From 1e9de42f4324b91ce2e9da0939cab8fc6ae93893 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 7 Jan 2014 17:51:42 +0000 Subject: ASoC: dpcm: Explicitly set BE DAI link supported stream directions Some BE DAIs can be "dummy" (when the DSP is controlling the DAI) and as such wont have set a minimum number of playback or capture channels required for BE DAI registration (to establish supported stream directions). Force machine drivers to explicitly set whether they support playback and capture stream directions for every BE DAIs. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/include/sound/soc.h b/include/sound/soc.h index 1f741cb..a5ef14f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -886,6 +886,10 @@ struct snd_soc_dai_link { /* This DAI link can route to other DAI links at runtime (Frontend)*/ unsigned int dynamic:1; + /* DPCM capture and Playback support */ + unsigned int dpcm_capture:1; + unsigned int dpcm_playback:1; + /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 42782c0..141a302 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2026,10 +2026,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) int ret = 0, playback = 0, capture = 0; if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { - if (cpu_dai->driver->playback.channels_min) - playback = 1; - if (cpu_dai->driver->capture.channels_min) - capture = 1; + playback = rtd->dai_link->dpcm_playback; + capture = rtd->dai_link->dpcm_capture; } else { if (codec_dai->driver->playback.channels_min && cpu_dai->driver->playback.channels_min) -- cgit v0.10.2 From 16d7ea9167839d0b971ab29030886280595dd5fc Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jan 2014 14:19:16 +0100 Subject: ASoC: Allow PCMs to restrict the supported formats Some DMA cores might add additional restrictions on which in memory audio formats can be supported by the compound sound card. If the PCM component specifies a set of formats it supports (by setting the formats field to non 0) take these into account when calculating the format set for the compound sound card. Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Signed-off-by: Mark Brown diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 141a302..e7f16b5 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -158,7 +158,10 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw, cpu_stream->channels_min); hw->channels_max = min(codec_stream->channels_max, cpu_stream->channels_max); - hw->formats = codec_stream->formats & cpu_stream->formats; + if (hw->formats) + hw->formats &= codec_stream->formats & cpu_stream->formats; + else + hw->formats = codec_stream->formats & cpu_stream->formats; hw->rates = codec_stream->rates & cpu_stream->rates; if (codec_stream->rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) -- cgit v0.10.2 From 817873f4b155b22a24c48d6a38ee32007e2d856e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Jan 2014 10:24:40 +0100 Subject: ASoC: pcm: Properly initialize hw->rate_max If none of the components (CODEC or CPU DAI) sets a maximum sample rate we'll end up with the rate_max field of the runtime hardware set to 0. (Note that it is still possible for the components to constrain the supported sample rates using other methods, e.g. setting a list constraint) If rate_max is 0 this means that the sound card doesn't support any rates at all, which is not the desired result. So initialize rate_max to UINT_MAX. For symmetry reasons also set rate_min to 0. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1a617fd..2b89496 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -170,6 +170,9 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) hw->rates |= codec_stream->rates; + hw->rate_min = 0; + hw->rate_max = UINT_MAX; + snd_pcm_limit_hw_rates(runtime); hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); -- cgit v0.10.2 From 24710c97960ac343c613786d250a1e0063555faa Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Jan 2014 10:24:41 +0100 Subject: ASoC: fsl: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates SNDRV_PCM_RATE_CONTINUOUS means that all rates (possibly limited to a certain interval) are supported. There is no need to manually set other rate bits. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 35e2773..dd5e6a7 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -79,8 +79,7 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) * ALSA that we support all rates and let the codec driver decide what rates * are really supported. */ -#define FSLSSI_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ - SNDRV_PCM_RATE_CONTINUOUS) +#define FSLSSI_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS /** * FSLSSI_I2S_FORMATS: audio formats supported by the SSI diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index f4efaad..5d07e8a 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -26,8 +26,7 @@ * ALSA that we support all rates and let the codec driver decide what rates * are really supported. */ -#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ - SNDRV_PCM_RATE_CONTINUOUS) +#define PSC_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS /** * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode -- cgit v0.10.2 From bf103eb4af73596edbab5faab67e29ea1e87c769 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Jan 2014 10:24:42 +0100 Subject: ASoC: s6000: Don't mix SNDRV_PCM_RATE_CONTINUOUS with specific rates MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit SNDRV_PCM_RATE_CONTINUOUS means that all rates (possibly limited to a certain interval) are supported. There is no need to manually set other rate bits. Signed-off-by: Lars-Peter Clausen Acked-by: Daniel Glöckner Signed-off-by: Mark Brown diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 73bb99f..7eba797 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -405,8 +405,7 @@ static int s6000_i2s_dai_probe(struct snd_soc_dai *dai) return 0; } -#define S6000_I2S_RATES (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_192000) +#define S6000_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS #define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops s6000_i2s_dai_ops = { -- cgit v0.10.2 From e3a9269f874067fcefc5eb8037466161fb0fe3f4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Jan 2014 10:24:43 +0100 Subject: ALSA: Add helper function for intersecting two rate masks A bit of special care is necessary when creating the intersection of two rate masks. This comes from the special meaning of the SNDRV_PCM_RATE_CONTINUOUS and SNDRV_PCM_RATE_KNOT bits, which needs special handling when intersecting two rate masks. SNDRV_PCM_RATE_CONTINUOUS means the hardware supports all rates in a specific interval. SNDRV_PCM_RATE_KNOT means the hardware supports a set of discrete rates specified by a list constraint. For all other cases the supported rates are specified directly in the rate mask. Signed-off-by: Lars-Peter Clausen Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 84b10f9..d017091 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -901,6 +901,8 @@ extern const struct snd_pcm_hw_constraint_list snd_pcm_known_rates; int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime); unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate); unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit); +unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a, + unsigned int rates_b); static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream, struct snd_dma_buffer *bufp) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 43f24cc..4560ca0 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -514,3 +514,42 @@ unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit) return 0; } EXPORT_SYMBOL(snd_pcm_rate_bit_to_rate); + +static unsigned int snd_pcm_rate_mask_sanitize(unsigned int rates) +{ + if (rates & SNDRV_PCM_RATE_CONTINUOUS) + return SNDRV_PCM_RATE_CONTINUOUS; + else if (rates & SNDRV_PCM_RATE_KNOT) + return SNDRV_PCM_RATE_KNOT; + return rates; +} + +/** + * snd_pcm_rate_mask_intersect - computes the intersection between two rate masks + * @rates_a: The first rate mask + * @rates_b: The second rate mask + * + * This function computes the rates that are supported by both rate masks passed + * to the function. It will take care of the special handling of + * SNDRV_PCM_RATE_CONTINUOUS and SNDRV_PCM_RATE_KNOT. + * + * Return: A rate mask containing the rates that are supported by both rates_a + * and rates_b. + */ +unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a, + unsigned int rates_b) +{ + rates_a = snd_pcm_rate_mask_sanitize(rates_a); + rates_b = snd_pcm_rate_mask_sanitize(rates_b); + + if (rates_a & SNDRV_PCM_RATE_CONTINUOUS) + return rates_b; + else if (rates_b & SNDRV_PCM_RATE_CONTINUOUS) + return rates_a; + else if (rates_a & SNDRV_PCM_RATE_KNOT) + return rates_b; + else if (rates_b & SNDRV_PCM_RATE_KNOT) + return rates_a; + return rates_a & rates_b; +} +EXPORT_SYMBOL_GPL(snd_pcm_rate_mask_intersect); -- cgit v0.10.2 From 55dcdb5051930dee75e9e2c0da90bc82ee3dcd77 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 11 Jan 2014 10:24:44 +0100 Subject: ASoC: pcm: Use snd_pcm_rate_mask_intersect() helper Instead of open-coding the intersecting of two rate masks (and getting slightly wrong for some of the corner cases) use the new snd_pcm_rate_mask_intersect() helper function. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2b89496..4bbda0a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -162,13 +162,8 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, hw->formats &= codec_stream->formats & cpu_stream->formats; else hw->formats = codec_stream->formats & cpu_stream->formats; - hw->rates = codec_stream->rates & cpu_stream->rates; - if (codec_stream->rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - hw->rates |= cpu_stream->rates; - if (cpu_stream->rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - hw->rates |= codec_stream->rates; + hw->rates = snd_pcm_rate_mask_intersect(codec_stream->rates, + cpu_stream->rates); hw->rate_min = 0; hw->rate_max = UINT_MAX; -- cgit v0.10.2