pdev->dev, &sdm845_ops, match);
+}
+
+static int sdm845_snd_platform_remove(struct platform_device *pdev)
+{
+ component_master_del(&pdev->dev, &sdm845_ops);
+ return 0;
+}
+
+static const struct of_device_id sdm845_snd_device_id[] = {
+ { .compatible = "qcom,sdm845-sndcard" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, sdm845_snd_device_id);
+
+static struct platform_driver sdm845_snd_driver = {
+ .probe = sdm845_snd_platform_probe,
+ .remove = sdm845_snd_platform_remove,
+ .driver = {
+ .name = "msm-snd-sdm845",
+ .pm = &sdm845_pm_ops,
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(sdm845_snd_device_id),
+ },
+};
+module_platform_driver(sdm845_snd_driver);
+
+MODULE_DESCRIPTION("sdm845 ASoC Machine Driver");
+MODULE_LICENSE("GPL v2");
--
Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc.,
is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.
^ permalink raw reply related [flat|nested] 9+ messages in thread
* Re: [alsa-devel] [PATCH] ASoC: qcom: add sdm845 sound card support
2018-06-18 11:16 [PATCH] ASoC: qcom: add sdm845 sound card support Rohit kumar
@ 2018-06-19 5:05 ` Vinod
2018-06-19 13:50 ` Rohit Kumar
2018-06-19 8:46 ` Srinivas Kandagatla
1 sibling, 1 reply; 9+ messages in thread
From: Vinod @ 2018-06-19 5:05 UTC (permalink / raw)
To: Rohit kumar
Cc: lgirdwood, broonie, robh+dt, mark.rutland, plai, bgoswami, perex,
srinivas.kandagatla, tiwai, alsa-devel, devicetree, linux-kernel
On 18-06-18, 16:46, Rohit kumar wrote:
> +struct sdm845_snd_data {
> + struct snd_soc_card *card;
> + struct regulator *vdd_supply;
> + struct snd_soc_dai_link dai_link[];
> +};
> +
> +static struct mutex pri_mi2s_res_lock;
> +static struct mutex quat_tdm_res_lock;
any reason why the locks can't be part of sdm845_snd_data?
Also why do we need two locks ?
> +static atomic_t pri_mi2s_clk_count;
> +static atomic_t quat_tdm_clk_count;
Any specific reason for using atomic variables?
> +static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
> +
> +static int sdm845_tdm_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 *cpu_dai = rtd->cpu_dai;
> + int ret = 0;
> + int channels, slot_width;
> +
> + channels = params_channels(params);
> + if (channels < 1 || channels > 8) {
I though ch = 0 would be caught by framework and IIRC ASoC doesn't
support more than 8 channels
> + pr_err("%s: invalid param channels %d\n",
> + __func__, channels);
> + return -EINVAL;
> + }
> +
> + switch (params_format(params)) {
> + case SNDRV_PCM_FORMAT_S32_LE:
> + case SNDRV_PCM_FORMAT_S24_LE:
> + case SNDRV_PCM_FORMAT_S16_LE:
> + slot_width = 32;
> + break;
> + default:
> + pr_err("%s: invalid param format 0x%x\n",
> + __func__, params_format(params));
why not use dev_err, bonus you get device name printer with the logs :)
> +static int sdm845_snd_startup(struct snd_pcm_substream *substream)
> +{
> + unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS;
> + struct snd_soc_pcm_runtime *rtd = substream->private_data;
> + struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
> +
> + pr_debug("%s: dai_id: 0x%x\n", __func__, cpu_dai->id);
It is good for debug but not very useful here, so removing it would be
good
> + switch (cpu_dai->id) {
> + case PRIMARY_MI2S_RX:
> + case PRIMARY_MI2S_TX:
> + mutex_lock(&pri_mi2s_res_lock);
> + if (atomic_inc_return(&pri_mi2s_clk_count) == 1) {
> + snd_soc_dai_set_sysclk(cpu_dai,
> + Q6AFE_LPASS_CLK_ID_MCLK_1,
> + DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
> + snd_soc_dai_set_sysclk(cpu_dai,
> + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
> + DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
> + }
> + mutex_unlock(&pri_mi2s_res_lock);
why do we need locking here? Can you please explain that.
> + snd_soc_dai_set_fmt(cpu_dai, fmt);
> + break;
empty line after break helps in readability
> +static int sdm845_sbc_parse_of(struct snd_soc_card *card)
> +{
> + struct device *dev = card->dev;
> + struct snd_soc_dai_link *link;
> + struct device_node *np, *codec, *platform, *cpu, *node;
> + int ret, num_links;
> + struct sdm845_snd_data *data;
> +
> + ret = snd_soc_of_parse_card_name(card, "qcom,model");
> + if (ret) {
> + dev_err(dev, "Error parsing card name: %d\n", ret);
> + return ret;
> + }
> +
> + node = dev->of_node;
> +
> + /* DAPM routes */
> + if (of_property_read_bool(node, "qcom,audio-routing")) {
> + ret = snd_soc_of_parse_audio_routing(card,
> + "qcom,audio-routing");
> + if (ret)
> + return ret;
> + }
so if we dont find audio-routing, then? we seems to continue..
--
~Vinod
^ permalink raw reply [flat|nested] 9+ messages in thread
* Re: [PATCH] ASoC: qcom: add sdm845 sound card support
2018-06-18 11:16 [PATCH] ASoC: qcom: add sdm845 sound card support Rohit kumar
2018-06-19 5:05 ` [alsa-devel] " Vinod
@ 2018-06-19 8:46 ` Srinivas Kandagatla
2018-06-19 13:58 ` [alsa-devel] " Rohit Kumar
1 sibling, 1 reply; 9+ messages in thread
From: Srinivas Kandagatla @ 2018-06-19 8:46 UTC (permalink / raw)
To: Rohit kumar, lgirdwood, broonie, robh+dt, mark.rutland, plai,
bgoswami, perex, tiwai, alsa-devel, devicetree, linux-kernel
Thanks Rohit for the patch!
On 18/06/18 12:16, Rohit kumar wrote:
> This patch adds sdm845 audio machine driver support.
>
> Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
> ---
> .../devicetree/bindings/sound/qcom,sdm845.txt | 87 ++++
> sound/soc/qcom/Kconfig | 9 +
> sound/soc/qcom/Makefile | 2 +
> sound/soc/qcom/sdm845.c | 539 +++++++++++++++++++++
> 4 files changed, 637 insertions(+)
> create mode 100644 Documentation/devicetree/bindings/sound/qcom,sdm845.txt
Split the bindings into a separate patch!
> create mode 100644 sound/soc/qcom/sdm845.c
>
> diff --git a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
> new file mode 100644
> index 0000000..fc0e98c
> --- /dev/null
> +++ b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
> @@ -0,0 +1,87 @@
> +* Qualcomm Technologies Inc. SDM845 ASoC sound card driver
> +
> +This binding describes the SDM845 sound card, which uses qdsp for audio.
> +
> +- compatible:
> + Usage: required
> + Value type: <stringlist>
> + Definition: must be "qcom,sdm845-sndcard"
> +
> +- qcom,audio-routing:
> + Usage: Optional
> + Value type: <stringlist>
> + Definition: A list of the connections between audio components.
> + Each entry is a pair of strings, the first being the
> + connection's sink, the second being the connection's
> + source. Valid names could be power supplies, MicBias
> + of codec and the jacks on the board.
> +
> +- cdc-vdd-supply:
> + Usage: Optional
> + Value type: <phandle>
> + Definition: phandle of regulator supply required for codec vdd.
> +
> += dailinks
> +Each subnode of sndcard represents either a dailink, and subnodes of each
> +dailinks would be cpu/codec/platform dais.
> +
> +- link-name:
> + Usage: required
> + Value type: <string>
> + Definition: User friendly name for dai link
> +
> += CPU, PLATFORM, CODEC dais subnodes
> +- cpu:
> + Usage: required
> + Value type: <subnode>
> + Definition: cpu dai sub-node
> +
> +- codec:
> + Usage: required
> + Value type: <subnode>
> + Definition: codec dai sub-node
> +
> +- platform:
> + Usage: opional
Optional
> + Value type: <subnode>
> + Definition: platform dai sub-node
> +
> +- sound-dai:
> + Usage: required
> + Value type: <phandle>
> + Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node.
> +
> +Example:
> +
> +audio {
> + compatible = "qcom,sdm845-sndcard";
> + qcom,model = "sdm845-snd-card";
> + pinctrl-names = "pri_active", "pri_sleep";
> + pinctrl-0 = <&pri_mi2s_active &pri_mi2s_ws_active>;
> + pinctrl-1 = <&pri_mi2s_sleep &pri_mi2s_ws_sleep>;
> +
> + qcom,audio-routing = "PRI_MI2S_RX Audio Mixer", "Pri-mi2s-gpio";
> +
> + cdc-vdd-supply = <&pm8998_l14>;
> +
> + fe@1 {
Lets not use fe or be reference here, its just a link.
> + link-name = "MultiMedia1";
> + cpu {
> + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>;
> + };
> + platform {
> + sound-dai = <&q6asmdai>;
> + };
asmdai has sound-cell specifier as 1, so this will dtc will throw
warning for this.
have a look at how 8996 is done.
> + };
> +
> + be@1 {
> + link-name = "PRI MI2S Playback";
> + cpu {
> + sound-dai = <&q6afedai PRIMARY_MI2S_RX>;
> + };
> +
> + platform {
> + sound-dai = <&q6routing>;
> + };
> + };
> +};
> diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
> index 87838fa..09de50d 100644
> --- a/sound/soc/qcom/Kconfig
> +++ b/sound/soc/qcom/Kconfig
> @@ -90,3 +90,12 @@ config SND_SOC_MSM8996
> Support for Qualcomm Technologies LPASS audio block in
> APQ8096 SoC-based systems.
> Say Y if you want to use audio device on this SoCs
> +
> +config SND_SOC_SDM845
> + tristate "SoC Machine driver for SDM845 boards"
> + depends on QCOM_APR
> + select SND_SOC_QDSP6
> + help
> + To add support for audio on Qualcomm Technologies Inc.
> + SDM845 SoC-based systems.
> + Say Y if you want to use audio device on this SoCs
> diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile
> index 206945b..ac9609e 100644
> --- a/sound/soc/qcom/Makefile
> +++ b/sound/soc/qcom/Makefile
> @@ -14,10 +14,12 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o
> snd-soc-storm-objs := storm.o
> snd-soc-apq8016-sbc-objs := apq8016_sbc.o
> snd-soc-apq8096-objs := apq8096.o
> +snd-soc-sdm845-objs := sdm845.o
>
> obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o
> obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o
> obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o
> ++obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o
?? looks like typo here.
>
> #DSP lib
> obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/
> diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
> new file mode 100644
> index 0000000..70d2611
> --- /dev/null
> +++ b/sound/soc/qcom/sdm845.c
> @@ -0,0 +1,539 @@
> +// SPDX-License-Identifier: GPL-2.0
> +/*
> + * Copyright (c) 2018, The Linux Foundation. All rights reserved.
> + */
> +
> +#include <linux/module.h>
> +#include <linux/component.h>
> +#include <linux/platform_device.h>
> +#include <linux/regulator/consumer.h>
> +#include <linux/atomic.h>
> +#include <linux/of_device.h>
> +#include <sound/pcm.h>
> +#include <sound/pcm_params.h>
> +#include <sound/soc.h>
> +#include <linux/soc/qcom/apr.h>
> +#include "qdsp6/q6afe.h"
> +
> +#define DEFAULT_SAMPLE_RATE_48K 48000
> +#define DEFAULT_MCLK_RATE 24576000
> +#define DEFAULT_BCLK_RATE 1536000
> +
> +struct sdm845_snd_data {
> + struct snd_soc_card *card;
> + struct regulator *vdd_supply;
> + struct snd_soc_dai_link dai_link[];
> +};
> +
> +static struct mutex pri_mi2s_res_lock;
> +static struct mutex quat_tdm_res_lock;
> +static atomic_t pri_mi2s_clk_count;
> +static atomic_t quat_tdm_clk_count;
> +
> +static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
> +static int sdm845_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 *cpu_dai = rtd->cpu_dai;
> + int ret = 0;
> +
> + switch (cpu_dai->id) {
> + case QUATERNARY_TDM_RX_0:
> + case QUATERNARY_TDM_TX_0:
> + ret = sdm845_tdm_snd_hw_params(substream, params);
> + break;
> + default:
> + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
> + break;
> + }
> + return ret;
> +}
> +
> +static int sdm845_snd_startup(struct snd_pcm_substream *substream)
> +{
> + unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS;
> + struct snd_soc_pcm_runtime *rtd = substream->private_data;
> + struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
> +
> + pr_debug("%s: dai_id: 0x%x\n", __func__, cpu_dai->id);
> + switch (cpu_dai->id) {
> + case PRIMARY_MI2S_RX:
> + case PRIMARY_MI2S_TX:
> + mutex_lock(&pri_mi2s_res_lock);
Mutex and atomic variables looks redundant here.
Can you explain why do you need both?
> + if (atomic_inc_return(&pri_mi2s_clk_count) == 1) { > + snd_soc_dai_set_sysclk(cpu_dai,
> + Q6AFE_LPASS_CLK_ID_MCLK_1,
> + DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
> + snd_soc_dai_set_sysclk(cpu_dai,
> + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
> + DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
> + }
> + mutex_unlock(&pri_mi2s_res_lock);
> + snd_soc_dai_set_fmt(cpu_dai, fmt);
> + break;
> + case QUATERNARY_TDM_RX_0:
> + case QUATERNARY_TDM_TX_0:
> + mutex_lock(&quat_tdm_res_lock);
> + if (atomic_inc_return(&quat_tdm_clk_count) == 1) {
> + snd_soc_dai_set_sysclk(cpu_dai,
> + Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT,
> + DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
> + }
> + mutex_unlock(&quat_tdm_res_lock);
> + break;
> + default:
> + pr_err("%s: invalid dai id 0x%x\n", __func__, cpu_dai->id);
> + break;
> + }
> + return 0;
> +}
> +
...
> +
> +static int sdm845_sbc_parse_of(struct snd_soc_card *card)
> +{
> + struct device *dev = card->dev;
> + struct snd_soc_dai_link *link;
> + struct device_node *np, *codec, *platform, *cpu, *node;
> + int ret, num_links;
> + struct sdm845_snd_data *data;
> +
> + ret = snd_soc_of_parse_card_name(card, "qcom,model");
> + if (ret) {
> + dev_err(dev, "Error parsing card name: %d\n", ret);
> + return ret;
> + }
> +
> + node = dev->of_node;
> +
> + /* DAPM routes */
> + if (of_property_read_bool(node, "qcom,audio-routing")) {
> + ret = snd_soc_of_parse_audio_routing(card,
> + "qcom,audio-routing");
> + if (ret)
> + return ret;
> + }
> +
> + /* Populate links */
> + num_links = of_get_child_count(node);
> +
> + dev_info(dev, "Found %d child audio dai links..\n", num_links);
Looks unnessary!
> + /* Allocate the private data and the DAI link array */
> + data = kzalloc(sizeof(*data) + sizeof(*link) * num_links,
> + GFP_KERNEL);
> + if (!data)
> + return -ENOMEM;
> +
> + card->dai_link = &data->dai_link[0];
> + card->num_links = num_links;
> + card->dapm_widgets = sdm845_widgets;
> + card->num_dapm_widgets = ARRAY_SIZE(sdm845_widgets);
> +
> + link = data->dai_link;
> + data->card = card;
> +
> + for_each_child_of_node(node, np) {
> + cpu = of_get_child_by_name(np, "cpu");
> + platform = of_get_child_by_name(np, "platform");
> + codec = of_get_child_by_name(np, "codec");
> +
> + if (!cpu) {
> + dev_err(dev, "Can't find cpu DT node\n");
> + ret = -EINVAL;
> + goto fail;
> + }
> +
> + link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
> + if (!link->cpu_of_node) {
> + dev_err(card->dev, "error getting cpu phandle\n");
> + ret = -EINVAL;
> + goto fail;
> + }
> +
> + link->platform_of_node = of_parse_phandle(platform,
> + "sound-dai", 0);
> + if (!link->platform_of_node) {
> + dev_err(card->dev, "error getting platform phandle\n");
> + ret = -EINVAL;
> + goto fail;
> + }
> +
> + ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
> + if (ret) {
> + dev_err(card->dev, "error getting cpu dai name\n");
> + goto fail;
> + }
> +
> + if (codec) {
> + ret = snd_soc_of_get_dai_link_codecs(dev, codec, link);
> + if (ret < 0) {
> + dev_err(card->dev, "error getting codec dai name\n");
> + goto fail;
> + }
> + link->no_pcm = 1;
> + link->ignore_suspend = 1;
> + link->ignore_pmdown_time = 1;
> + link->ops = &sdm845_be_ops;
> + link->be_hw_params_fixup = sdm845_be_hw_params_fixup;
> + } else {
> + link->dynamic = 1;
> + link->codec_dai_name = "snd-soc-dummy-dai";
> + link->codec_name = "snd-soc-dummy";
> + }
> +
You could optimize this code, have a look at apq8096.c which does
exactly same thing.
> + ret = of_property_read_string(np, "link-name", &link->name);
> + if (ret) {
> + dev_err(card->dev,
> + "error getting codec dai_link name\n");
> + goto fail;
> + }
> +
> + link->dpcm_playback = 1;
> + link->dpcm_capture = 1;
> + link->stream_name = link->name;
> + link++;
> + }
> + dev_set_drvdata(dev, card);
> + snd_soc_card_set_drvdata(card, data);
> +
> + return ret;
> +fail:
> + kfree(data);
> + return ret;
> +}
...
> +
> +static void sdm845_unbind(struct device *dev)
> +{
> + struct snd_soc_card *card = dev_get_drvdata(dev);
> + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
> +
> + mutex_destroy(&pri_mi2s_res_lock);
> + mutex_destroy(&quat_tdm_res_lock);
> + if (data->vdd_supply)
> + regulator_put(data->vdd_supply);
> + component_unbind_all(dev, card);
> + snd_soc_unregister_card(card);
> + kfree(data);
> + kfree(card);
> +}
> +
> +static const struct component_master_ops sdm845_ops = {
> + .bind = sdm845_bind,
> + .unbind = sdm845_unbind,
> +};
> +
> +static int sdm845_runtime_resume(struct device *dev)
> +{
> + struct snd_soc_card *card = dev_get_drvdata(dev);
> + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
> +
> + if (!data->vdd_supply) {
> + dev_dbg(dev, "no supplies defined\n");
> + return 0;
> + }
> +
> + if (regulator_enable(data->vdd_supply))
> + dev_err(dev, "Enable regulator supply failed\n");
> +
> + return 0;
> +}
> +
> +static struct platform_driver sdm845_snd_driver = {
> + .probe = sdm845_snd_platform_probe,
> + .remove = sdm845_snd_platform_remove,
> + .driver = {
> + .name = "msm-snd-sdm845",
> + .pm = &sdm845_pm_ops,
> + .owner = THIS_MODULE,
not necessary!
> + .of_match_table = of_match_ptr(sdm845_snd_device_id),
> + },
> +};
> +module_platform_driver(sdm845_snd_driver);
> +
> +MODULE_DESCRIPTION("sdm845 ASoC Machine Driver");
> +MODULE_LICENSE("GPL v2");
>
^ permalink raw reply [flat|nested] 9+ messages in thread
* Re: [alsa-devel] [PATCH] ASoC: qcom: add sdm845 sound card support
2018-06-19 5:05 ` [alsa-devel] " Vinod
@ 2018-06-19 13:50 ` Rohit Kumar
2018-06-19 16:22 ` Vinod
0 siblings, 1 reply; 9+ messages in thread
From: Rohit Kumar @ 2018-06-19 13:50 UTC (permalink / raw)
To: Vinod
Cc: lgirdwood, broonie, robh+dt, mark.rutland, plai, bgoswami, perex,
srinivas.kandagatla, tiwai, alsa-devel, devicetree, linux-kernel
Thanks Vinod for reviewing.
On 6/19/2018 10:35 AM, Vinod wrote:
> On 18-06-18, 16:46, Rohit kumar wrote:
>
>> +struct sdm845_snd_data {
>> + struct snd_soc_card *card;
>> + struct regulator *vdd_supply;
>> + struct snd_soc_dai_link dai_link[];
>> +};
>> +
>> +static struct mutex pri_mi2s_res_lock;
>> +static struct mutex quat_tdm_res_lock;
> any reason why the locks can't be part of sdm845_snd_data?
> Also why do we need two locks ?
No specific reason, I will move it to sdm845_snd_data.
These locks are used to protect enable/disable of bit clocks. We have
Primary MI2S RX/TX
and Quaternary TDM RX/TX interfaces. For primary mi2s rx/tx, we have
single clock which is
synchronized with pri_mi2s_res_lock. For Quat TDM RX/TX, we are using
quat_tdm_res_lock.
We need two locks as we are protecting two different resources.
>
>> +static atomic_t pri_mi2s_clk_count;
>> +static atomic_t quat_tdm_clk_count;
> Any specific reason for using atomic variables?
Nothing as such. As we are using mutex to synchronize, we can make it
non- atomic.
Will do it in next-spin.
>
>> +static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
>> +
>> +static int sdm845_tdm_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 *cpu_dai = rtd->cpu_dai;
>> + int ret = 0;
>> + int channels, slot_width;
>> +
>> + channels = params_channels(params);
>> + if (channels < 1 || channels > 8) {
> I though ch = 0 would be caught by framework and IIRC ASoC doesn't
> support more than 8 channels
OK. Will check and remove.
>> + pr_err("%s: invalid param channels %d\n",
>> + __func__, channels);
>> + return -EINVAL;
>> + }
>> +
>> + switch (params_format(params)) {
>> + case SNDRV_PCM_FORMAT_S32_LE:
>> + case SNDRV_PCM_FORMAT_S24_LE:
>> + case SNDRV_PCM_FORMAT_S16_LE:
>> + slot_width = 32;
>> + break;
>> + default:
>> + pr_err("%s: invalid param format 0x%x\n",
>> + __func__, params_format(params));
> why not use dev_err, bonus you get device name printer with the logs :)
Sure. Will change it.
>> +static int sdm845_snd_startup(struct snd_pcm_substream *substream)
>> +{
>> + unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS;
>> + struct snd_soc_pcm_runtime *rtd = substream->private_data;
>> + struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
>> +
>> + pr_debug("%s: dai_id: 0x%x\n", __func__, cpu_dai->id);
> It is good for debug but not very useful here, so removing it would be
> good
OK
>> + switch (cpu_dai->id) {
>> + case PRIMARY_MI2S_RX:
>> + case PRIMARY_MI2S_TX:
>> + mutex_lock(&pri_mi2s_res_lock);
>> + if (atomic_inc_return(&pri_mi2s_clk_count) == 1) {
>> + snd_soc_dai_set_sysclk(cpu_dai,
>> + Q6AFE_LPASS_CLK_ID_MCLK_1,
>> + DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
>> + snd_soc_dai_set_sysclk(cpu_dai,
>> + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
>> + DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
>> + }
>> + mutex_unlock(&pri_mi2s_res_lock);
> why do we need locking here? Can you please explain that.
So, we can have two usecases: one with primary mi2s rx and other with
primary mi2s tx.
Lock is required to increment pri_mi2s_clk_count and enable clock so
that disable of one
usecase does not disable the clock.
>
>> + snd_soc_dai_set_fmt(cpu_dai, fmt);
>> + break;
> empty line after break helps in readability
Sure. Will add that change.
>> +static int sdm845_sbc_parse_of(struct snd_soc_card *card)
>> +{
>> + struct device *dev = card->dev;
>> + struct snd_soc_dai_link *link;
>> + struct device_node *np, *codec, *platform, *cpu, *node;
>> + int ret, num_links;
>> + struct sdm845_snd_data *data;
>> +
>> + ret = snd_soc_of_parse_card_name(card, "qcom,model");
>> + if (ret) {
>> + dev_err(dev, "Error parsing card name: %d\n", ret);
>> + return ret;
>> + }
>> +
>> + node = dev->of_node;
>> +
>> + /* DAPM routes */
>> + if (of_property_read_bool(node, "qcom,audio-routing")) {
>> + ret = snd_soc_of_parse_audio_routing(card,
>> + "qcom,audio-routing");
>> + if (ret)
>> + return ret;
>> + }
> so if we dont find audio-routing, then? we seems to continue..
Right. Its not mandatory to have qcom,audio-routing in device tree.
Regards,
Rohit
--
Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc.,
is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.
^ permalink raw reply [flat|nested] 9+ messages in thread
* Re: [alsa-devel] [PATCH] ASoC: qcom: add sdm845 sound card support
2018-06-19 8:46 ` Srinivas Kandagatla
@ 2018-06-19 13:58 ` Rohit Kumar
0 siblings, 0 replies; 9+ messages in thread
From: Rohit Kumar @ 2018-06-19 13:58 UTC (permalink / raw)
To: Srinivas Kandagatla, lgirdwood, broonie, robh+dt, mark.rutland,
plai, bgoswami, perex, tiwai, alsa-devel, devicetree,
linux-kernel
Thanks Srinivas for reviewing.
On 6/19/2018 2:16 PM, Srinivas Kandagatla wrote:
> Thanks Rohit for the patch!
>
> On 18/06/18 12:16, Rohit kumar wrote:
>> This patch adds sdm845 audio machine driver support.
>>
>> Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
>> ---
>> .../devicetree/bindings/sound/qcom,sdm845.txt | 87 ++++
>> sound/soc/qcom/Kconfig | 9 +
>> sound/soc/qcom/Makefile | 2 +
>> sound/soc/qcom/sdm845.c | 539
>> +++++++++++++++++++++
>> 4 files changed, 637 insertions(+)
>> create mode 100644
>> Documentation/devicetree/bindings/sound/qcom,sdm845.txt
>
> Split the bindings into a separate patch!
Sure, will do it in next spin.
>
>> create mode 100644 sound/soc/qcom/sdm845.c
>>
>> diff --git a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
>> b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
>> new file mode 100644
>> index 0000000..fc0e98c
>> --- /dev/null
>> +++ b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt
>> @@ -0,0 +1,87 @@
>> +* Qualcomm Technologies Inc. SDM845 ASoC sound card driver
>> +
>> +This binding describes the SDM845 sound card, which uses qdsp for
>> audio.
>>
[..]
>> +
>> +- codec:
>> + Usage: required
>> + Value type: <subnode>
>> + Definition: codec dai sub-node
>> +
>> +- platform:
>> + Usage: opional
>
> Optional
okay
>> + Value type: <subnode>
>> + Definition: platform dai sub-node
>> +
>> +- sound-dai:
>> + Usage: required
>> + Value type: <phandle>
>> + Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node.
>> +
>> +Example:
>> +
>> +audio {
>> + compatible = "qcom,sdm845-sndcard";
>> + qcom,model = "sdm845-snd-card";
>> + pinctrl-names = "pri_active", "pri_sleep";
>> + pinctrl-0 = <&pri_mi2s_active &pri_mi2s_ws_active>;
>> + pinctrl-1 = <&pri_mi2s_sleep &pri_mi2s_ws_sleep>;
>> +
>> + qcom,audio-routing = "PRI_MI2S_RX Audio Mixer", "Pri-mi2s-gpio";
>> +
>> + cdc-vdd-supply = <&pm8998_l14>;
>> +
>> + fe@1 {
> Lets not use fe or be reference here, its just a link.
okay
>
>> + link-name = "MultiMedia1";
>> + cpu {
>> + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>;
>> + };
>> + platform {
>> + sound-dai = <&q6asmdai>;
>> + };
> asmdai has sound-cell specifier as 1, so this will dtc will throw
> warning for this.
>
> have a look at how 8996 is done.
ok, sure
>
>> + };
>> +
>> + be@1 {
>>
[..]
>> diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile
>> index 206945b..ac9609e 100644
>> --- a/sound/soc/qcom/Makefile
>> +++ b/sound/soc/qcom/Makefile
>> @@ -14,10 +14,12 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) +=
>> snd-soc-lpass-apq8016.o
>> snd-soc-storm-objs := storm.o
>> snd-soc-apq8016-sbc-objs := apq8016_sbc.o
>> snd-soc-apq8096-objs := apq8096.o
>> +snd-soc-sdm845-objs := sdm845.o
>> obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o
>> obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o
>> obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o
>> ++obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o
>
> ?? looks like typo here.
>
Right. Will update.
>
>> #DSP lib
>> obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/
>> diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
>> new file mode 100644
>> index 0000000..70d2611
>> --- /dev/null
>> +++ b/sound/soc/qcom/sdm845.c
>> @@ -0,0 +1,539 @@
>> +// SPDX-License-Identifier: GPL-2.0
>> +/*
>> + * Copyright (c) 2018, The Linux Foundation. All rights reserved.
>> + */
>> +
>> +#include <linux/module.h>
>> +#include <linux/component.h>
[..]
>> +}
>> +
>> +static int sdm845_snd_startup(struct snd_pcm_substream *substream)
>> +{
>> + unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS;
>> + struct snd_soc_pcm_runtime *rtd = substream->private_data;
>> + struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
>> +
>> + pr_debug("%s: dai_id: 0x%x\n", __func__, cpu_dai->id);
>> + switch (cpu_dai->id) {
>> + case PRIMARY_MI2S_RX:
>> + case PRIMARY_MI2S_TX:
>> + mutex_lock(&pri_mi2s_res_lock);
>
> Mutex and atomic variables looks redundant here.
> Can you explain why do you need both?
Right. Only mutex is required here. I will make count as non-atomic.
>
> ...
>> +
>> +static int sdm845_sbc_parse_of(struct snd_soc_card *card)
>> +{
>> + struct device *dev = card->dev;
>> + struct snd_soc_dai_link *link;
>> + struct device_node *np, *codec, *platform, *cpu, *node;
>> + int ret, num_links;
>> + struct sdm845_snd_data *data;
>> +
>> + ret = snd_soc_of_parse_card_name(card, "qcom,model");
>> + if (ret) {
>> + dev_err(dev, "Error parsing card name: %d\n", ret);
>> + return ret;
>> + }
>> +
>> + node = dev->of_node;
>> +
>> + /* DAPM routes */
>> + if (of_property_read_bool(node, "qcom,audio-routing")) {
>> + ret = snd_soc_of_parse_audio_routing(card,
>> + "qcom,audio-routing");
>> + if (ret)
>> + return ret;
>> + }
>> +
>> + /* Populate links */
>> + num_links = of_get_child_count(node);
>> +
>> + dev_info(dev, "Found %d child audio dai links..\n", num_links);
>
> Looks unnessary!
Ok . Will remove it in next patchset.
>
>> +
>> + link->platform_of_node = of_parse_phandle(platform,
>> + "sound-dai", 0);
>> + if (!link->platform_of_node) {
>> + dev_err(card->dev, "error getting platform phandle\n");
>> + ret = -EINVAL;
>> + goto fail;
>> + }
>> +
>> + ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
>> + if (ret) {
>> + dev_err(card->dev, "error getting cpu dai name\n");
>> + goto fail;
>> + }
>> +
>> + if (codec) {
>> + ret = snd_soc_of_get_dai_link_codecs(dev, codec, link);
>> + if (ret < 0) {
>> + dev_err(card->dev, "error getting codec dai name\n");
>> + goto fail;
>> + }
>> + link->no_pcm = 1;
>> + link->ignore_suspend = 1;
>> + link->ignore_pmdown_time = 1;
>> + link->ops = &sdm845_be_ops;
>> + link->be_hw_params_fixup = sdm845_be_hw_params_fixup;
>> + } else {
>> + link->dynamic = 1;
>> + link->codec_dai_name = "snd-soc-dummy-dai";
>> + link->codec_name = "snd-soc-dummy";
>> + }
>> +
>
> You could optimize this code, have a look at apq8096.c which does
> exactly same thing.
>
Okay. will check and update.
> ...
>> +
>> +static void sdm845_unbind(struct device *dev)
>> +{
>> + struct snd_soc_card *card = dev_get_drvdata(dev);
>> + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
>> +
>> + mutex_destroy(&pri_mi2s_res_lock);
>> + mutex_destroy(&quat_tdm_res_lock);
>> + if (data->vdd_supply)
>> + regulator_put(data->vdd_supply);
>> + component_unbind_all(dev, card);
>> + snd_soc_unregister_card(card);
>> + kfree(data);
>> + kfree(card);
>> +}
>> +
>> +static const struct component_master_ops sdm845_ops = {
>> + .bind = sdm845_bind,
>> + .unbind = sdm845_unbind,
>> +};
>> +
>> +static int sdm845_runtime_resume(struct device *dev)
>> +{
>> + struct snd_soc_card *card = dev_get_drvdata(dev);
>> + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
>> +
>> + if (!data->vdd_supply) {
>> + dev_dbg(dev, "no supplies defined\n");
>> + return 0;
>> + }
>> +
>> + if (regulator_enable(data->vdd_supply))
>> + dev_err(dev, "Enable regulator supply failed\n");
>> +
>> + return 0;
>> +}
>> +
>> +static struct platform_driver sdm845_snd_driver = {
>> + .probe = sdm845_snd_platform_probe,
>> + .remove = sdm845_snd_platform_remove,
>> + .driver = {
>> + .name = "msm-snd-sdm845",
>> + .pm = &sdm845_pm_ops,
>> + .owner = THIS_MODULE,
> not necessary!
Okay. Will remove it.
>> + .of_match_table = of_match_ptr(sdm845_snd_device_id),
>> + },
>> +};
>> +module_platform_driver(sdm845_snd_driver);
>> +
>> +MODULE_DESCRIPTION("sdm845 ASoC Machine Driver");
>> +MODULE_LICENSE("GPL v2");
>>
> _______________________________________________
> Alsa-devel mailing list
> Alsa-devel@alsa-project.org
> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Regards,
Rohit
--
Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc.,
is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.
^ permalink raw reply [flat|nested] 9+ messages in thread
* Re: [alsa-devel] [PATCH] ASoC: qcom: add sdm845 sound card support
2018-06-19 13:50 ` Rohit Kumar
@ 2018-06-19 16:22 ` Vinod
[not found] ` <f7891675-4c98-7fc6-b80f-840ada9a1c5b@codeaurora.org>
0 siblings, 1 reply; 9+ messages in thread
From: Vinod @ 2018-06-19 16:22 UTC (permalink / raw)
To: Rohit Kumar
Cc: lgirdwood, broonie, robh+dt, mark.rutland, plai, bgoswami, perex,
srinivas.kandagatla, tiwai, alsa-devel, devicetree, linux-kernel
Hi Rohit,
On 19-06-18, 19:20, Rohit Kumar wrote:
> On 6/19/2018 10:35 AM, Vinod wrote:
> > On 18-06-18, 16:46, Rohit kumar wrote:
> >
> > > +struct sdm845_snd_data {
> > > + struct snd_soc_card *card;
> > > + struct regulator *vdd_supply;
> > > + struct snd_soc_dai_link dai_link[];
> > > +};
> > > +
> > > +static struct mutex pri_mi2s_res_lock;
> > > +static struct mutex quat_tdm_res_lock;
> > any reason why the locks can't be part of sdm845_snd_data?
> > Also why do we need two locks ?
> No specific reason, I will move it to sdm845_snd_data.
> These locks are used to protect enable/disable of bit clocks. We have
> Primary MI2S RX/TX
> and Quaternary TDM RX/TX interfaces. For primary mi2s rx/tx, we have single
> clock which is
> synchronized with pri_mi2s_res_lock. For Quat TDM RX/TX, we are using
> quat_tdm_res_lock.
> We need two locks as we are protecting two different resources.
I think bigger question is why do you need any locks? What is the race
scenario you envision which needs protection
--
~Vinod
^ permalink raw reply [flat|nested] 9+ messages in thread
* Re: [alsa-devel] [PATCH] ASoC: qcom: add sdm845 sound card support
[not found] ` <f7891675-4c98-7fc6-b80f-840ada9a1c5b@codeaurora.org>
@ 2018-06-20 9:31 ` Vinod
2018-06-20 9:53 ` Srinivas Kandagatla
0 siblings, 1 reply; 9+ messages in thread
From: Vinod @ 2018-06-20 9:31 UTC (permalink / raw)
To: Rohit Kumar
Cc: lgirdwood, broonie, robh+dt, mark.rutland, plai, bgoswami, perex,
srinivas.kandagatla, tiwai, alsa-devel, devicetree, linux-kernel
Hi Rohit,
On 20-06-18, 13:07, Rohit Kumar wrote:
> > On 19-06-18, 19:20, Rohit Kumar wrote:
> > > On 6/19/2018 10:35 AM, Vinod wrote:
> > > > On 18-06-18, 16:46, Rohit kumar wrote:
> > > >
> > > > > +struct sdm845_snd_data {
> > > > > + struct snd_soc_card *card;
> > > > > + struct regulator *vdd_supply;
> > > > > + struct snd_soc_dai_link dai_link[];
> > > > > +};
> > > > > +
> > > > > +static struct mutex pri_mi2s_res_lock;
> > > > > +static struct mutex quat_tdm_res_lock;
> > > > any reason why the locks can't be part of sdm845_snd_data?
> > > > Also why do we need two locks ?
> > > No specific reason, I will move it to sdm845_snd_data.
> > > These locks are used to protect enable/disable of bit clocks. We have
> > > Primary MI2S RX/TX
> > > and Quaternary TDM RX/TX interfaces. For primary mi2s rx/tx, we have single
> > > clock which is
> > > synchronized with pri_mi2s_res_lock. For Quat TDM RX/TX, we are using
> > > quat_tdm_res_lock.
> > > We need two locks as we are protecting two different resources.
> > I think bigger question is why do you need any locks? What is the race
> > scenario you envision which needs protection
> >
>
> Below is one of the race condition:
>
> Thread1 | Thread2
> ----------------------------------------------------------
> startup() |
> count++; | startup()
> read count (count = 1) |
> enable_clock() | count++; //count = 2
> shutdown() |
> count--;// count = 1 |
> | read count (count = 1)
> | enable_clock()
>
> Here clock will be enabled twice but disable will be called only once when
> count = 0.
>
> This will make the clock always enabled. So, I think we should keep either
> mutex lock or atomic variable to synchronize this.
we are using DPCM here right?
--
~Vinod
^ permalink raw reply [flat|nested] 9+ messages in thread
* Re: [alsa-devel] [PATCH] ASoC: qcom: add sdm845 sound card support
2018-06-20 9:31 ` Vinod
@ 2018-06-20 9:53 ` Srinivas Kandagatla
2018-06-20 10:28 ` Vinod
0 siblings, 1 reply; 9+ messages in thread
From: Srinivas Kandagatla @ 2018-06-20 9:53 UTC (permalink / raw)
To: Vinod, Rohit Kumar
Cc: lgirdwood, broonie, robh+dt, mark.rutland, plai, bgoswami, perex,
tiwai, alsa-devel, devicetree, linux-kernel
On 20/06/18 10:31, Vinod wrote:
>> Here clock will be enabled twice but disable will be called only once when
>> count = 0.
>>
>> This will make the clock always enabled. So, I think we should keep either
>> mutex lock or atomic variable to synchronize this.
> we are using DPCM here right?
We should probably model this clk as DAPM widget so we do not need to
handle this in machine code.
--srini
^ permalink raw reply [flat|nested] 9+ messages in thread
* Re: [alsa-devel] [PATCH] ASoC: qcom: add sdm845 sound card support
2018-06-20 9:53 ` Srinivas Kandagatla
@ 2018-06-20 10:28 ` Vinod
0 siblings, 0 replies; 9+ messages in thread
From: Vinod @ 2018-06-20 10:28 UTC (permalink / raw)
To: Srinivas Kandagatla
Cc: Rohit Kumar, lgirdwood, broonie, robh+dt, mark.rutland, plai,
bgoswami, perex, tiwai, alsa-devel, devicetree, linux-kernel
On 20-06-18, 10:53, Srinivas Kandagatla wrote:
> On 20/06/18 10:31, Vinod wrote:
> > > Here clock will be enabled twice but disable will be called only once when
> > > count = 0.
> > >
> > > This will make the clock always enabled. So, I think we should keep either
> > > mutex lock or atomic variable to synchronize this.
> > we are using DPCM here right?
>
> We should probably model this clk as DAPM widget so we do not need to handle
> this in machine code.
Sure that would be even better, but my point was that we need not worry
about .startup racing in case of DPCM as it holds a card mutex before it
calls PCM ops.. so it is already serialized..
--
~Vinod
^ permalink raw reply [flat|nested] 9+ messages in thread
end of thread, other threads:[~2018-06-20 10:28 UTC | newest]
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2018-06-18 11:16 [PATCH] ASoC: qcom: add sdm845 sound card support Rohit kumar
2018-06-19 5:05 ` [alsa-devel] " Vinod
2018-06-19 13:50 ` Rohit Kumar
2018-06-19 16:22 ` Vinod
[not found] ` <f7891675-4c98-7fc6-b80f-840ada9a1c5b@codeaurora.org>
2018-06-20 9:31 ` Vinod
2018-06-20 9:53 ` Srinivas Kandagatla
2018-06-20 10:28 ` Vinod
2018-06-19 8:46 ` Srinivas Kandagatla
2018-06-19 13:58 ` [alsa-devel] " Rohit Kumar
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