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

[alsa-devel] [PATCH] ASoC: mediatek: Use platform_of_node for machine drivers
by Koro Chen 09 Jul '15

09 Jul '15

This replaces the platform_name in snd_soc_dai_link by device tree node. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- .../devicetree/bindings/sound/mt8173-max98090.txt | 2 ++ .../bindings/sound/mt8173-rt5650-rt5676.txt | 2 ++ sound/soc/mediatek/mt8173-max98090.c | 17 +++++++++++++---- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 19 +++++++++++++++---- 4 files changed, 32 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-max98090.txt b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt index 829bd26..519e97c 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-max98090.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt @@ -3,11 +3,13 @@ MT8173 with MAX98090 CODEC Required properties: - compatible : "mediatek,mt8173-max98090" - mediatek,audio-codec: the phandle of the MAX98090 audio codec +- mediatek,platform: the phandle of MT8173 ASoC platform Example: sound { compatible = "mediatek,mt8173-max98090"; mediatek,audio-codec = <&max98090>; + mediatek,platform = <&afe>; }; diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt index 61e98c9..f205ce9 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt @@ -3,11 +3,13 @@ MT8173 with RT5650 RT5676 CODECS Required properties: - compatible : "mediatek,mt8173-rt5650-rt5676" - mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs +- mediatek,platform: the phandle of MT8173 ASoC platform Example: sound { compatible = "mediatek,mt8173-rt5650-rt5676"; mediatek,audio-codec = <&rt5650 &rt5676>; + mediatek,platform = <&afe>; }; diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c index 4d44b58..2d2536a 100644 --- a/sound/soc/mediatek/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173-max98090.c @@ -103,7 +103,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { .name = "MAX98090 Playback", .stream_name = "MAX98090 Playback", .cpu_dai_name = "DL1", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -114,7 +113,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { .name = "MAX98090 Capture", .stream_name = "MAX98090 Capture", .cpu_dai_name = "VUL", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -125,7 +123,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { { .name = "Codec", .cpu_dai_name = "I2S", - .platform_name = "11220000.mt8173-afe-pcm", .no_pcm = 1, .codec_dai_name = "HiFi", .init = mt8173_max98090_init, @@ -152,9 +149,21 @@ static struct snd_soc_card mt8173_max98090_card = { static int mt8173_max98090_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_max98090_card; - struct device_node *codec_node; + struct device_node *codec_node, *platform_node; int ret, i; + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + for (i = 0; i < card->num_links; i++) { + if (mt8173_max98090_dais[i].platform_name) + continue; + mt8173_max98090_dais[i].platform_of_node = platform_node; + } + codec_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); if (!codec_node) { diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c index 0940553..6f52eca 100644 --- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c @@ -138,7 +138,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .name = "rt5650_rt5676 Playback", .stream_name = "rt5650_rt5676 Playback", .cpu_dai_name = "DL1", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -149,7 +148,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .name = "rt5650_rt5676 Capture", .stream_name = "rt5650_rt5676 Capture", .cpu_dai_name = "VUL", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -161,7 +159,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { { .name = "Codec", .cpu_dai_name = "I2S", - .platform_name = "11220000.mt8173-afe-pcm", .no_pcm = 1, .codecs = mt8173_rt5650_rt5676_codecs, .num_codecs = 2, @@ -209,7 +206,21 @@ static struct snd_soc_card mt8173_rt5650_rt5676_card = { static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_rt5676_card; - int ret; + struct device_node *platform_node; + int i, ret; + + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + if (mt8173_rt5650_rt5676_dais[i].platform_name) + continue; + mt8173_rt5650_rt5676_dais[i].platform_of_node = platform_node; + } mt8173_rt5650_rt5676_codecs[0].of_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); -- 1.8.1.1.dirty

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[alsa-devel] [PATCH] ASoC: arizona: Use the more idiomatic params_width
by Charles Keepax 09 Jul '15

09 Jul '15

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a88202d..c246fbd 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1504,7 +1504,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; - wl = snd_pcm_format_width(params_format(params)); + wl = params_width(params); if (tdm_slots) { arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", -- 1.7.2.5

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[alsa-devel] [PATCH v8 0/4] ASoC: Intel: Add skylake PCM driver
by Vinod Koul 09 Jul '15

09 Jul '15

This patch series adds the SKL PCM driver in first two patches and then adds decoupling of the controller for splitting the links in last patch this version fixes - superflous cast from/to void *, both gcc and sparse are happy! - remove buffer memset - check return of pm_runtime_get_sync - set dma_data to NULL on free - remove ifdef PM for pm_runtime_active - fix pci_set_drvdata calls Jeeja KP (4): ASoC: Intel: Add Skylake HDA platform driver ASoC: Intel: Add Skylake HDA audio driver ASoC: Intel: Add makefile support for SKL driver ASoC: Intel: Add support for decoupled mode in skl driver sound/soc/intel/Kconfig | 4 + sound/soc/intel/Makefile | 1 + sound/soc/intel/skylake/Makefile | 3 + sound/soc/intel/skylake/skl-pcm.c | 916 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 536 ++++++++++++++++++++++ sound/soc/intel/skylake/skl.h | 71 +++ 6 files changed, 1531 insertions(+) create mode 100644 sound/soc/intel/skylake/Makefile create mode 100644 sound/soc/intel/skylake/skl-pcm.c create mode 100644 sound/soc/intel/skylake/skl.c create mode 100644 sound/soc/intel/skylake/skl.h -- 1.9.1

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[alsa-devel] [PATCH 0/6] ASoC: Intel: fix kconfig, symbols and sparse
by Vinod Koul 09 Jul '15

09 Jul '15

So this series has two parts part 1, first three patches which fix the issue of BYT driver not getting compiled in -next and then cleans up symbol usage in various driver Second part fixes (next three) spare warnings on drivers Vinod Koul (6): ASoC: Intel: use CONFIG_SND_SOC for intel boards ASoC: Intel: kconfig - Move DW_DMAC_CORE dependency to machines ASoC: intel - kconfig: remove SND_SOC_INTEL_SST prompt ASOC: intel - hsw: fix sparse unused varible warn ASoC: intel - atom: statify sst_slot_enum_info ASoC: Intel -atom: fix sst_platform_compr_ops declaration sound/soc/intel/Kconfig | 24 +++++++++++++----------- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/sst-atom-controls.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 - sound/soc/intel/atom/sst-mfld-platform.h | 1 + sound/soc/intel/haswell/sst-haswell-pcm.c | 1 - 6 files changed, 16 insertions(+), 15 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 1/3] Revert "ASoC: wm0010: Add missing IRQF_ONESHOT"
by Axel Lin 09 Jul '15

09 Jul '15

This reverts 58d468328646 ("ASoC: wm0010: Add missing IRQF_ONESHOT"). The coccinelle warnings is false positive because the original code does set IRQF_ONESHOT by "trigger |= IRQF_ONESHOT;". Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 6560a66..ff924032 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -953,7 +953,7 @@ static int wm0010_spi_probe(struct spi_device *spi) trigger = IRQF_TRIGGER_FALLING; trigger |= IRQF_ONESHOT; - ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger | IRQF_ONESHOT, + ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, "wm0010", wm0010); if (ret) { dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", -- 2.1.0

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[alsa-devel] [PATCH] ASoC: dapm: Fix kcontrol widget name memory management
by Lars-Peter Clausen 09 Jul '15

09 Jul '15

The name field of the widget template is only used inside snd_soc_dapm_new_control_unlocked() which allocates a copy for the actual widget. This means we need to free the name allocated for the template in dapm_kcontrol_data_alloc() and not the name of the actual widget in dapm_kcontrol_free(). Otherwise we get a double free on the widget name and a memory leak on the template name. Fixes: 773da9b358bf ("ASoC: dapm: Append "Autodisable" to autodisable widget names") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6bac491..1628f0c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -358,9 +358,10 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->widget = snd_soc_dapm_new_control_unlocked(widget->dapm, &template); + kfree(name); if (!data->widget) { ret = -ENOMEM; - goto err_name; + goto err_data; } } break; @@ -391,9 +392,10 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->widget = snd_soc_dapm_new_control_unlocked( widget->dapm, &template); + kfree(name); if (!data->widget) { ret = -ENOMEM; - goto err_name; + goto err_data; } snd_soc_dapm_add_path(widget->dapm, data->widget, @@ -408,8 +410,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, return 0; -err_name: - kfree(name); err_data: kfree(data); return ret; @@ -418,8 +418,6 @@ err_data: static void dapm_kcontrol_free(struct snd_kcontrol *kctl) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl); - if (data->widget) - kfree(data->widget->name); kfree(data->wlist); kfree(data); } -- 2.1.4

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Re: [alsa-devel] Fw: Using loopback card to Connect GSM two way call to the real sound card UDA1345TS
by Srinivasan S 09 Jul '15

09 Jul '15

Dear Clemens, Many Thanks a lot for all your support w.r.t this thread so far Am facing overrun & underrun issues, when I run the the above GSM application with the attached asound.conf I have some understanding by googling & found why this overrun & underrun issues occurs ie., unable to transfer audio frames from hardware buffers to application buffers(for capture) & the viceversa(for playback), As I googled, I didn't get detailed information with bits & pieces here & there, could you please help me out in resolving this overrun & underrun issue ie., how to set this buffer size or only buffer_size is causing this issue or any other parameters like period_size, period_time values should also be considered inorder to fix this overrun & underrun issue Now I have randomly set the buffer size ie., 4097 for 8Khz sampling rate But I have no idea logically how this buffer size 4097 is working for 8Khz & when I copied the same buffer size to 48Khz, the audio gets cut & again overrun & underrun issue is seen Could you please help me out how to fix this overrun & underrun issue logically, rather than I have that an understanding of programming some random values like 4097 Is it possible to fix this issue in asound.conf itself, rather than using multithreading in GSM application So that fixing these types of issues henceforth will be faster & addressed logically & will be helpful for others as well Kindly do the needful as early as possible Many Thanks in advance once again ________________________________________ From: Srinivasan S Sent: Wednesday, July 1, 2015 12:54 PM To: Clemens Ladisch Cc: alsa-devel(a)alsa-project.org; linux-audio-dev(a)lists.linuxaudio.org Subject: Re: [alsa-devel] Fw: Using loopback card to Connect GSM two way call to the real sound card UDA1345TS Dear Clemens, Many Thanks a lot for all your support w.r.t this thread so far Am facing overrun & underrun issues on TI AM335x, when I run the the above GSM application with the attached asound.conf I have some understanding by googling & found why this overrun & underrun issues occurs ie., unable to transfer audio frames from hardware buffers to application buffers(for capture) & the viceversa(for playback), As I googled, I didn't get detailed information with bits & pieces here & there, could you please help me out in resolving this overrun & underrun issue ie., how to set this buffer size or only buffer_size is causing this issue or any other parameters like period_size, period_time values should also be considered inorder to fix this overrun & underrun issue Now I have randomly set the buffer size ie., 4097 for 8Khz sampling rate But I have no idea logically how this buffer size 4097 is working for 8Khz & when I copied the same buffer size to 48Khz, the audio gets cut & again overrun & underrun issue is seen Could you please help me out how to fix this overrun & underrun issue logically, rather than I have that an understanding of programming some random values like 4097 Is it possible to fix this issue in asound.conf itself, rather than using multithreading in GSM application So that fixing these types of issues henceforth will be faster & addressed logically & will be helpful for others as well Kindly do the needful as early as possible Many Thanks in advance once again ________________________________________ From: alsa-devel-bounces(a)alsa-project.org <alsa-devel-bounces(a)alsa-project.org> on behalf of Srinivasan S <srinivasan.s(a)tataelxsi.co.in> Sent: Wednesday, June 3, 2015 11:35 AM To: Clemens Ladisch Cc: alsa-devel(a)alsa-project.org; linux-audio-dev(a)lists.linuxaudio.org Subject: Re: [alsa-devel] Fw: Using loopback card to Connect GSM two way call to the real sound card UDA1345TS Dear Clemens As I didn't find Jack mailing lists by googling, could you please loop Jack mailing lists to this mail chain Kindly do the needful as early as possible Many Thanks in advance ________________________________________ From: Clemens Ladisch <clemens(a)ladisch.de> Sent: Monday, June 1, 2015 8:09 PM To: Srinivasan S Cc: alsa-devel(a)alsa-project.org; linux-audio-dev(a)lists.linuxaudio.org Subject: Re: [alsa-devel] Fw: Using loopback card to Connect GSM two way call to the real sound card UDA1345TS Srinivasan S wrote: > 2. Could you please let me know, I have downloaded jack-1.9.10.tar.bz2, how this needs to be installed in my rootfs IIRC Jack uses some non-standard build system. Try asking on the Jack mailing list how to cross-compile it. Please note that the ALSA Jack plugin is part of the alsa-plugins package. And as I already mentioned, it is unlikely that Jack will use less CPU than dshare. Regards, Clemens _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] Tascam US-16x08 0644:8047
by Greg 09 Jul '15

09 Jul '15

Greetings, I am fairly new to USB dev (in linux in particular, but also in general), but I would very much like to try to get support for the above device working in snd-usb-audio. - Is this an appropriate place to discuss snd-usb-audio? - Are there any recommended reading pointers for the quirk table? - or any highly recommended USB reading material that I might not have found? I patched parse_audio_format_rates_v2(), get_sample_rate_v2(), and set_sample_rate_v2(), and through some sort of beginner luck was able to get aplay audio out of the first two channels. That was incomplete hackery though (eg fixed sample rate), and I would like to learn how to properly add quirk support. There have been other reports that this device worked OOTB, but I fail to see how! I've also been examining the traffic to the device with wireshark and a win7 vm, but the learning curve for USB is a bit steep, so I am digesting. (: If anyone can provide suggestions on lsusb output alone, here's what I have: http://pastebin.com/pA9MLQet cheers, Greg

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[alsa-devel] [PATCH v7 0/4] ASoC: intel - add skylake PCM driver
by Vinod Koul 09 Jul '15

09 Jul '15

This patch series adds the SKL PCM driver in first two patches and then adds decoupling of the controller for splitting the links in last patch This version is rebased on 4.2-rc1 and has fixed the style issues identfied in last review Jeeja KP (3): ASoC: Intel: add Skylake HDA platform driver ASoC: Intel - add Skylake HDA audio driver ASoC: Intel - add makefile support for SKL driver Vinod Koul (1): ASoC: intel - adds support for decoupled mode in skl driver sound/soc/intel/Kconfig | 7 + sound/soc/intel/Makefile | 1 + sound/soc/intel/skylake/Makefile | 3 + sound/soc/intel/skylake/skl-pcm.c | 936 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 543 ++++++++++++++++++++++ sound/soc/intel/skylake/skl.h | 71 +++ 6 files changed, 1561 insertions(+) create mode 100644 sound/soc/intel/skylake/Makefile create mode 100644 sound/soc/intel/skylake/skl-pcm.c create mode 100644 sound/soc/intel/skylake/skl.c create mode 100644 sound/soc/intel/skylake/skl.h -- 1.9.1

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[alsa-devel] [PATCH 0/2] ASoC: Intel: skylake - match braces for if-else
by Vinod Koul 09 Jul '15

09 Jul '15

Coding style mandates that if-else should match braces even if one of them is a single line statement. Fix it in two places missed earlier Mark, you can also fold these two patches with SKL driver series (1st and 2nd patch can fold in 1st and 2nd patch respectively) or apply independetly, either way okay Vinod Koul (2): ASoC: Intel: skl-pcm.c match braces for if-else ASoC: Intel: skl.c match braces for if-else sound/soc/intel/skylake/skl-pcm.c | 3 ++- sound/soc/intel/skylake/skl.c | 4 ++-- 2 files changed, 4 insertions(+), 3 deletions(-) -- 1.9.1

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