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SubjectRe: [PATCH] ASoC: qcom: add sdm845 sound card support
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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");
>

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