January 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: samsung: Replace depends on REGMAP_I2C with depends on I2C
by Lars-Peter Clausen 16 Jan '15

16 Jan '15

regmap is a library function that gets selected by drivers that need it. No driver modules should depend on it, since this can create dependency loops. E.g. if machine driver depends on REGMAP_I2C and selects the CODEC driver and the CODEC driver selects REGMAP_I2C, then the machine driver selects by extension one of its dependencies. The proper way to handle this is for machine drivers to depend on the same dependencies as the CODEC driver they select. In this case it is I2C. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index fc67f97..e817a2f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -54,7 +54,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) - depends on REGMAP_I2C + depends on I2C select SND_SOC_WM8580 select SND_SAMSUNG_I2S help @@ -167,7 +167,7 @@ config SND_SOC_SAMSUNG_SMDK_SPDIF config SND_SOC_SMDK_WM8580_PCM tristate "SoC PCM Audio support for WM8580 on SMDK" depends on SND_SOC_SAMSUNG && (MACH_SMDKV210 || MACH_SMDKC110) - depends on REGMAP_I2C + depends on I2C select SND_SOC_WM8580 select SND_SAMSUNG_PCM help -- 1.8.0

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[alsa-devel] [PATCH 1/2] ASoC: jive: Automatically disconnect non-connected pins
by Lars-Peter Clausen 16 Jan '15

16 Jan '15

All DAPM input and output pins of the wm8750 are either used in the card's DAPM routing table or are marked as not connected. Set the fully_routed flag of the card instead of manually marking the unused inputs and outputs as not connected. This makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/jive_wm8750.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 6c3b359..7fcb51f 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -83,22 +83,6 @@ static struct snd_soc_ops jive_ops = { .hw_params = jive_hw_params, }; -static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* These endpoints are not being used. */ - snd_soc_dapm_nc_pin(dapm, "LINPUT2"); - snd_soc_dapm_nc_pin(dapm, "RINPUT2"); - snd_soc_dapm_nc_pin(dapm, "LINPUT3"); - snd_soc_dapm_nc_pin(dapm, "RINPUT3"); - snd_soc_dapm_nc_pin(dapm, "OUT3"); - snd_soc_dapm_nc_pin(dapm, "MONO"); - - return 0; -} - static struct snd_soc_dai_link jive_dai = { .name = "wm8750", .stream_name = "WM8750", @@ -106,7 +90,6 @@ static struct snd_soc_dai_link jive_dai = { .codec_dai_name = "wm8750-hifi", .platform_name = "s3c2412-i2s", .codec_name = "wm8750.0-001a", - .init = jive_wm8750_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ops = &jive_ops, @@ -123,6 +106,7 @@ static struct snd_soc_card snd_soc_machine_jive = { .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), + .fully_routed = true, }; static struct platform_device *jive_snd_device; -- 1.8.0

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[alsa-devel] [PATCH v8] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 16 Jan '15

16 Jan '15

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 155 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 157 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 5c0a798..a2098e4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -531,6 +532,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index edd4a0a..7556885 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -940,6 +940,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ededb97..893a6e5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1230,7 +1230,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1242,7 +1243,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index bc2d7a0..286413b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -853,6 +853,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* create control for links with > 1 config */ + if (w->num_params <= 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* We implement power down on suspend by checking the power state of * the ALSA card - when we are suspending the ALSA state for the card * is set to D3. @@ -2724,6 +2754,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3198,7 +3231,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, { struct snd_soc_dapm_path *source_p, *sink_p; struct snd_soc_dai *source, *sink; - const struct snd_soc_pcm_stream *config = w->params; + const struct snd_soc_pcm_stream *config = w->params + w->params_select; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; u64 fmt; @@ -3290,8 +3323,39 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + /* Can't change the config when widget is already powered */ + if (w->power) + return -EBUSY; + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3299,14 +3363,52 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) { + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + } else { + w_param_text[count] = devm_kmemdup(card->dev, + config->stream_name, + strlen(config->stream_name) + 1, + GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3314,6 +3416,30 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + /* duplicate w_param_enum on heap so that memory persists */ + private_value = + (unsigned long) devm_kmemdup(card->dev, + (void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + /* duplicate kcontrol_dai_link on heap so that memory persists */ + template.kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3321,15 +3447,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + devm_kfree(card->dev, w); +outfree_kcontrol_news: + devm_kfree(card->dev, (void *)template.kcontrol_news); +outfree_private_value: + devm_kfree(card->dev, (void *)private_value); +outfree_link_name: + devm_kfree(card->dev, link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

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[alsa-devel] [PATCH] soc: intel: sst-haswell-ipc: Remove some unused functions
by Rickard Strandqvist 16 Jan '15

16 Jan '15

Removes some functions that are not used anywhere: sst_hsw_stream_unmute() sst_hsw_stream_mute() msg_set_stage_type() sst_hsw_dx_get_state() sst_hsw_dx_set_state() sst_hsw_stream_set_write_position() sst_hsw_stream_get_vol_reg() sst_hsw_stream_get_peak_reg() sst_hsw_stream_get_pointer_reg() sst_hsw_stream_get_read_reg() sst_hsw_stream_get_mixer_id() sst_hsw_stream_get_hw_id() sst_hsw_mixer_set_volume_curve() sst_hsw_mixer_unmute() sst_hsw_mixer_mute() sst_hsw_stream_set_volume_curve() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/soc/intel/sst-haswell-ipc.c | 226 ------------------------------------- sound/soc/intel/sst-haswell-ipc.h | 34 ------ 2 files changed, 260 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index b629151..26f856c 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -333,12 +333,6 @@ static inline u32 msg_get_stage_type(u32 msg) return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; } -static inline u32 msg_set_stage_type(u32 msg, u32 type) -{ - return (msg & ~IPC_STG_TYPE_MASK) + - (type << IPC_STG_TYPE_SHIFT); -} - static inline u32 msg_get_stream_id(u32 msg) { return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT; @@ -965,45 +959,6 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, } /* Mixer Controls */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel, - &stream->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - stream->mute[channel] = 1; - return 0; -} - -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) - -{ - int ret; - - stream->mute[channel] = 0; - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, - stream->mute_volume[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - return 0; -} - int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume) { @@ -1017,17 +972,6 @@ int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream return 0; } -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - stream->vol_req.curve_duration = curve_duration; - stream->vol_req.curve_type = curve; - - return 0; -} - /* stream volume */ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume) @@ -1065,42 +1009,6 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, return 0; } -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel, - &hsw->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 1; - return 0; -} - -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, - hsw->mixer_info.volume_register_address[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 0; - return 0; -} - int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume) { @@ -1114,16 +1022,6 @@ int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, return 0; } -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - hsw->curve_duration = curve_duration; - hsw->curve_type = curve; - - return 0; -} - /* global mixer volume */ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume) @@ -1425,50 +1323,6 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) return 0; } -/* Stream Information - these calls could be inline but we want the IPC - ABI to be opaque to client PCM drivers to cope with any future ABI changes */ -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.stream_hw_id; -} - -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.mixer_hw_id; -} - -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.read_position_register_address; -} - -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.presentation_position_register_address; -} - -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.peak_meter_register_address[channel]; -} - -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.volume_register_address[channel]; -} - int sst_hsw_mixer_get_info(struct sst_hsw *hsw) { struct sst_hsw_ipc_stream_info_reply *reply; @@ -1591,30 +1445,6 @@ u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, return ppos; } -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position) -{ - u32 header; - int ret; - - trace_stream_write_position(stream->reply.stream_hw_id, position); - - header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | - IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); - header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); - header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT); - header |= (stage_id << IPC_STG_ID_SHIFT); - stream->wpos.position = position; - - ret = ipc_tx_message_nowait(hsw, header, &stream->wpos, - sizeof(stream->wpos)); - if (ret < 0) - dev_err(hsw->dev, "error: stream %d set position %d failed\n", - stream->reply.stream_hw_id, position); - - return ret; -} - /* physical BE config */ int sst_hsw_device_set_config(struct sst_hsw *hsw, enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, @@ -1644,63 +1474,7 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, } EXPORT_SYMBOL_GPL(sst_hsw_device_set_config); -/* DX Config */ -int sst_hsw_dx_set_state(struct sst_hsw *hsw, - enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx) -{ - u32 header, state_; - int ret, item; - - header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE); - state_ = state; - - trace_ipc_request("PM enter Dx state", state); - - ret = ipc_tx_message_wait(hsw, header, &state_, sizeof(state_), - dx, sizeof(*dx)); - if (ret < 0) { - dev_err(hsw->dev, "ipc: error set dx state %d failed\n", state); - return ret; - } - - for (item = 0; item < dx->entries_no; item++) { - dev_dbg(hsw->dev, - "Item[%d] offset[%x] - size[%x] - source[%x]\n", - item, dx->mem_info[item].offset, - dx->mem_info[item].size, - dx->mem_info[item].source); - } - dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n", - dx->entries_no, state); - - memcpy(&hsw->dx, dx, sizeof(*dx)); - return 0; -} - /* Used to save state into hsw->dx_reply */ -int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, - u32 *offset, u32 *size, u32 *source) -{ - struct sst_hsw_ipc_dx_memory_item *dx_mem; - struct sst_hsw_ipc_dx_reply *dx_reply; - int entry_no; - - dx_reply = &hsw->dx; - entry_no = dx_reply->entries_no; - - trace_ipc_request("PM get Dx state", entry_no); - - if (item >= entry_no) - return -EINVAL; - - dx_mem = &dx_reply->mem_info[item]; - *offset = dx_mem->offset; - *size = dx_mem->size; - *source = dx_mem->source; - - return 0; -} - static int msg_empty_list_init(struct sst_hsw *hsw) { int i; diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 2ac194a..b6297cb 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -371,32 +371,18 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, u32 create_channel_map(enum sst_hsw_channel_config config); /* Stream Mixer Controls - */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume); int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve); - /* Global Mixer Controls - */ -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel); -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel); - int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume); int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve); - /* Stream API */ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data), @@ -436,18 +422,6 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); int sst_hsw_mixer_get_info(struct sst_hsw *hsw); /* Stream ALSA trigger operations */ @@ -462,8 +436,6 @@ int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position); u32 sst_hsw_get_dsp_position(struct sst_hsw *hsw, struct sst_hsw_stream *stream); u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, @@ -474,12 +446,6 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, enum sst_hsw_device_mode mode, u32 clock_divider); -/* DX Config */ -int sst_hsw_dx_set_state(struct sst_hsw *hsw, - enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx); -int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, - u32 *offset, u32 *size, u32 *source); - /* init */ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata); -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: omap-mcbsp: Correct CBM_CFS dai format configuration
by Peter Ujfalusi 16 Jan '15

16 Jan '15

We should select FSR also to be driven by McBSP, not only FSX. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Clearly the CBM_CFS is not widely used with McBSP, this is why it has not been noticed I believe. Peter sound/soc/omap/omap-mcbsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 8b79cafab1e2..c7eb9dd67f60 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -434,7 +434,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_CBM_CFS: /* McBSP slave. FS clock as output */ regs->srgr2 |= FSGM; - regs->pcr0 |= FSXM; + regs->pcr0 |= FSXM | FSRM; break; case SND_SOC_DAIFMT_CBM_CFM: /* McBSP slave */ -- 2.2.1

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[alsa-devel] Playback in infinitely loop - bug in alsa-jack-plugins?
by Valentin Corfu 16 Jan '15

16 Jan '15

Hello guys, Few days ago I have started to look over a new challenging issue with alsa.. The problem is that the sound entry in a loop infinitely and it seems that the last samples in the buffer are just repeated over and over again. This behaviour it happens when I have too less and only sporadically data ready to be playbacked on my embedded system configured with alsa jack pcm plugin. To solve this issue, (I don't want to use snd_pcm_drain()), I see 2 ways: 1) to clear(or overwrite with silence) the "just played" frames from the ring buffer 2) when I am in this situation, to gracefully stop/close the alsa opened port In both situations I have tried without success, _set_silence_threshold(), set_stop_threshold and other googled solutions.. More over, I also tested with "default" device and the result is as expected... the sound can be hear just once, then it's going into underrun state. I also created a small test(based on pcm_min.c example), where I intentionally put a ~1s(can be more) delay after _writei() function, to see the issue reproduced. (http://pastebin.com/4KyUG0BA) log with pcm jack plugin: http://pastebin.com/MtKQ4ASz log with "default": http://pastebin.com/G0zq6fmJ Now I'm suspecting one issue with the underrun, over alsa jack plugin... (In case I'm not wrong with the test case or my assumption..) - Isn't it implemented underrun detection over jack plugin or it is a new bug here? Best Regards, Valentin Corfu

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[alsa-devel] [PATCH v2 10/11] ASoC: tegra: Add a control for the headphone switch
by Tomeu Vizoso 15 Jan '15

15 Jan '15

To be used by userspace when the headphones jack is plugged in. Signed-off-by: Tomeu Vizoso <tomeu.vizoso(a)collabora.com> --- sound/soc/tegra/tegra_max98090.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index af3fb99..8df71a4 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -136,6 +136,7 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { }; static const struct snd_kcontrol_new tegra_max98090_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), }; -- 1.9.3

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[alsa-devel] [PATCH v2 11/11] ASoC: tegra: add sink for the internal mic to tegra_max98090
by Tomeu Vizoso 15 Jan '15

15 Jan '15

This is used in the Tegra-based Chromebooks. Signed-off-by: Tomeu Vizoso <tomeu.vizoso(a)collabora.com> --- Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + sound/soc/tegra/tegra_max98090.c | 2 ++ 2 files changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt index c949abc..c3495be 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt @@ -18,6 +18,7 @@ Required properties: * Headphones * Speakers * Mic Jack + * Int Mic - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's connected to the CODEC. diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 8df71a4..29ea87c 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -133,11 +133,13 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), }; static const struct snd_kcontrol_new tegra_max98090_controls[] = { SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Int Mic"), }; static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) -- 1.9.3

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[alsa-devel] [PATCH v4] ALSA: ASoC: soc-compress.c: fix NULL dereference
by Qais Yousef 15 Jan '15

15 Jan '15

In soc_new_compress() when rtd->dai_link->dynamic is set, we create the pcm substreams with this call: ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, 1, 0, &be_pcm); which passes 0 as capture_count leading to be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream being NULL, hence when trying to set rtd a few lines below we get an oops. Fix by using rtd->dai_link->dpcm_playback and rtd->dai_link->dpcm_capture as playback_count and capture_count to snd_pcm_new_internal(). Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- v4->v3: - the typo fix went wrong, apologies for the noise v3->v2: - fix a typo s/dynmic/dynamic/ in commit message - use if else if because either dpcm_playback or dpcm_capture should be on for compressed devices v2->v1: - use better way to fix it than just removing the line that caused the oops sound/soc/soc-compress.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 590a82f01d0b..025c38fbe3c0 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, - 1, 0, &be_pcm); + rtd->dai_link->dpcm_playback, + rtd->dai_link->dpcm_capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n", rtd->dai_link->name); @@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = be_pcm; rtd->fe_compr = 1; - be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + if (rtd->dai_link->dpcm_playback) + be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + else if (rtd->dai_link->dpcm_capture) + be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); -- 2.1.0

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[alsa-devel] [PATCH 00/17] ASoC: Remove codec field from snd_soc_dapm_widget
by Lars-Peter Clausen 15 Jan '15

15 Jan '15

This series is the last part of the codec field removal from the snd_soc_dapm_widget. One small fix for an issue that slipped through earlier and the conversion of the TI and Realtek CODEC drivers and then finally the removal itself. The reason why it is removed is because it doesn't fit to well anymore in the componentized ASoC hierarchy. And the alternative of snd_soc_dapm_to_codec(w->dapm) typically generates the same amount of code, so there is really no reason to keep it. Depends on topic/w-codec and topic/rt5677. - Lars Lars-Peter Clausen (17): ASoC: mioa701_wm9713: Fix speaker event ASoC: pcm3008: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: tlv320aic31xx: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: tlv320aic3x: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: tlv320dac33: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: twl4030: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: twl6040: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: alc5623: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: alc5632: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt286: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5631: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5640: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5645: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5651: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5670: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5677: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: Remove codec field from snd_soc_dapm_widget include/sound/soc-dapm.h | 1 - sound/soc/codecs/alc5623.c | 8 +++--- sound/soc/codecs/alc5632.c | 8 +++--- sound/soc/codecs/pcm3008.c | 4 +-- sound/soc/codecs/rt286.c | 15 ++++++----- sound/soc/codecs/rt5631.c | 28 ++++++++++++-------- sound/soc/codecs/rt5640.c | 11 ++++---- sound/soc/codecs/rt5645.c | 16 +++++++----- sound/soc/codecs/rt5651.c | 18 +++++++------ sound/soc/codecs/rt5670.c | 16 +++++++----- sound/soc/codecs/rt5677.c | 23 +++++++++-------- sound/soc/codecs/tlv320aic31xx.c | 9 ++++--- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 9 ++++--- sound/soc/codecs/twl4030.c | 55 +++++++++++++++++++++++++--------------- sound/soc/codecs/twl6040.c | 4 +-- sound/soc/pxa/mioa701_wm9713.c | 2 +- sound/soc/soc-dapm.c | 2 -- 18 files changed, 132 insertions(+), 99 deletions(-) -- 1.8.0

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