[alsa-devel] [PATCH 1/4] ASoC: qcom: sdm845: Add board specific dapm widgets
Add board specific dapm widgets so these widgets can be used in the route.
Signed-off-by: Rohit kumar rohitkr@codeaurora.org Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org --- sound/soc/qcom/sdm845.c | 10 ++++++++++ 1 file changed, 10 insertions(+)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 58593db2ab151..95d8d4422dae0 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -212,6 +212,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; }
+static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + static void sdm845_add_be_ops(struct snd_soc_card *card) { struct snd_soc_dai_link *link; @@ -243,6 +251,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) goto data_alloc_fail; }
+ card->dapm_widgets = sdm845_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card);
Add a callback for init ops on dai_link to create and setup jack.
Signed-off-by: Rohit kumar rohitkr@codeaurora.org Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org --- sound/soc/qcom/sdm845.c | 57 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 55 insertions(+), 2 deletions(-)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 95d8d4422dae0..43c03f8e8cdc2 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -8,6 +8,8 @@ #include <linux/of_device.h> #include <sound/pcm.h> #include <sound/pcm_params.h> +#include <sound/jack.h> +#include <uapi/linux/input-event-codes.h> #include "common.h" #include "qdsp6/q6afe.h"
@@ -17,6 +19,8 @@ #define MI2S_BCLK_RATE 1536000
struct sdm845_snd_data { + struct snd_soc_jack jack; + bool jack_setup; struct snd_soc_card *card; uint32_t pri_mi2s_clk_count; uint32_t sec_mi2s_clk_count; @@ -100,6 +104,54 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, return ret; }
+static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component; + struct snd_soc_dai_link *dai_link = rtd->dai_link; + struct snd_soc_card *card = rtd->card; + struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); + int i, rval; + + if (!pdata->jack_setup) { + struct snd_jack *jack; + + rval = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &pdata->jack, NULL, 0); + + if (rval < 0) { + dev_err(card->dev, "Unable to add Headphone Jack\n"); + return rval; + } + + jack = pdata->jack.jack; + + snd_jack_set_key(jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + pdata->jack_setup = true; + } + + for (i = 0 ; i < dai_link->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; + + component = dai->component; + rval = snd_soc_component_set_jack( + component, &pdata->jack, NULL); + if (rval != 0 && rval != -ENOTSUPP) { + dev_warn(card->dev, "Failed to set jack: %d\n", rval); + return rval; + } + } + + return 0; +} + + static int sdm845_snd_startup(struct snd_pcm_substream *substream) { unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; @@ -220,7 +272,7 @@ static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), };
-static void sdm845_add_be_ops(struct snd_soc_card *card) +static void sdm845_add_ops(struct snd_soc_card *card) { struct snd_soc_dai_link *link; int i; @@ -230,6 +282,7 @@ static void sdm845_add_be_ops(struct snd_soc_card *card) link->ops = &sdm845_be_ops; link->be_hw_params_fixup = sdm845_be_hw_params_fixup; } + link->init = sdm845_dai_init; } }
@@ -264,7 +317,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) data->card = card; snd_soc_card_set_drvdata(card, data);
- sdm845_add_be_ops(card); + sdm845_add_ops(card); ret = snd_soc_register_card(card); if (ret) { dev_err(dev, "Sound card registration failed\n");
On 24/11/18 11:09, Cheng-Yi Chiang wrote:
Add a callback for init ops on dai_link to create and setup jack.
Signed-off-by: Rohit kumar rohitkr@codeaurora.org Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org
Acked-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org
sound/soc/qcom/sdm845.c | 57 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 55 insertions(+), 2 deletions(-)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 95d8d4422dae0..43c03f8e8cdc2 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -8,6 +8,8 @@ #include <linux/of_device.h> #include <sound/pcm.h> #include <sound/pcm_params.h> +#include <sound/jack.h> +#include <uapi/linux/input-event-codes.h> #include "common.h" #include "qdsp6/q6afe.h"
@@ -17,6 +19,8 @@ #define MI2S_BCLK_RATE 1536000
struct sdm845_snd_data {
- struct snd_soc_jack jack;
- bool jack_setup; struct snd_soc_card *card; uint32_t pri_mi2s_clk_count; uint32_t sec_mi2s_clk_count;
@@ -100,6 +104,54 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, return ret; }
+static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) +{
- struct snd_soc_component *component;
- struct snd_soc_dai_link *dai_link = rtd->dai_link;
- struct snd_soc_card *card = rtd->card;
- struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card);
- int i, rval;
- if (!pdata->jack_setup) {
struct snd_jack *jack;
rval = snd_soc_card_jack_new(card, "Headset Jack",
SND_JACK_HEADSET |
SND_JACK_HEADPHONE |
SND_JACK_BTN_0 | SND_JACK_BTN_1 |
SND_JACK_BTN_2 | SND_JACK_BTN_3,
&pdata->jack, NULL, 0);
if (rval < 0) {
dev_err(card->dev, "Unable to add Headphone Jack\n");
return rval;
}
jack = pdata->jack.jack;
snd_jack_set_key(jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
snd_jack_set_key(jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
snd_jack_set_key(jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
snd_jack_set_key(jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
pdata->jack_setup = true;
- }
- for (i = 0 ; i < dai_link->num_codecs; i++) {
struct snd_soc_dai *dai = rtd->codec_dais[i];
component = dai->component;
rval = snd_soc_component_set_jack(
component, &pdata->jack, NULL);
if (rval != 0 && rval != -ENOTSUPP) {
dev_warn(card->dev, "Failed to set jack: %d\n", rval);
return rval;
}
- }
- return 0;
+}
- static int sdm845_snd_startup(struct snd_pcm_substream *substream) { unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS;
@@ -220,7 +272,7 @@ static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), };
-static void sdm845_add_be_ops(struct snd_soc_card *card) +static void sdm845_add_ops(struct snd_soc_card *card) { struct snd_soc_dai_link *link; int i; @@ -230,6 +282,7 @@ static void sdm845_add_be_ops(struct snd_soc_card *card) link->ops = &sdm845_be_ops; link->be_hw_params_fixup = sdm845_be_hw_params_fixup; }
} }link->init = sdm845_dai_init;
@@ -264,7 +317,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) data->card = card; snd_soc_card_set_drvdata(card, data);
- sdm845_add_be_ops(card);
- sdm845_add_ops(card); ret = snd_soc_register_card(card); if (ret) { dev_err(dev, "Sound card registration failed\n");
Set TDM time slots and DAI format for speaker codec. Set DAI format and clock for headset.
Signed-off-by: Rohit kumar rohitkr@codeaurora.org Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org --- 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); + 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); + 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; + + 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:
Thanks for the patch Jimmy,
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@codeaurora.org Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org
Overall the patch looks good for me, but this needs to be split into two patches + few more minor nits!
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!
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.
if (ret < 0)
dev_err(rtd->dev,
"snd_soc_dai_set_sysclk err = %d\n", ret);
case QUATERNARY_TDM_RX_0: case QUATERNARY_TDM_TX_0: ret = sdm845_tdm_snd_hw_params(substream, params);break;
@@ -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.
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:
On Tue, Nov 27, 2018 at 5:32 PM Srinivas Kandagatla srinivas.kandagatla@linaro.org 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@codeaurora.org Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org
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:
Select SND_SOC_RT5663 and SND_SOC_MAX98927 for SND_SOC_SDM845.
Signed-off-by: Rohit kumar rohitkr@codeaurora.org Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org --- sound/soc/qcom/Kconfig | 2 ++ 1 file changed, 2 insertions(+)
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 2a4c912d1e484..3528c4279cbae 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -100,6 +100,8 @@ config SND_SOC_SDM845 depends on QCOM_APR select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON + select SND_SOC_RT5663 + select SND_SOC_MAX98927 help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems.
On 24/11/18 11:09, Cheng-Yi Chiang wrote:
Select SND_SOC_RT5663 and SND_SOC_MAX98927 for SND_SOC_SDM845.
Signed-off-by: Rohit kumar rohitkr@codeaurora.org Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org
Acked-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org
sound/soc/qcom/Kconfig | 2 ++ 1 file changed, 2 insertions(+)
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 2a4c912d1e484..3528c4279cbae 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -100,6 +100,8 @@ config SND_SOC_SDM845 depends on QCOM_APR select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON
- select SND_SOC_RT5663
- select SND_SOC_MAX98927 help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems.
On 24/11/18 11:09, Cheng-Yi Chiang wrote:
Add board specific dapm widgets so these widgets can be used in the route.
Signed-off-by: Rohit kumar rohitkr@codeaurora.org Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org
Acked-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org
sound/soc/qcom/sdm845.c | 10 ++++++++++ 1 file changed, 10 insertions(+)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 58593db2ab151..95d8d4422dae0 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -212,6 +212,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; }
+static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_SPK("Left Spk", NULL),
- SND_SOC_DAPM_SPK("Right Spk", NULL),
- SND_SOC_DAPM_MIC("Int Mic", NULL),
+};
- static void sdm845_add_be_ops(struct snd_soc_card *card) { struct snd_soc_dai_link *link;
@@ -243,6 +251,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) goto data_alloc_fail; }
- card->dapm_widgets = sdm845_snd_widgets;
- card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card);
participants (2)
-
Cheng-Yi Chiang
-
Srinivas Kandagatla