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

[alsa-devel] [PATCH] ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs.
by Liam Girdwood 07 Jun '17

07 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

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[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

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[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

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[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

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[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

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[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

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[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

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[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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