March 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/8] ASoC: pxa: Pass correct DAPM context to {corgi, poodle, spitz}_ext_control
by Lars-Peter Clausen 03 Mar '14

03 Mar '14

When calling {corgi,poodle,spitz}_ext_control() from the startup callback we pass the CODEC's DAPM context instead of the card's DAPM context. This is not a problem per se since all the DAPM functions in ext_control() fallback to widgets from other DAPM contexts, but passing the card's context is more consistent. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/pxa/corgi.c | 3 +-- sound/soc/pxa/poodle.c | 3 +-- sound/soc/pxa/spitz.c | 3 +-- 3 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index b4db3b9..5a88136 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -101,10 +101,9 @@ static void corgi_ext_control(struct snd_soc_dapm_context *dapm) static int corgi_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; /* check the jack status at stream startup */ - corgi_ext_control(&codec->dapm); + corgi_ext_control(&rtd->card->dapm); return 0; } diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 27c6c03..c6bdc6c 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -74,10 +74,9 @@ static void poodle_ext_control(struct snd_soc_dapm_context *dapm) static int poodle_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; /* check the jack status at stream startup */ - poodle_ext_control(&codec->dapm); + poodle_ext_control(&rtd->card->dapm); return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index a3a13c9..1373b01 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -111,10 +111,9 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) static int spitz_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; /* check the jack status at stream startup */ - spitz_ext_control(&codec->dapm); + spitz_ext_control(&rtd->card->dapm); return 0; } -- 1.8.0

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[alsa-devel] [PATCH 1/2] ASoC: neo1973_wm8753: Post gta01 support removal cleanup
by Lars-Peter Clausen 03 Mar '14

03 Mar '14

With GTA01 support gone from the driver there is no need to keep a separate initilization routine for the GTA02 specific bits. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/neo1973_wm8753.c | 116 ++++++++++++------------------------- 1 file changed, 37 insertions(+), 79 deletions(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 98a04c1..0125d9d 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -192,44 +192,6 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; -/* Shared routes and controls */ - -static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = { - SND_SOC_DAPM_LINE("GSM Line Out", NULL), - SND_SOC_DAPM_LINE("GSM Line In", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Handset Mic", NULL), -}; - -static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = { - /* Connections to the GSM Module */ - {"GSM Line Out", NULL, "MONO1"}, - {"GSM Line Out", NULL, "MONO2"}, - {"RXP", NULL, "GSM Line In"}, - {"RXN", NULL, "GSM Line In"}, - - /* Connections to Headset */ - {"MIC1", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Headset Mic"}, - - /* Call Mic */ - {"MIC2", NULL, "Mic Bias"}, - {"MIC2N", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Handset Mic"}, - - /* Connect the ALC pins */ - {"ACIN", NULL, "ACOP"}, -}; - -static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { - SOC_DAPM_PIN_SWITCH("GSM Line Out"), - SOC_DAPM_PIN_SWITCH("GSM Line In"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Handset Mic"), -}; - -/* GTA02 specific routes and controls */ - static int gta02_speaker_enabled; static int lm4853_set_spk(struct snd_kcontrol *kcontrol, @@ -257,7 +219,34 @@ static int lm4853_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_route neo1973_gta02_routes[] = { +static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = { + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Handset Mic", NULL), + SND_SOC_DAPM_SPK("Handset Spk", NULL), + SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), +}; + +static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = { + /* Connections to the GSM Module */ + {"GSM Line Out", NULL, "MONO1"}, + {"GSM Line Out", NULL, "MONO2"}, + {"RXP", NULL, "GSM Line In"}, + {"RXN", NULL, "GSM Line In"}, + + /* Connections to Headset */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Call Mic */ + {"MIC2", NULL, "Mic Bias"}, + {"MIC2N", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Handset Mic"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, + /* Connections to the amp */ {"Stereo Out", NULL, "LOUT1"}, {"Stereo Out", NULL, "ROUT1"}, @@ -267,7 +256,11 @@ static const struct snd_soc_dapm_route neo1973_gta02_routes[] = { {"Handset Spk", NULL, "ROUT2"}, }; -static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = { +static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { + SOC_DAPM_PIN_SWITCH("GSM Line Out"), + SOC_DAPM_PIN_SWITCH("GSM Line In"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Handset Mic"), SOC_DAPM_PIN_SWITCH("Handset Spk"), SOC_DAPM_PIN_SWITCH("Stereo Out"), @@ -276,39 +269,6 @@ static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = { lm4853_set_spk), }; -static const struct snd_soc_dapm_widget neo1973_gta02_wm8753_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Handset Spk", NULL), - SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), -}; - -static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - ret = snd_soc_dapm_new_controls(dapm, neo1973_gta02_wm8753_dapm_widgets, - ARRAY_SIZE(neo1973_gta02_wm8753_dapm_widgets)); - if (ret) - return ret; - - ret = snd_soc_dapm_add_routes(dapm, neo1973_gta02_routes, - ARRAY_SIZE(neo1973_gta02_routes)); - if (ret) - return ret; - - ret = snd_soc_add_card_controls(codec->card, neo1973_gta02_wm8753_controls, - ARRAY_SIZE(neo1973_gta02_wm8753_controls)); - if (ret) - return ret; - - snd_soc_dapm_disable_pin(dapm, "Stereo Out"); - snd_soc_dapm_disable_pin(dapm, "Handset Spk"); - snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); - snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); - - return 0; -} - static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; @@ -344,18 +304,16 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "GSM Line In"); snd_soc_dapm_disable_pin(dapm, "Headset Mic"); snd_soc_dapm_disable_pin(dapm, "Handset Mic"); + snd_soc_dapm_disable_pin(dapm, "Stereo Out"); + snd_soc_dapm_disable_pin(dapm, "Handset Spk"); /* allow audio paths from the GSM modem to run during suspend */ snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out"); snd_soc_dapm_ignore_suspend(dapm, "GSM Line In"); snd_soc_dapm_ignore_suspend(dapm, "Headset Mic"); snd_soc_dapm_ignore_suspend(dapm, "Handset Mic"); - - if (machine_is_neo1973_gta02()) { - ret = neo1973_gta02_wm8753_init(codec); - if (ret) - return ret; - } + snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); + snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); return 0; } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: pcm: free path list before exiting from error conditions
by Patrick Lai 02 Mar '14

02 Mar '14

dpcm_path_get() allocates dynamic memory to hold path list. Corresponding dpcm_get_path() must be called to free the memory. dpcm_get_path() is not called under several error conditions. This leads to memory leak. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 47e1ce7..28522bd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1989,6 +1989,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); if (paths < 0) { + dpcm_path_put(&list); dev_warn(fe->dev, "ASoC: %s no valid %s path\n", fe->dai_link->name, "playback"); mutex_unlock(&card->mutex); @@ -2018,6 +2019,7 @@ capture: paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); if (paths < 0) { + dpcm_path_put(&list); dev_warn(fe->dev, "ASoC: %s no valid %s path\n", fe->dai_link->name, "capture"); mutex_unlock(&card->mutex); @@ -2082,6 +2084,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) fe->dpcm[stream].runtime = fe_substream->runtime; if (dpcm_path_get(fe, stream, &list) <= 0) { + dpcm_path_put(&list); dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); } -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, hosted by The Linux Foundation

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[alsa-devel] [RFC 00/19] Enable platform HDA drivers
by Dylan Reid 01 Mar '14

01 Mar '14

This series rearranges some code in the HDA driver to break dependencies on PCI and allow for most of the HDA code to be reused for non-PCI based HDA controllers. I tried to break it up to make it less scary and to keep changes away from copies of code. It is mostly copies with a few modifications to make more of the code usable by both HDA drivers. The main changes were to avoid using pci to get a device pointer, and to make the bus reads/writes into callbacks so that the Tegra driver could ensure 32-bit access to the HDA registers. The final patch is a work in progress included here to provide context for the preceding patches. The Tegra HDA driver still needs some cleaning up, but nothing that will affect the other changes. The last infrastructure issue is that the Tegra driver uses DT, causing it to be probed before the codec patches. I haven't determined how to fix that yet. I didn't include the patch to move the hda directory from under pci because of its size and how easy it will be to regenerate later. Thanks for looking. Let me know if you think any of these should be squashed or re-ordered and if the changes and breakup of code makes sense. Change since v1: - Rename hda_shared to hda_controller - Add more 'ops' to break out hda_intel specific functions, such as postponed interrupt handling and page allocation - Re order a few patches to make rebases easier - Make bdl_pos_adj optional - I left the position fix code in hda_controller, I don't have a good way to test a refactoring of that, there are a few combinations of settings. It will probably be useful to other controllers as well, but moving it wouldn't be hard if we really want to. Thanks again, Dylan -- Dylan Reid (19): ALSA: hda - Move some definitions to new hda_priv.h ALSA: hda - Allow different ops to read/write registers ALSA: hda - Keep pointer to bdl_pos_fix in chip struct ALSA: hda - Store device pointer in chip structure ALSA: hda - Move pcm ops and support funcs to shared file ALSA: hda - Pull pages allocation to shared file ALSA: hda - Move the dsp loader to hda_shared ALSA: hda - Add function pointer for disabling MSI ALSA: hda - Relocate RIRB/CORB interface to hda_shared ALSA: hda - move alloc_cmd_io to hda_shared. ALSA: hda - Move low level functions to hda_shared ALSA: hda - remove unused clear of STATESTS ALSA: hda - Move azx_interrupt to hda_shared ALSA: hda - Add jackpoll_ms to struct azx ALSA: hda - Pass max_slots and power_save to codec_create ALSA: hda - Move codec create to hda_shared ALSA: core - Define snd_pci_quirk without CONFIG_PCI ALSA: hda - remove PCI dependency in Kconfig WIP: ALSA: hda - Add driver for Tegra SoC HDA include/sound/core.h | 15 +- sound/pci/Kconfig | 4 +- sound/pci/hda/Kconfig | 55 +- sound/pci/hda/Makefile | 8 +- sound/pci/hda/hda_intel.c | 2832 +++----------------------------------------- sound/pci/hda/hda_priv.h | 446 +++++++ sound/pci/hda/hda_shared.c | 2175 ++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_shared.h | 52 + sound/pci/hda/hda_tegra.c | 858 ++++++++++++++ 9 files changed, 3778 insertions(+), 2667 deletions(-) create mode 100644 sound/pci/hda/hda_priv.h create mode 100644 sound/pci/hda/hda_shared.c create mode 100644 sound/pci/hda/hda_shared.h create mode 100644 sound/pci/hda/hda_tegra.c -- 1.8.1.3.605.g02339dd

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[alsa-devel] [PATCH 0/3] To simplify the code for ASoC probe.
by Xiubo Li 01 Mar '14

01 Mar '14

Xiubo Li (3): ASoC: io: Clean up snd_soc_codec_set_cache_io() ASoC: core: Set the default I/O up try regmap. ASoC: SGTL5000: Simplify ASoC probe code. sound/soc/codecs/sgtl5000.c | 8 -------- sound/soc/soc-core.c | 8 ++++---- sound/soc/soc-io.c | 5 ----- 3 files changed, 4 insertions(+), 17 deletions(-) -- 1.8.4

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[alsa-devel] [PATCH 14/28] Remove MACH_SMDKC210
by Richard Weinberger 01 Mar '14

01 Mar '14

The symbol is an orphan, get rid of it. Signed-off-by: Richard Weinberger <richard(a)nod.at> --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 454f41c..7ea576c 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -145,7 +145,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.8.4.2

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[alsa-devel] [PATCH v2 00/11] ASoC: Consolidate enum controls and DAPM MUXs
by Lars-Peter Clausen 01 Mar '14

01 Mar '14

Changes since v1: * Rename the xmax parameter for the SOC_ENUM_SINGLE_VIRT() macro to xitems. Original cover letter below. We currently have 5 nearly identical implementations of enum controls. Two for normal controls and three for DAPM controls. The only difference between them is how and if the selected enum item is mapped to a register value. For normal enums there is a direct mapping of the selected item to the register value, for value enums there is an additional look-up table that maps the selected item to a register value. DAPM also as a virtual enum which is not backed by any hardware register and the enum is solely used to influence DAPM routing decisions. The different implementations can easily be consolidating by making a few small changes. The series first introduces two helper functions to translate between the selected item and the register value. These functions will do the right thing depending on whether the enum has a direct mapping or is a value enum. These functions will be used to consolidate enums and value enums as well as MUXs and value MUXs. The next step is consolidate MUXs and virtual MUXs. This requires enums that are only used as virtual enums to be marked as such. This is done the same way as for virtual mixer controls by setting reg to SND_SOC_NOPM. All users of virtual MUXs are updated accordingly and then virtual MUXs and MUXs are updated to share the same implementation. The last patch in the series does cleanup of dapm_set_path_status() on top of all this. Once this has been merged I'll send a second patch-set replacing and removing the now deprecated ENUM and MUX macros. The series depends on topic/dapm, topic/enum, topic/sign, topic/adau1373, topic/max98090, topic/mc13783, fix/wm8994, topic/wm899 Lars-Peter Clausen (11): ASoC: Remove unused 'reg2' field from soc_enum struct ASoC: Consolidate enum and value enum controls ASoC: dapm: Consolidate MUXs and value MUXs ASoC: Add macros for defining virtual enums ASoC: adau1373: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: max98090: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: mc13783: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8994: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: wm8995: Use SOC_ENUM_SINGLE_VIRT_DECL() ASoC: dapm: Consolidate MUXs and virtual MUXs ASoC: dapm: Break dapm_set_path_status() appart include/sound/soc-dapm.h | 38 +---- include/sound/soc.h | 41 +++-- sound/soc/codecs/adau1373.c | 4 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/mc13783.c | 6 +- sound/soc/codecs/wm8994.c | 3 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/soc-core.c | 101 ++---------- sound/soc/soc-dapm.c | 369 +++++++++----------------------------------- 9 files changed, 136 insertions(+), 430 deletions(-) -- 1.8.0

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