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

[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

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[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

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[alsa-devel] [PATCH v10] sun4i-codec: Add Line-In, FM-In, Mic 2, Capture Source, Differential Line-In
by Danny Milosavljevic 07 Jun '17

07 Jun '17

This patch adds support for some mixer controls to sun4i-codec: - Line-In - FM-In - Mic 2 - Capture Source - Differential Line-In It also adds a mux for the capture source and the PGA for the MIC2 preamp. Where possible, it uses SOC_DAPM_DOUBLE in order to cut down on the number of distinct controls in alsamixer. v10 changes compared to v9 are: - Use SOC_DAPM_DOUBLE where possible and it makes sense in order to cut down on the number of controls. v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (1): sun4i-codec: Add mixer controls: Line-In, FM-In, Mic 2, Capture Source, Differential Line-In. sound/soc/sunxi/sun4i-codec.c | 296 ++++++++++++++++++++++++++++++++++++++---- 1 file changed, 269 insertions(+), 27 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH 0/2] ASoC: Intel: Add Kabylake machine driver for RT5663+RT5514+MAX98927
by Harsha Priya 07 Jun '17

07 Jun '17

This patch set adds a machine driver for Kabylake platform which has RT5663, RT5514 and MAX9827 codecs. It also adds the quirk data for the platform driver to load the correct machine driver. For this particular machine, * the RT5663 is a headset codec connected on SSP0 * the RT5514 is a dmic codec connected on SSP1 * the MAX98987 is a speaker codec connected on SSP1 Harsha Priya (2): ASoC: Intel: Add Kabylake machine driver for RT5514, RT5663 and MAX98927 ASoC: Intel: Add Kabylake RT5663+RT5514+MAX98927 machine driver entry .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 640 +++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 29 +- 2 files changed, 665 insertions(+), 4 deletions(-) create mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c -- 1.9.1

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[alsa-devel] [PATCH v2][RESEND v2] ASoC: rsnd: control kctrl items acceptance anytime/runtime
by Kuninori Morimoto 07 Jun '17

07 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current SRC/DVC/CTU adds kctrl for each device, and SRC can adjust its sampling rate during playback, thus, this feature should be enabled only *during* playback. This patch controls it more clearly Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- - resend v2 sound/soc/sh/rcar/core.c | 17 +++++++++++++++++ sound/soc/sh/rcar/ctu.c | 6 ++++++ sound/soc/sh/rcar/dvc.c | 5 +++++ sound/soc/sh/rcar/rsnd.h | 16 ++++++++++------ sound/soc/sh/rcar/src.c | 2 ++ 5 files changed, 40 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1744015..d3c9078 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -982,6 +982,9 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); int i, change = 0; + if (!cfg->accept(cfg->io)) + return 0; + for (i = 0; i < cfg->size; i++) { if (cfg->texts) { change |= (uc->value.enumerated.item[i] != cfg->val[i]); @@ -998,6 +1001,18 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, return change; } +int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io) +{ + return 1; +} + +int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io) +{ + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + + return !!runtime; +} + struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg) { cfg->cfg.val = cfg->val; @@ -1016,6 +1031,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, const unsigned char *name, + int (*accept)(struct rsnd_dai_stream *io), void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod), struct rsnd_kctrl_cfg *cfg, @@ -1050,6 +1066,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, cfg->texts = texts; cfg->max = max; cfg->size = size; + cfg->accept = accept; cfg->update = update; cfg->card = card; cfg->kctrl = kctrl; diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 9dcc1f9..4ba8f2f 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -279,12 +279,14 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* CTU Pass */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU Pass", + rsnd_kctrl_accept_anytime, NULL, &ctu->pass, RSND_MAX_CHANNELS, 0xC); /* ROW0 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV0", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv0, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -293,6 +295,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW1 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV1", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv1, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -301,6 +304,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW2 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV2", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv2, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -309,6 +313,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW3 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV3", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv3, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -317,6 +322,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* Reset */ ret = rsnd_kctrl_new_s(mod, io, rtd, "CTU Reset", + rsnd_kctrl_accept_anytime, rsnd_ctu_value_reset, &ctu->reset, 1); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 463de83..75af6e7 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -257,6 +257,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? "DVC Out Playback Volume" : "DVC In Capture Volume", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->volume, slots, 0x00800000 - 1); @@ -267,6 +268,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? "DVC Out Mute Switch" : "DVC In Mute Switch", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->mute, slots, 1); @@ -277,6 +279,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_s(mod, io, rtd, is_play ? "DVC Out Ramp Switch" : "DVC In Ramp Switch", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->ren, 1); if (ret < 0) @@ -285,6 +288,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_e(mod, io, rtd, is_play ? "DVC Out Ramp Up Rate" : "DVC In Ramp Up Rate", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->rup, dvc_ramp_rate); @@ -294,6 +298,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_e(mod, io, rtd, is_play ? "DVC Out Ramp Down Rate" : "DVC In Ramp Down Rate", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->rdown, dvc_ramp_rate); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index dbf4163..f68aecb 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -594,6 +594,7 @@ struct rsnd_kctrl_cfg { unsigned int size; u32 *val; const char * const *texts; + int (*accept)(struct rsnd_dai_stream *io); void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod); struct rsnd_dai_stream *io; struct snd_card *card; @@ -611,12 +612,15 @@ struct rsnd_kctrl_cfg_s { u32 val; }; +int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io); +int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io); struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg); struct rsnd_kctrl_cfg *rsnd_kctrl_init_s(struct rsnd_kctrl_cfg_s *cfg); int rsnd_kctrl_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, const unsigned char *name, + int (*accept)(struct rsnd_dai_stream *io), void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod), struct rsnd_kctrl_cfg *cfg, @@ -624,16 +628,16 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, int size, u32 max); -#define rsnd_kctrl_new_m(mod, io, rtd, name, update, cfg, size, max) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_m(cfg), \ +#define rsnd_kctrl_new_m(mod, io, rtd, name, accept, update, cfg, size, max) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_m(cfg), \ NULL, size, max) -#define rsnd_kctrl_new_s(mod, io, rtd, name, update, cfg, max) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_s(cfg), \ +#define rsnd_kctrl_new_s(mod, io, rtd, name, accept, update, cfg, max) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_s(cfg), \ NULL, 1, max) -#define rsnd_kctrl_new_e(mod, io, rtd, name, update, cfg, texts) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_s(cfg), \ +#define rsnd_kctrl_new_e(mod, io, rtd, name, accept, update, cfg, texts) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_s(cfg), \ texts, 1, ARRAY_SIZE(texts)) /* diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index e4bde0c..e395fcb 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -489,6 +489,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate Switch" : "SRC In Rate Switch", + rsnd_kctrl_accept_anytime, rsnd_src_set_convert_rate, &src->sen, 1); if (ret < 0) @@ -498,6 +499,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate" : "SRC In Rate", + rsnd_kctrl_accept_runtime, rsnd_src_set_convert_rate, &src->sync, 192000); -- 1.9.1

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[alsa-devel] How to handle stream rates if CPU supports many channels ?
by Kuninori Morimoto 07 Jun '17

07 Jun '17

Hi ALSA ML I want to ask you about struct snd_soc_pcm_stream::rates which indicates available sampling rates (= SNDRV_PCM_RATE_xxxx). Basic use case of my CPU driver is 2ch output, but, it can handle 6ch, 8ch, 16ch sounds. And it can be clock master. In this case, for example, if CPU has 12288000 system clock and it uses 32bit for each 1ch data (it has divider inside). Max 2ch sound Hz = 12288000 / 32 / 2 = 192000Hz Max 8ch sound Hz = 12288000 / 32 / 8 = 48000Hz If my understanding was correct, struct snd_soc_pcm_stream::rates is needed when sound device "open" timing (= struct snd_soc_dai_ops::startup), but, we can receive used channels number when "hw_params" timing ? My question is that if system has 12288000, and if it can support both 2ch and 8ch, what struct snd_soc_pcm_stream::rates should have ?? SNDRV_PCM_RATE_8000_192000 ? SNDRV_PCM_RATE_8000_48000 ? Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: rsnd: remove unsed rsnd_ssi_non_ops
by Kuninori Morimoto 06 Jun '17

06 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_ssi_non_ops is never used. Let's remove it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3e61376..a524862 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -804,13 +804,6 @@ int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod) /* - * Non SSI - */ -static struct rsnd_mod_ops rsnd_ssi_non_ops = { - .name = SSI_NAME, -}; - -/* * ssi mod function */ static void rsnd_ssi_connect(struct rsnd_mod *mod, @@ -1010,7 +1003,6 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) goto rsnd_ssi_probe_done; } - ops = &rsnd_ssi_non_ops; if (of_property_read_bool(np, "pio-transfer")) ops = &rsnd_ssi_pio_ops; else -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: remove unused rsnd_dai_path_info
by Kuninori Morimoto 06 Jun '17

06 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 2ea2cc86db7c ("ASoC: rsnd: remove struct rcar_snd_info") removed all struct rsnd_dai_path_info related code. This patch removes unused rsnd_dai_path_info. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsnd.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index cc929fc..ec82888 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -423,7 +423,6 @@ struct rsnd_dai_stream { char name[RSND_DAI_NAME_SIZE]; struct snd_pcm_substream *substream; struct rsnd_mod *mod[RSND_MOD_MAX]; - struct rsnd_dai_path_info *info; /* rcar_snd.h */ struct rsnd_dai *rdai; u32 parent_ssi_status; int byte_pos; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: simple-scu-card: tidyup "Sampling Rate Conversion"
by Kuninori Morimoto 06 Jun '17

06 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "Sampling Rate Conversion" is better than "Sampling Rate Convert" Reported-by: James Cameron <quozl(a)laptop.org> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/simple-scu-card.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/simple-scu-card.txt b/Documentation/devicetree/bindings/sound/simple-scu-card.txt index e894cef..327d229 100644 --- a/Documentation/devicetree/bindings/sound/simple-scu-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-scu-card.txt @@ -38,7 +38,7 @@ Optional CPU/CODEC subnodes properties: - clocks / system-clock-frequency : see simple-audio-card.txt -Example 1. Sampling Rate Covert +Example 1. Sampling Rate Conversion sound { compatible = "simple-scu-audio-card"; -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: audio-graph-scu-card: add missing MIX binding example
by Kuninori Morimoto 06 Jun '17

06 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - "Sampling Rate Convert" -> "Sampling Rate Conversion" .../bindings/sound/audio-graph-scu-card.txt | 47 +++++++++++++++++++++- 1 file changed, 46 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt index b2dd23f..b63c559 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt +++ b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt @@ -35,7 +35,7 @@ Required properties: - compatible : "audio-graph-scu-card"; - dais : list of CPU DAI port{s} -Example +Example 1. Sampling Rate Conversion sound_card { compatible = "audio-graph-scu-card"; @@ -70,3 +70,48 @@ Example }; }; }; + +Example 2. 2 CPU 1 Codec (Mixing) + + sound_card { + compatible = "audio-graph-scu-card"; + + label = "sound-card"; + prefix = "codec"; + routing = "codec Playback", "DAI0 Playback", + "codec Playback", "DAI1 Playback"; + convert-rate = <48000>; + + dais = <&cpu_port0 + &cpu_port1>; + }; + + audio-codec { + ... + + port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint0>; + }; + }; + }; + + dai-controller { + ... + ports { + cpu_port0: port { + cpu_endpoint0: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + ... + }; + }; + cpu_port1: port { + cpu_endpoint1: endpoint { + dai-format = "left_j"; + ... + }; + }; + }; + }; -- 1.9.1

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