[alsa-devel] [PATCH v2 3/5] ASoC: sdm845: Add TDM configuration for speaker
Cheng-Yi Chiang
cychiang at chromium.org
Wed Nov 28 10:00:35 CET 2018
Set TDM time slots and DAI format for speaker codec.
Signed-off-by: Cheng-Yi Chiang <cychiang at 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:
--
2.20.0.rc0.387.gc7a69e6b6c-goog
More information about the Alsa-devel
mailing list