Re: [alsa-devel] [PATCH 3/4] ASoC: qcom: sdm845: Add codec related configuration for sdm845

From: Cheng-yi Chiang
Date: Wed Nov 28 2018 - 05:56:53 EST

On Tue, Nov 27, 2018 at 5:32 PM Srinivas Kandagatla
<srinivas.kandagatla@xxxxxxxxxx> wrote:
>
> Thanks for the patch Jimmy,
>
Hi Srini,
Thanks for the review!
I have updated the patch series in v2.

> On 24/11/18 11:09, Cheng-Yi Chiang wrote:
> > Set TDM time slots and DAI format for speaker codec.
> > Set DAI format and clock for headset. >
> > Signed-off-by: Rohit kumar <rohitkr@xxxxxxxxxxxxxx>
> > Signed-off-by: Cheng-Yi Chiang <cychiang@xxxxxxxxxxxx>
>
> Overall the patch looks good for me, but this needs to be split into two
> patches + few more minor nits!
Separated for speaker and headset in v2.
> > ---
> > sound/soc/qcom/sdm845.c | 82 ++++++++++++++++++++++++++++++++++++++++-
> > 1 file changed, 81 insertions(+), 1 deletion(-)
> >
> > diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
> > index 43c03f8e8cdc2..d815040e98dc9 100644
> > --- a/sound/soc/qcom/sdm845.c
> > +++ b/sound/soc/qcom/sdm845.c
> > @@ -6,12 +6,15 @@
> > #include <linux/module.h>
> > #include <linux/platform_device.h>
> > #include <linux/of_device.h>
> > +#include <sound/core.h>
> > #include <sound/pcm.h>
> > #include <sound/pcm_params.h>
> > #include <sound/jack.h>
> > +#include <sound/soc.h>
> > #include <uapi/linux/input-event-codes.h>
> > #include "common.h"
> > #include "qdsp6/q6afe.h"
> > +#include "../codecs/rt5663.h"
> >
> > #define DEFAULT_SAMPLE_RATE_48K 48000
> > #define DEFAULT_MCLK_RATE 24576000
> > @@ -34,7 +37,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
> > {
> > struct snd_soc_pcm_runtime *rtd = substream->private_data;
> > struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
> > - int ret = 0;
> > + int ret = 0, j;
> > int channels, slot_width;
> >
> > switch (params_format(params)) {
> > @@ -81,6 +84,31 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
> > goto end;
> > }
> > }
> > +
> > + for (j = 0; j < rtd->num_codecs; j++) {
> > + struct snd_soc_dai *codec_dai = rtd->codec_dais[j];
> > +
> > + if (!strcmp(codec_dai->component->name_prefix, "Left")) {
> > + ret = snd_soc_dai_set_tdm_slot(
> > + codec_dai, 0x30, 0x3, 8, slot_width);
>
> These constants needs some kind of defines to make the code more readable!
Fixed in v2.
>
> > + if (ret < 0) {
> > + dev_err(rtd->dev,
> > + "DEV0 TDM slot err:%d\n", ret);
> > + return ret;
> > + }
> > + }
> > +
> > + if (!strcmp(codec_dai->component->name_prefix, "Right")) {
> > + ret = snd_soc_dai_set_tdm_slot(
> > + codec_dai, 0xC0, 0x3, 8, slot_width);
> > + if (ret < 0) {
> > + dev_err(rtd->dev,
> > + "DEV1 TDM slot err:%d\n", ret);
> > + return ret;
> > + }
> > + }
> > + }
> > +
> > end:
> > return ret;
> > }
> > @@ -90,9 +118,26 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream,
> > {
> > struct snd_soc_pcm_runtime *rtd = substream->private_data;
> > struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
> > + struct snd_soc_dai *codec_dai = rtd->codec_dai;
> > int ret = 0;
> >
> > switch (cpu_dai->id) {
> > + case PRIMARY_MI2S_RX:
> > + case PRIMARY_MI2S_TX:
> > + /*
> > + * Use ASRC for internal clocks, as PLL rate isn't multiple
> > + * of BCLK.
> > + */
> > + rt5663_sel_asrc_clk_src(
> > + codec_dai->component,
> > + RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER,
> > + RT5663_CLK_SEL_I2S1_ASRC);
> > + ret = snd_soc_dai_set_sysclk(codec_dai,
> > + RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN);
>
> Use DEFAULT_MCLK_RATE here.
Fixed in v2.
>
>
> > + if (ret < 0)
> > + dev_err(rtd->dev,
> > + "snd_soc_dai_set_sysclk err = %d\n", ret);
> > + break;
> > case QUATERNARY_TDM_RX_0:
> > case QUATERNARY_TDM_TX_0:
> > ret = sdm845_tdm_snd_hw_params(substream, params);
> > @@ -155,14 +200,20 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd)
> > static int sdm845_snd_startup(struct snd_pcm_substream *substream)
> > {
> > unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS;
> > + unsigned int codec_dai_fmt = SND_SOC_DAIFMT_CBS_CFS;
> > struct snd_soc_pcm_runtime *rtd = substream->private_data;
> > struct snd_soc_card *card = rtd->card;
> > struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
> > struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
> > + struct snd_soc_dai *codec_dai = rtd->codec_dai;
> > +
> Unnecessary New line here.
Fixed in v2.
>
> > + int j;
> > + int ret;
> >
> > switch (cpu_dai->id) {
> > case PRIMARY_MI2S_RX:
> > case PRIMARY_MI2S_TX:
> > + codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF;
> > if (++(data->pri_mi2s_clk_count) == 1) {
> > snd_soc_dai_set_sysclk(cpu_dai,
> > Q6AFE_LPASS_CLK_ID_MCLK_1,
> > @@ -172,6 +223,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream)
> > MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
> > }
> > snd_soc_dai_set_fmt(cpu_dai, fmt);
> > + snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt);
> > break;
> >
> > case SECONDARY_MI2S_TX:
> > @@ -190,6 +242,34 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream)
> > Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT,
> > TDM_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
> > }
> > +
> > + codec_dai_fmt |= SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_DSP_B;
> > +
> > + for (j = 0; j < rtd->num_codecs; j++) {
> > + codec_dai = rtd->codec_dais[j];
> > +
> > + if (!strcmp(codec_dai->component->name_prefix,
> > + "Left")) {
> > + ret = snd_soc_dai_set_fmt(
> > + codec_dai, codec_dai_fmt);
> > + if (ret < 0) {
> > + dev_err(rtd->dev,
> > + "Left TDM fmt err:%d\n", ret);
> > + return ret;
> > + }
> > + }
> > +
> > + if (!strcmp(codec_dai->component->name_prefix,
> > + "Right")) {
> > + ret = snd_soc_dai_set_fmt(
> > + codec_dai, codec_dai_fmt);
> > + if (ret < 0) {
> > + dev_err(rtd->dev,
> > + "Right TDM slot err:%d\n", ret);
> > + return ret;
> > + }
> > + }
> > + }
> > break;
> >
> > default:
> >