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authorMark Brown <broonie@kernel.org>2022-06-09 11:26:00 +0100
committerMark Brown <broonie@kernel.org>2022-06-09 11:56:32 +0100
commitd3d8feadcc54c28015b62436a1f69d1080b85fb5 (patch)
tree1313057e514a83f76cb86fc687e4fca6b1488fee /sound/soc/qcom
parentff87d619ac180444db297f043962a5c325ded47b (diff)
parent28086d05ada6d03daa886aad0e469854b811311c (diff)
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Specify clock provider directly to CPU DAIs
Merge series from Charles Keepax <ckeepax@opensource.cirrus.com>: Currently the set_fmt callback always passes clock provider/consumer with respect to the CODEC. This made sense when the framework was directly broken down into platforms and CODECs. However, as things are now broken down into components which can be connected as either the CPU or CODEC side of a DAI link it simplifies things if each side of the link is just told if it is provider or consumer of the clocks. Making this change allows us to remove one of the last parts of the ASoC core that needs to know if a driver is a CODEC driver, where it flips the clock format specifier if a CODEC driver is used on the CPU side of a DAI link, as well as just being conceptually more consistent with componentisation. The basic idea of this patch chain is to change the set_fmt callback from specifying if the CODEC is provider/consumer into directly specifying if the component is provider/consumer. To do this we add some new defines, and then to preserve bisectability, the migration is done by adding a new callback, converting over all existing CPU side drivers, converting the core, and then finally reverting back to the old callback. Converting the platform drivers makes sense as the existing defines are from the perspective of the CODEC and there are more CODEC drivers than platform drivers. Obviously a fair amount of this patch chain I was only able to build test, so any testing that can be done would be greatly appreciated.
Diffstat (limited to 'sound/soc/qcom')
-rw-r--r--sound/soc/qcom/apq8016_sbc.c2
-rw-r--r--sound/soc/qcom/qdsp6/audioreach.c4
-rw-r--r--sound/soc/qcom/qdsp6/q6afe.c6
-rw-r--r--sound/soc/qcom/sc7180.c2
-rw-r--r--sound/soc/qcom/sdm845.c6
-rw-r--r--sound/soc/qcom/sm8250.c4
6 files changed, 12 insertions, 12 deletions
diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index b0a4f7ca2751..e54b8961112f 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -172,7 +172,7 @@ static int msm8916_qdsp6_dai_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
- snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS);
+ snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_BP_FP);
return apq8016_dai_init(rtd, qdsp6_dai_get_lpass_id(cpu_dai));
}
diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index 98c0efa1d0fe..01dac32c50fd 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -732,10 +732,10 @@ static int audioreach_i2s_set_media_format(struct q6apm_graph *graph,
intf_cfg->cfg.sd_line_idx = module->sd_line_idx;
switch (cfg->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
- case SND_SOC_DAIFMT_CBC_CFC:
+ case SND_SOC_DAIFMT_BP_FP:
intf_cfg->cfg.ws_src = CONFIG_I2S_WS_SRC_INTERNAL;
break;
- case SND_SOC_DAIFMT_CBP_CFP:
+ case SND_SOC_DAIFMT_BC_FC:
/* CPU is slave */
intf_cfg->cfg.ws_src = CONFIG_I2S_WS_SRC_EXTERNAL;
break;
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 625724852a7f..919e326b9462 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -1328,11 +1328,11 @@ int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg)
pcfg->i2s_cfg.bit_width = cfg->bit_width;
pcfg->i2s_cfg.data_format = AFE_LINEAR_PCM_DATA;
- switch (cfg->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
- case SND_SOC_DAIFMT_CBS_CFS:
+ switch (cfg->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
+ case SND_SOC_DAIFMT_BP_FP:
pcfg->i2s_cfg.ws_src = AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL;
break;
- case SND_SOC_DAIFMT_CBM_CFM:
+ case SND_SOC_DAIFMT_BC_FC:
/* CPU is slave */
pcfg->i2s_cfg.ws_src = AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL;
break;
diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c
index efccb5c0b3e0..f5f7c64b23a2 100644
--- a/sound/soc/qcom/sc7180.c
+++ b/sound/soc/qcom/sc7180.c
@@ -155,7 +155,7 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream)
}
snd_soc_dai_set_fmt(codec_dai,
- SND_SOC_DAIFMT_CBS_CFS |
+ SND_SOC_DAIFMT_BC_FC |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_I2S);
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 61fda790f375..d8d35563af00 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -316,8 +316,8 @@ 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;
+ unsigned int fmt = SND_SOC_DAIFMT_BP_FP;
+ unsigned int codec_dai_fmt = SND_SOC_DAIFMT_BC_FC;
struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
struct snd_soc_card *card = rtd->card;
struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card);
@@ -356,7 +356,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream)
snd_soc_dai_set_sysclk(cpu_dai,
Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT,
MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
- snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS);
+ snd_soc_dai_set_fmt(cpu_dai, fmt);
break;
diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c
index 6e1184c8b672..ce4a5713386a 100644
--- a/sound/soc/qcom/sm8250.c
+++ b/sound/soc/qcom/sm8250.c
@@ -96,8 +96,8 @@ static int sm8250_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
static int sm8250_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;
+ unsigned int fmt = SND_SOC_DAIFMT_BP_FP;
+ unsigned int codec_dai_fmt = SND_SOC_DAIFMT_BC_FC;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);