For almost all machines the DAI format is a constant, always set to the same thing. This means that not only should we normally set it on init rather than in hw_params() (where it has been for historical reasons) we should also allow users to configure this by setting a variable in the dai_link structure. The combination of these two will make many machine drivers even more data driven.

Implement a new dai_fmt field in the dai_link doing just that. Since 0 is a valid value for many format flags and we need to be able to tell if the field is actually set also add one to all the values used to configure formats.

Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- include/sound/soc-dai.h | 34 +++++++++++++++++----------------- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 22 ++++++++++++++++++++++ 3 files changed, 41 insertions(+), 17 deletions(-)

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 12d98b4..2413acc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -24,13 +24,13 @@ struct snd_pcm_substream; * Describes the physical PCM data formating and clocking. Add new formats * to the end. */ -#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */ -#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */ -#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */ -#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */ -#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */ -#define SND_SOC_DAIFMT_AC97 5 /* AC97 */ -#define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */ +#define SND_SOC_DAIFMT_I2S 1 /* I2S mode */ +#define SND_SOC_DAIFMT_RIGHT_J 2 /* Right Justified mode */ +#define SND_SOC_DAIFMT_LEFT_J 3 /* Left Justified mode */ +#define SND_SOC_DAIFMT_DSP_A 4 /* L data MSB after FRM LRC */ +#define SND_SOC_DAIFMT_DSP_B 5 /* L data MSB during FRM LRC */ +#define SND_SOC_DAIFMT_AC97 6 /* AC97 */ +#define SND_SOC_DAIFMT_PDM 7 /* Pulse density modulation */

/* left and right justified also known as MSB and LSB respectively */ #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J @@ -42,8 +42,8 @@ struct snd_pcm_substream; * DAI bit clocks can be be gated (disabled) when the DAI is not * sending or receiving PCM data in a frame. This can be used to save power. */ -#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */ -#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */ +#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */ +#define SND_SOC_DAIFMT_GATED (2 << 4) /* clock is gated */

/* * DAI hardware signal inversions. @@ -51,10 +51,10 @@ struct snd_pcm_substream; * Specifies whether the DAI can also support inverted clocks for the specified * format. */ -#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ -#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */ -#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */ -#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */ +#define SND_SOC_DAIFMT_NB_NF (1 << 8) /* normal bit clock + frame */ +#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ +#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */ +#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */

/* * DAI hardware clock masters. @@ -63,10 +63,10 @@ struct snd_pcm_substream; * i.e. if the codec is clk and FRM master then the interface is * clk and frame slave. */ -#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */ -#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */ -#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ -#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */ +#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */ +#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */ +#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */ +#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */

#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 diff --git a/include/sound/soc.h b/include/sound/soc.h index b499b37..a4dc699 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -713,6 +713,8 @@ struct snd_soc_dai_link { const char *cpu_dai_name; const char *codec_dai_name;

+ unsigned int dai_fmt; /* format to set on init */ + /* Keep DAI active over suspend */ unsigned int ignore_suspend:1;

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bd20154..b0cec3c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1317,6 +1317,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_codec *codec; struct snd_soc_codec_conf *codec_conf; enum snd_soc_compress_type compress_type; + struct snd_soc_dai_link *dai_link; int ret, i, order;

mutex_lock(&card->mutex); @@ -1429,6 +1430,27 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes);

+ for (i = 0; i < card->num_links; i++) { + dai_link = &card->dai_link[i]; + + if (dai_link->dai_fmt) { + pr_crit("%d %p\n", i, card->rtd[i].codec_dai); + ret = snd_soc_dai_set_fmt(card->rtd[i].codec_dai, + dai_link->dai_fmt); + if (ret != 0) + dev_warn(card->rtd[i].codec_dai->dev, + "Failed to set DAI format: %d\n", + ret); + + ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, + dai_link->dai_fmt); + if (ret != 0) + dev_warn(card->rtd[i].cpu_dai->dev, + "Failed to set DAI format: %d\n", + ret); + } + } + snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),