June 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: firewire: arrange common PCM info/constraints for AMDTP engine applications
by Takashi Sakamoto 08 Jun '17

08 Jun '17

In ALSA firewire stack, 8 drivers uses IEC 61883-1/6 engine for data transmission. They have common PCM info/constraints and duplicated codes. This commit unifies the codes into fireiwre-lib. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 19 +++++++++++++++++ sound/firewire/bebob/bebob_pcm.c | 21 ------------------- sound/firewire/dice/dice-pcm.c | 20 ------------------ sound/firewire/digi00x/digi00x-pcm.c | 33 ++++++++++-------------------- sound/firewire/fireface/ff-pcm.c | 21 ------------------- sound/firewire/fireworks/fireworks_pcm.c | 21 ------------------- sound/firewire/motu/motu-pcm.c | 20 ------------------ sound/firewire/oxfw/oxfw-pcm.c | 20 ------------------ sound/firewire/tascam/tascam-pcm.c | 35 ++++++-------------------------- 9 files changed, 36 insertions(+), 174 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 4316d9db404d..9678bc75dc5b 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -148,8 +148,27 @@ EXPORT_SYMBOL(amdtp_rate_table); int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, struct snd_pcm_runtime *runtime) { + struct snd_pcm_hardware *hw = &runtime->hw; int err; + hw->info = SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_JOINT_DUPLEX | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID; + + /* SNDRV_PCM_INFO_BATCH */ + hw->periods_min = 2; + hw->periods_max = UINT_MAX; + + /* bytes for a frame */ + hw->period_bytes_min = 4 * hw->channels_max; + + /* Just to prevent from allocating much pages. */ + hw->period_bytes_max = hw->period_bytes_min * 2048; + hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; + /* * Currently firewire-lib processes 16 packets in one software * interrupt callback. This equals to 2msec but actually the diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index 657e15a27e5c..e6adab3ef42e 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -92,19 +92,6 @@ limit_channels_and_rates(struct snd_pcm_hardware *hw, } } -static void -limit_period_and_buffer(struct snd_pcm_hardware *hw) -{ - hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ - hw->periods_max = UINT_MAX; - - hw->period_bytes_min = 4 * hw->channels_max; /* bytes for a frame */ - - /* Just to prevent from allocating much pages. */ - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; -} - static int pcm_init_hw_params(struct snd_bebob *bebob, struct snd_pcm_substream *substream) @@ -114,13 +101,6 @@ pcm_init_hw_params(struct snd_bebob *bebob, struct snd_bebob_stream_formation *formations; int err; - runtime->hw.info = SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS; s = &bebob->tx_stream; @@ -132,7 +112,6 @@ pcm_init_hw_params(struct snd_bebob *bebob, } limit_channels_and_rates(&runtime->hw, formations); - limit_period_and_buffer(&runtime->hw); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, formations, diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 2dda74695069..7cb9e9713ac3 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -51,18 +51,6 @@ static int limit_channels_and_rates(struct snd_dice *dice, return 0; } -static void limit_period_and_buffer(struct snd_pcm_hardware *hw) -{ - hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ - hw->periods_max = UINT_MAX; - - hw->period_bytes_min = 4 * hw->channels_max; /* byte for a frame */ - - /* Just to prevent from allocating much pages. */ - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; -} - static int init_hw_info(struct snd_dice *dice, struct snd_pcm_substream *substream) { @@ -74,13 +62,6 @@ static int init_hw_info(struct snd_dice *dice, unsigned int count, size; int err; - hw->info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_BLOCK_TRANSFER; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { hw->formats = AM824_IN_PCM_FORMAT_BITS; dir = AMDTP_IN_STREAM; @@ -107,7 +88,6 @@ static int init_hw_info(struct snd_dice *dice, substream->pcm->device, size); if (err < 0) return err; - limit_period_and_buffer(hw); return amdtp_am824_add_pcm_hw_constraints(stream, runtime); } diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index f76cf5e383db..796f4b4645f5 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -58,31 +58,11 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, static int pcm_init_hw_params(struct snd_dg00x *dg00x, struct snd_pcm_substream *substream) { - static const struct snd_pcm_hardware hardware = { - .info = SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000, - .rate_min = 44100, - .rate_max = 96000, - .channels_min = 10, - .channels_max = 18, - .period_bytes_min = 4 * 18, - .period_bytes_max = 4 * 18 * 2048, - .buffer_bytes_max = 4 * 18 * 2048 * 2, - .periods_min = 2, - .periods_max = UINT_MAX, - }; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hardware *hw = &runtime->hw; struct amdtp_stream *s; int err; - substream->runtime->hw = hardware; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; @@ -92,6 +72,15 @@ static int pcm_init_hw_params(struct snd_dg00x *dg00x, s = &dg00x->rx_stream; } + hw->channels_min = 10; + hw->channels_max = 18; + + hw->rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + snd_pcm_limit_hw_rates(runtime); + err = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, NULL, diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index ad974b5a2561..d12a0e3a4219 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -91,18 +91,6 @@ static void limit_channels_and_rates(struct snd_pcm_hardware *hw, } } -static void limit_period_and_buffer(struct snd_pcm_hardware *hw) -{ - hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ - hw->periods_max = UINT_MAX; - - hw->period_bytes_min = 4 * hw->channels_max; /* bytes for a frame */ - - /* Just to prevent from allocating much pages. */ - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; -} - static int pcm_init_hw_params(struct snd_ff *ff, struct snd_pcm_substream *substream) { @@ -111,13 +99,6 @@ static int pcm_init_hw_params(struct snd_ff *ff, const unsigned int *pcm_channels; int err; - runtime->hw.info = SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; s = &ff->tx_stream; @@ -128,9 +109,7 @@ static int pcm_init_hw_params(struct snd_ff *ff, pcm_channels = ff->spec->pcm_playback_channels; } - /* limit rates */ limit_channels_and_rates(&runtime->hw, pcm_channels); - limit_period_and_buffer(&runtime->hw); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, (void *)pcm_channels, diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index 346e2647ed1f..40faed5e6968 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -129,19 +129,6 @@ limit_channels(struct snd_pcm_hardware *hw, unsigned int *pcm_channels) } } -static void -limit_period_and_buffer(struct snd_pcm_hardware *hw) -{ - hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ - hw->periods_max = UINT_MAX; - - hw->period_bytes_min = 4 * hw->channels_max; /* bytes for a frame */ - - /* Just to prevent from allocating much pages. */ - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; -} - static int pcm_init_hw_params(struct snd_efw *efw, struct snd_pcm_substream *substream) @@ -151,13 +138,6 @@ pcm_init_hw_params(struct snd_efw *efw, unsigned int *pcm_channels; int err; - runtime->hw.info = SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS; s = &efw->tx_stream; @@ -173,7 +153,6 @@ pcm_init_hw_params(struct snd_efw *efw, snd_pcm_limit_hw_rates(runtime); limit_channels(&runtime->hw, pcm_channels); - limit_period_and_buffer(&runtime->hw); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, pcm_channels, diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index e3ef89cee565..a2b50df70874 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -96,18 +96,6 @@ static void limit_channels_and_rates(struct snd_motu *motu, snd_pcm_limit_hw_rates(runtime); } -static void limit_period_and_buffer(struct snd_pcm_hardware *hw) -{ - hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ - hw->periods_max = UINT_MAX; - - hw->period_bytes_min = 4 * hw->channels_max; /* byte for a frame */ - - /* Just to prevent from allocating much pages. */ - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; -} - static int init_hw_info(struct snd_motu *motu, struct snd_pcm_substream *substream) { @@ -117,13 +105,6 @@ static int init_hw_info(struct snd_motu *motu, struct snd_motu_packet_format *formats; int err; - hw->info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_BLOCK_TRANSFER; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { hw->formats = SNDRV_PCM_FMTBIT_S32; stream = &motu->tx_stream; @@ -135,7 +116,6 @@ static int init_hw_info(struct snd_motu *motu, } limit_channels_and_rates(motu, runtime, formats); - limit_period_and_buffer(hw); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, motu_rate_constraint, formats, diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index bc1a3a36ab06..3dd46285c0e2 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -106,18 +106,6 @@ static void limit_channels_and_rates(struct snd_pcm_hardware *hw, u8 **formats) } } -static void limit_period_and_buffer(struct snd_pcm_hardware *hw) -{ - hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ - hw->periods_max = UINT_MAX; - - hw->period_bytes_min = 4 * hw->channels_max; /* bytes for a frame */ - - /* Just to prevent from allocating much pages. */ - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; -} - static int init_hw_params(struct snd_oxfw *oxfw, struct snd_pcm_substream *substream) { @@ -126,13 +114,6 @@ static int init_hw_params(struct snd_oxfw *oxfw, struct amdtp_stream *stream; int err; - runtime->hw.info = SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS; stream = &oxfw->tx_stream; @@ -144,7 +125,6 @@ static int init_hw_params(struct snd_oxfw *oxfw, } limit_channels_and_rates(&runtime->hw, formats); - limit_period_and_buffer(&runtime->hw); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, formats, diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index 3c4482aa7231..6ec8ec634d4d 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -8,41 +8,14 @@ #include "tascam.h" -static void set_buffer_params(struct snd_pcm_hardware *hw) -{ - hw->period_bytes_min = 4 * hw->channels_min; - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * 2; - - hw->periods_min = 2; - hw->periods_max = UINT_MAX; -} - static int pcm_init_hw_params(struct snd_tscm *tscm, struct snd_pcm_substream *substream) { - static const struct snd_pcm_hardware hardware = { - .info = SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000, - .rate_min = 44100, - .rate_max = 96000, - .channels_min = 10, - .channels_max = 18, - }; struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hardware *hw = &runtime->hw; struct amdtp_stream *stream; unsigned int pcm_channels; - runtime->hw = hardware; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; stream = &tscm->tx_stream; @@ -59,7 +32,11 @@ static int pcm_init_hw_params(struct snd_tscm *tscm, pcm_channels += 2; runtime->hw.channels_min = runtime->hw.channels_max = pcm_channels; - set_buffer_params(&runtime->hw); + hw->rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + snd_pcm_limit_hw_rates(runtime); return amdtp_tscm_add_pcm_hw_constraints(stream, runtime); } -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs.
by Liam Girdwood 08 Jun '17

08 Jun '17

Add some DAPM widget types to better support the construction of DAPM graphs within DSPs. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dapm.h | 6 ++++++ include/uapi/sound/asoc.h | 9 ++++++++- sound/soc/soc-topology.c | 7 +++++++ 3 files changed, 21 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a466f4bdc835..f142dace1c14 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -510,6 +510,12 @@ enum snd_soc_dapm_type { snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ + snd_soc_dapm_buffer, /* DSP/CODEC internal buffer */ + snd_soc_dapm_pipeline, /* DSP/CODEC internal pipeline */ + snd_soc_dapm_effect, /* DSP/CODEC effect component */ + snd_soc_dapm_src, /* DSP/CODEC SRC component */ + snd_soc_dapm_asrc, /* DSP/CODEC ASRC component */ + snd_soc_dapm_codec, /* FW/SW coder/decoder component */ }; enum snd_soc_dapm_subclass { diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 6702533c8bd8..b8993b3c55c6 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -73,7 +73,14 @@ #define SND_SOC_TPLG_DAPM_DAI_IN 13 #define SND_SOC_TPLG_DAPM_DAI_OUT 14 #define SND_SOC_TPLG_DAPM_DAI_LINK 15 -#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK +#define SND_SOC_TPLG_DAPM_BUFFER 16 +#define SND_SOC_TPLG_DAPM_PIPELINE 17 +#define SND_SOC_TPLG_DAPM_EFFECT 18 +#define SND_SOC_TPLG_DAPM_SIGGEN 19 +#define SND_SOC_TPLG_DAPM_SRC 20 +#define SND_SOC_TPLG_DAPM_ASRC 21 +#define SND_SOC_TPLG_DAPM_CODEC 22 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_CODEC /* Header magic number and string sizes */ #define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index e10dc353621a..468e74640570 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -242,6 +242,13 @@ static const struct soc_tplg_map dapm_map[] = { {SND_SOC_TPLG_DAPM_DAI_IN, snd_soc_dapm_dai_in}, {SND_SOC_TPLG_DAPM_DAI_OUT, snd_soc_dapm_dai_out}, {SND_SOC_TPLG_DAPM_DAI_LINK, snd_soc_dapm_dai_link}, + {SND_SOC_TPLG_DAPM_BUFFER, snd_soc_dapm_buffer}, + {SND_SOC_TPLG_DAPM_PIPELINE, snd_soc_dapm_pipeline}, + {SND_SOC_TPLG_DAPM_EFFECT, snd_soc_dapm_effect}, + {SND_SOC_TPLG_DAPM_SIGGEN, snd_soc_dapm_siggen}, + {SND_SOC_TPLG_DAPM_SRC, snd_soc_dapm_src}, + {SND_SOC_TPLG_DAPM_ASRC, snd_soc_dapm_asrc}, + {SND_SOC_TPLG_DAPM_CODEC, snd_soc_dapm_codec}, }; static int tplc_chan_get_reg(struct soc_tplg *tplg, -- 2.11.0

3 3

[alsa-devel] [PATCH] ASoC: topology: Fix build by declaring snd_kcontrol_new and soc_dai_link.
by Liam Girdwood 07 Jun '17

07 Jun '17

Both are used within this header so need to be declared. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-topology.h | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index f9cc7b9271ac..7ea2794e1c6a 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -28,6 +28,8 @@ struct snd_soc_component; struct snd_soc_tplg_pcm_fe; struct snd_soc_dapm_context; struct snd_soc_card; +struct snd_kcontrol_new; +struct snd_soc_dai_link; /* object scan be loaded and unloaded in groups with identfying indexes */ #define SND_SOC_TPLG_INDEX_ALL 0 /* ID that matches all FW objects */ @@ -116,12 +118,16 @@ struct snd_soc_tplg_ops { int (*widget_load)(struct snd_soc_component *, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); + int (*widget_ready)(struct snd_soc_component *, + struct snd_soc_dapm_widget *, + struct snd_soc_tplg_dapm_widget *); int (*widget_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* FE DAI - used for any driver specific init */ int (*dai_load)(struct snd_soc_component *, - struct snd_soc_dai_driver *dai_drv); + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); int (*dai_unload)(struct snd_soc_component *, struct snd_soc_dobj *); -- 2.11.0

2 7

[alsa-devel] [PATCH v2] ASoC: topology: Fix build by declaring snd_kcontrol_new and soc_dai_link.
by Liam Girdwood 07 Jun '17

07 Jun '17

Both are used within this header so need to be declared. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-topology.h | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index f9cc7b9271ac..7ea2794e1c6a 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -28,6 +28,8 @@ struct snd_soc_component; struct snd_soc_tplg_pcm_fe; struct snd_soc_dapm_context; struct snd_soc_card; +struct snd_kcontrol_new; +struct snd_soc_dai_link; /* object scan be loaded and unloaded in groups with identfying indexes */ #define SND_SOC_TPLG_INDEX_ALL 0 /* ID that matches all FW objects */ -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: hdmi-codec: remove unused ratec
by Kuninori Morimoto 07 Jun '17

07 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> struct snd_pcm_hw_constraint_list ratec is not used. Let's remove it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/hdmi-codec.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 6d05161..22ed0dc 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -282,7 +282,6 @@ struct hdmi_codec_priv { struct hdmi_codec_daifmt daifmt[2]; struct mutex current_stream_lock; struct snd_pcm_substream *current_stream; - struct snd_pcm_hw_constraint_list ratec; uint8_t eld[MAX_ELD_BYTES]; struct snd_pcm_chmap *chmap_info; unsigned int chmap_idx; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: ak4613: Improve counting DAI number
by Ryo Kodama 07 Jun '17

07 Jun '17

Add the startup function to count DAI instead of hw_params. This change matches the number of opened DAIs. If this change isn't applied, you may get unexpected error due to mismatching of count. Since the excution number of hw_params and shutdown may be different, the mismatching happens. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Ryo Kodama <ryo.kodama.vz(a)renesas.com> --- sound/soc/codecs/ak4613.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 9d2be7c..512a1e7 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -254,6 +254,17 @@ static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, mutex_unlock(&priv->lock); } +static int ak4613_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + priv->cnt++; + + return 0; +} + static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; @@ -358,7 +369,6 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, if ((priv->iface == NULL) || (priv->iface == iface)) { priv->iface = iface; - priv->cnt++; ret = 0; } mutex_unlock(&priv->lock); @@ -407,6 +417,7 @@ static int ak4613_set_bias_level(struct snd_soc_codec *codec, } static const struct snd_soc_dai_ops ak4613_dai_ops = { + .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, .set_fmt = ak4613_dai_set_fmt, .hw_params = ak4613_dai_hw_params, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: simple-card: remove duplicate parameter from asoc_simple_card_parse_of()
by Kuninori Morimoto 07 Jun '17

07 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 9453232..ca2b1a7 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -341,12 +341,12 @@ static int asoc_simple_card_parse_aux_devs(struct device_node *node, return 0; } -static int asoc_simple_card_parse_of(struct device_node *node, - struct simple_card_data *priv) +static int asoc_simple_card_parse_of(struct simple_card_data *priv) { struct device *dev = simple_priv_to_dev(priv); struct snd_soc_card *card = simple_priv_to_card(priv); struct device_node *dai_link; + struct device_node *node = dev->of_node; int ret; if (!node) @@ -445,7 +445,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (np && of_device_is_available(np)) { - ret = asoc_simple_card_parse_of(np, priv); + ret = asoc_simple_card_parse_of(priv); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); -- 1.9.1

2 1

[alsa-devel] [PATCH 2/2] ASoC: simple-scu-card: remove duplicate paramera from asoc_simple_card_parse_of
by Kuninori Morimoto 07 Jun '17

07 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-scu-card.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index a6a35b0..44da691 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -177,13 +177,13 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, return 0; } -static int asoc_simple_card_parse_of(struct device_node *node, - struct simple_card_data *priv) +static int asoc_simple_card_parse_of(struct simple_card_data *priv) { struct device *dev = simple_priv_to_dev(priv); struct device_node *np; struct snd_soc_card *card = simple_priv_to_card(priv); + struct device_node *node = dev->of_node; unsigned int daifmt = 0; bool is_fe; int ret, i; @@ -259,7 +259,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) card->codec_conf = &priv->codec_conf; card->num_configs = 1; - ret = asoc_simple_card_parse_of(np, priv); + ret = asoc_simple_card_parse_of(priv); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); -- 1.9.1

3 2

[alsa-devel] [PATCH 0/2] ASoC: rsnd: DMA pointer accurate for PulseAudio
by Kuninori Morimoto 07 Jun '17

07 Jun '17

Hi Mark This patch adjust to PulseAudio timer mode on Renesas sound driver. Kuninori Morimoto (2): ASoC: rsnd: update pointer more accurate ASoC: rsnd: cleanup pointer related code sound/soc/sh/rcar/core.c | 48 +++--------------------------- sound/soc/sh/rcar/dma.c | 32 ++++++++++++++++---- sound/soc/sh/rcar/rsnd.h | 13 ++++---- sound/soc/sh/rcar/ssi.c | 77 ++++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 112 insertions(+), 58 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH 0/3] ALSA: pcm: add tracepoints for PCM params refinement
by Takashi Sakamoto 07 Jun '17

07 Jun '17

Hi, This patchset is a part of my previous RFC and go for upstream. [alsa-devel] [PATCH RFC 00/21] ALSA: pcm: add tracepoints for PCM params operation http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120548.html In ALSA PCM interface, applications can get hardware capability by ioctl(2) with SNDRV_PCM_IOCTL_HW_REFINE/SNDRV_PCM_IOCT_HW_PARAMS in a shape of 'struct snd_pcm_hw_params'. In kernel side, relevant processing is somewhat complicated and developers sometimes have hard time to debug drivers for PCM constraints and rules. This patchset adds tracepoints to assist their work. Changes: - change print format - rename events - refactoring in the RFC will be done in my later work. This patchset adds two events; 'hw_interval_param' and 'hw_mask_param' in 'snd_pcm' subsystem. When these events are probed, tracer outputs below lines: hw_interval_param: 0,0,0,0 000/023 BUFFER_SIZE 0 0 [0 4294967295] 0 1 [0 4294967295] hw_interval_param: 0,0,0,0 000/023 BUFFER_BYTES 0 0 [0 4294967295] 0 1 [128 65536] hw_interval_param: 0,0,0,0 000/023 TICK_TIME 0 0 [0 4294967295] 0 0 [0 4294967295] hw_mask_param: 0,0,0,0 001/023 FORMAT 00000000000000000000001000000044 00000000000000000000001000000044 hw_interval_param: 0,0,0,0 002/023 SAMPLE_BITS 0 1 [0 4294967295] 0 1 [16 32] hw_interval_param: 0,0,0,0 003/023 SAMPLE_BITS 0 1 [16 32] 0 1 [16 32] The first field represents PCM character device and subdevice for probed runtime of PCM substream. In the above case, '/dev/snd/pcmC0D0p' and first subdevice are used for runtime of PCM substream. The second field represents id of each rule applied to the runtime, and the total number of rules added to the runtime. Basically, when the rule is applied to the runtime, these events are probed, but lines with id 0 are exceptions. They're application of constraints to the runtime. The third field is name of parameter in the runtime. The rest fields depends on type of the parameter (mask/interval). In ALSA PCM core, runtimes get 22 (or 21) rules as a default. In a below sample, the 21st rule is added by driver (snd-soc-imx-ssi.ko). The rest is added by snd-pcm.ko. In detail, please see 'snd_pcm_hw_constraints_init()' and 'snd_pcm_hw_constraints_complete()' in 'sound/core/pcm_native.c'. hw_interval_param: 0,0,0,0 019/023 PERIOD_TIME 0 0 [0 4294967295] 0 0 (166 4095875] hw_interval_param: 0,0,0,0 020/023 BUFFER_TIME 0 0 [0 4294967295] 0 0 (333 4096000] hw_interval_param: 0,0,0,0 021/023 PERIOD_SIZE 0 1 [16 32767] 0 1 [16 32766] hw_interval_param: 0,0,0,0 022/023 BUFFER_BYTES 0 1 [128 65536] 0 1 [128 65536] hw_interval_param: 0,0,0,0 023/023 RATE 0 0 [8000 96000] 0 0 [8000 96000] hw_mask_param: 0,0,0,0 001/023 FORMAT 00000000000000000000001000000044 00000000000000000000001000000044 As I noted in second patch, this patchset increases size of relocatable object. In general, kernel configuration for tracepoints should govern whether the features are enbled/disabled. However, for convenience to distributor, this patchset utilizes CONFIG_SND_DEBUG to switch the feature is enabled/disabled. $ size sound/core/pcm_native.o.before text data bss dec hex filename 28951 40 8 28999 7147 sound/core/pcm_native.o.before $ size sound/core/pcm_native.o.after text data bss dec hex filename 34942 2264 8 37214 915e sound/core/pcm_native.o.after Takashi Sakamoto (3): ALSA: pcm: tracepoints for refining PCM parameters ALSA: pcm: enable parameter tracepoints only when CONFIG_SND_DEBUG is enabled ALSA: pcm: obsolete RULES_DEBUG local macro sound/core/Makefile | 1 + sound/core/pcm_native.c | 125 ++++++++++++++----------------------- sound/core/pcm_param_trace.h | 142 +++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 188 insertions(+), 80 deletions(-) create mode 100644 sound/core/pcm_param_trace.h -- 2.11.0

2 8