[alsa-devel] [PATCH v2 1/5] 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 Wed, Nov 28, 2018 at 05:00:34PM +0800, 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
This looks like you're forwarding a patch from Rohit but shows you as the author?
Hello Mark,
On 12/7/2018 1:51 AM, Mark Brown wrote:
On Wed, Nov 28, 2018 at 05:00:34PM +0800, 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
This looks like you're forwarding a patch from Rohit but shows you as the author?
This was co-authored by me and Cheng-Yi. However, change is already merged from v1 itself.
Thanks, Rohit
Set TDM time slots and DAI format for speaker codec.
Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org --- sound/soc/qcom/sdm845.c | 69 ++++++++++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 43c03f8e8cdc2..d0df7ec6042e2 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -6,9 +6,11 @@ #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" @@ -17,6 +19,10 @@ #define DEFAULT_MCLK_RATE 24576000 #define TDM_BCLK_RATE 6144000 #define MI2S_BCLK_RATE 1536000 +#define LEFT_SPK_TDM_TX_MASK 0x30 +#define RIGHT_SPK_TDM_TX_MASK 0xC0 +#define SPK_TDM_RX_MASK 0x03 +#define NUM_TDM_SLOTS 8
struct sdm845_snd_data { struct snd_soc_jack jack; @@ -34,7 +40,7 @@ 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 +87,35 @@ 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, LEFT_SPK_TDM_TX_MASK, + SPK_TDM_RX_MASK, NUM_TDM_SLOTS, + 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, RIGHT_SPK_TDM_TX_MASK, + SPK_TDM_RX_MASK, NUM_TDM_SLOTS, + slot_width); + if (ret < 0) { + dev_err(rtd->dev, + "DEV1 TDM slot err:%d\n", ret); + return ret; + } + } + } + end: return ret; } @@ -155,10 +190,14 @@ 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: @@ -190,6 +229,34 @@ 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 28/11/18 09:00, Cheng-Yi Chiang wrote:
Set TDM time slots and DAI format for speaker codec.
Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org
Acked-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org
The patch
ASoC: sdm845: Add TDM configuration for speaker
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying to this mail.
Thanks, Mark
From 1d92332180659fab7d014eff5e0ab5e133424e3e Mon Sep 17 00:00:00 2001
From: Cheng-Yi Chiang cychiang@chromium.org Date: Wed, 28 Nov 2018 17:00:35 +0800 Subject: [PATCH] ASoC: sdm845: Add TDM configuration for speaker
Set TDM time slots and DAI format for speaker codec.
Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org Acked-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown broonie@kernel.org --- sound/soc/qcom/sdm845.c | 69 ++++++++++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 43c03f8e8cdc..d0df7ec6042e 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -6,9 +6,11 @@ #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" @@ -17,6 +19,10 @@ #define DEFAULT_MCLK_RATE 24576000 #define TDM_BCLK_RATE 6144000 #define MI2S_BCLK_RATE 1536000 +#define LEFT_SPK_TDM_TX_MASK 0x30 +#define RIGHT_SPK_TDM_TX_MASK 0xC0 +#define SPK_TDM_RX_MASK 0x03 +#define NUM_TDM_SLOTS 8
struct sdm845_snd_data { struct snd_soc_jack jack; @@ -34,7 +40,7 @@ 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 +87,35 @@ 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, LEFT_SPK_TDM_TX_MASK, + SPK_TDM_RX_MASK, NUM_TDM_SLOTS, + 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, RIGHT_SPK_TDM_TX_MASK, + SPK_TDM_RX_MASK, NUM_TDM_SLOTS, + slot_width); + if (ret < 0) { + dev_err(rtd->dev, + "DEV1 TDM slot err:%d\n", ret); + return ret; + } + } + } + end: return ret; } @@ -155,10 +190,14 @@ 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: @@ -190,6 +229,34 @@ 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:
The patch
ASoC: sdm845: Add TDM configuration for speaker
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying to this mail.
Thanks, Mark
From 09b382007aa54dbe2cc142ef78d69b9a7645fcd6 Mon Sep 17 00:00:00 2001
From: Cheng-Yi Chiang cychiang@chromium.org Date: Wed, 28 Nov 2018 17:00:35 +0800 Subject: [PATCH] ASoC: sdm845: Add TDM configuration for speaker
Set TDM time slots and DAI format for speaker codec.
Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org Acked-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown broonie@kernel.org --- sound/soc/qcom/sdm845.c | 69 ++++++++++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 43c03f8e8cdc..d0df7ec6042e 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -6,9 +6,11 @@ #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" @@ -17,6 +19,10 @@ #define DEFAULT_MCLK_RATE 24576000 #define TDM_BCLK_RATE 6144000 #define MI2S_BCLK_RATE 1536000 +#define LEFT_SPK_TDM_TX_MASK 0x30 +#define RIGHT_SPK_TDM_TX_MASK 0xC0 +#define SPK_TDM_RX_MASK 0x03 +#define NUM_TDM_SLOTS 8
struct sdm845_snd_data { struct snd_soc_jack jack; @@ -34,7 +40,7 @@ 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 +87,35 @@ 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, LEFT_SPK_TDM_TX_MASK, + SPK_TDM_RX_MASK, NUM_TDM_SLOTS, + 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, RIGHT_SPK_TDM_TX_MASK, + SPK_TDM_RX_MASK, NUM_TDM_SLOTS, + slot_width); + if (ret < 0) { + dev_err(rtd->dev, + "DEV1 TDM slot err:%d\n", ret); + return ret; + } + } + } + end: return ret; } @@ -155,10 +190,14 @@ 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: @@ -190,6 +229,34 @@ 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:
Set DAI format and sysclk for headset codec.
Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org --- sound/soc/qcom/sdm845.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index d0df7ec6042e2..1db8ef6682233 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -14,6 +14,7 @@ #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 @@ -125,9 +126,27 @@ 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, DEFAULT_MCLK_RATE, + 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); @@ -202,6 +221,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) 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, @@ -211,6 +231,7 @@ 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:
On 28/11/18 09:00, Cheng-Yi Chiang wrote:
Set DAI format and sysclk for headset codec.
Signed-off-by: Cheng-Yi Chiangcychiang@chromium.org
Acked-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org
The patch
ASoC: sdm845: Add configuration for headset codec
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying to this mail.
Thanks, Mark
From f7485875a68792d3e5d3984d521f2498ee385fa3 Mon Sep 17 00:00:00 2001
From: Cheng-Yi Chiang cychiang@chromium.org Date: Wed, 28 Nov 2018 17:00:36 +0800 Subject: [PATCH] ASoC: sdm845: Add configuration for headset codec
Set DAI format and sysclk for headset codec.
Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org Acked-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown broonie@kernel.org --- sound/soc/qcom/sdm845.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index d0df7ec6042e..1db8ef668223 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -14,6 +14,7 @@ #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 @@ -125,9 +126,27 @@ 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, DEFAULT_MCLK_RATE, + 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); @@ -202,6 +221,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) 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, @@ -211,6 +231,7 @@ 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:
The patch
ASoC: sdm845: Add configuration for headset codec
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying to this mail.
Thanks, Mark
From ce55698d32329b83ea695e0011b58d1bc11189b8 Mon Sep 17 00:00:00 2001
From: Cheng-Yi Chiang cychiang@chromium.org Date: Wed, 28 Nov 2018 17:00:36 +0800 Subject: [PATCH] ASoC: sdm845: Add configuration for headset codec
Set DAI format and sysclk for headset codec.
Signed-off-by: Cheng-Yi Chiang cychiang@chromium.org Acked-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown broonie@kernel.org --- sound/soc/qcom/sdm845.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index d0df7ec6042e..1db8ef668223 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -14,6 +14,7 @@ #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 @@ -125,9 +126,27 @@ 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, DEFAULT_MCLK_RATE, + 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); @@ -202,6 +221,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) 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, @@ -211,6 +231,7 @@ 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:
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 Wed, Nov 28, 2018 at 5:01 PM Cheng-Yi Chiang cychiang@chromium.org 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
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_RT5663select SND_SOC_MAX98927
This line is actually not needed.
We can drop this patch as there was another patch merged already: https://lkml.org/lkml/2018/12/10/875.
Thanks a lot for taking a look!
help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems.-- 2.20.0.rc0.387.gc7a69e6b6c-goog
Hi Jimmy,
On 28/11/18 09:00, 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
sound/soc/qcom/sdm845.c | 10 ++++++++++ 1 file changed, 10 insertions(+)
Am not sure why are you sending patches that are already applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/log/?h=for...
It just wastes other people time in looking at something which is already applied. Also it looks like you have not picked acked-bys on some of the patches!
thanks, srini
On Wed, Dec 05, 2018 at 02:18:25PM +0000, Srinivas Kandagatla wrote:
Am not sure why are you sending patches that are already applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/log/?h=for...
It just wastes other people time in looking at something which is already applied. Also it looks like you have not picked acked-bys on some of the patches!
I've applied the patches with acks, please resend anything else that needs applying.
participants (4)
-
Cheng-Yi Chiang -
Mark Brown -
Rohit Kumar -
Srinivas Kandagatla