July 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: jack: Use power efficient workqueue
by Mark Brown 18 Jul '13

18 Jul '13

From: Mark Brown <broonie(a)linaro.org> The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..7aa26b5 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -263,7 +263,7 @@ static irqreturn_t gpio_handler(int irq, void *data) if (device_may_wakeup(dev)) pm_wakeup_event(dev, gpio->debounce_time + 50); - schedule_delayed_work(&gpio->work, + queue_delayed_work(system_power_efficient_wq, &gpio->work, msecs_to_jiffies(gpio->debounce_time)); return IRQ_HANDLED; -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: Fix error path
by Fabio Estevam 18 Jul '13

18 Jul '13

If the 'failed to find codec platform device' error path is executed, it should jump to 'fail' label instead of returning an error immediately. 'fail' label will then free the ssi_np and codec_np previously acquired nodes. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-wm8962.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 52a36a9..1d70e27 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -217,7 +217,8 @@ static int imx_wm8962_probe(struct platform_device *pdev) codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev || !codec_dev->driver) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EINVAL; + ret = -EINVAL; + goto fail; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: tegra: depend on ARCH_TEGRA, not ARCH_TEGRA_*
by Stephen Warren 18 Jul '13

18 Jul '13

From: Stephen Warren <swarren(a)nvidia.com> I'm planning to remove CONFIG_ARCH_TEGRA_*_SOC, leaving just ARCH_TEGRA. This will reduce the number of configurations that need to be supported by core Tegra code, e.g. CPU idle, hotplug. As a precursor, we need to update Kconfig for all Tegra drivers not to reference ARCH_TEGRA_*_SOC. Stop selecting I2S/AC97/SPDIF controller drivers from the machine driver config options; this doesn't really work if we don't know which specific SoC(s) we're building for. However, set their default values based on SND_SOC_TEGRA, since most people will probably want to enable support for all SoCs. This also avoids the need to change any defconfig files. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/Kconfig | 36 +++++++++++++++--------------------- 1 file changed, 15 insertions(+), 21 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 68fd6d0..321a6c9 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -7,8 +7,9 @@ config SND_SOC_TEGRA Say Y or M here if you want support for SoC audio on Tegra. config SND_SOC_TEGRA20_AC97 - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 AC97 controller" + depends on SND_SOC_TEGRA + default SND_SOC_TEGRA select SND_SOC_AC97_BUS select SND_SOC_TEGRA20_DAS help @@ -18,15 +19,16 @@ config SND_SOC_TEGRA20_AC97 config SND_SOC_TEGRA20_DAS tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 DAS module. You will also need to select the individual machine drivers to support below. config SND_SOC_TEGRA20_I2S - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 I2C controller" + depends on SND_SOC_TEGRA + default SND_SOC_TEGRA select SND_SOC_TEGRA20_DAS help Say Y or M if you want to add support for codecs attached to the @@ -34,9 +36,9 @@ config SND_SOC_TEGRA20_I2S machine drivers to support below. config SND_SOC_TEGRA20_SPDIF - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC - default m + tristate "Tegra20 SPDIF controller" + default SND_SOC_TEGRA + depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support @@ -44,15 +46,16 @@ config SND_SOC_TEGRA20_SPDIF config SND_SOC_TEGRA30_AHUB tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 AHUB module. You will also need to select the individual machine drivers to support below. config SND_SOC_TEGRA30_I2S - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + tristate "Tegra30 I2S controller" + depends on SND_SOC_TEGRA + default SND_SOC_TEGRA select SND_SOC_TEGRA30_AHUB help Say Y or M if you want to add support for codecs attached to the @@ -62,8 +65,6 @@ config SND_SOC_TEGRA30_I2S config SND_SOC_TEGRA_RT5640 tristate "SoC Audio support for Tegra boards using an RT5640 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_RT5640 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -72,8 +73,6 @@ config SND_SOC_TEGRA_RT5640 config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8753 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -82,8 +81,6 @@ config SND_SOC_TEGRA_WM8753 config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -92,8 +89,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_WM9712 tristate "SoC Audio support for Tegra boards using a WM9712 codec" - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC && GPIOLIB - select SND_SOC_TEGRA20_AC97 + depends on SND_SOC_TEGRA && GPIOLIB select SND_SOC_WM9712 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -102,7 +98,6 @@ config SND_SOC_TEGRA_WM9712 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" depends on SND_SOC_TEGRA && I2C - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TLV320AIC23 help Say Y or M here if you want to add support for SoC audio on the @@ -111,7 +106,6 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_ALC5632 help Say Y or M here if you want to add support for SoC audio on the -- 1.8.1.5

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[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for Focusrite Scarlett 18i8
by Chris J Arges 18 Jul '13

18 Jul '13

This patch is similar to commit 1762a59d8e8b5e99f6f4a0f292b40f3cacb108ba. The same non-standard interface descriptor causes snd_usb_create_mixer() to fail for the Focusrite Scarlett 18i8 as well. Signed-off-by: Chris J Arges <christopherarges(a)gmail.com> --- sound/usb/quirks-table.h | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 7f1585b..b954ec5 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3232,6 +3232,52 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Focusrite Scarlett 18i8 + * + * Avoid mixer creation similar to the Scarlett 18i6 interface. + */ + USB_DEVICE(0x1235, 0x8014), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Focusrite", + .product_name = "Scarlett 18i8", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = & (const struct snd_usb_audio_quirk[]) { + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Control Device) */ + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + /* InterfaceSubClass 1 (Device Firmware Update) */ + .ifnum = 5, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, { /* -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: pcm: Use the power efficient workqueue for delayed powerdown
by Mark Brown 18 Jul '13

18 Jul '13

From: Mark Brown <broonie(a)linaro.org> There is no need to use a normal per-CPU workqueue for delayed power downs as they're not timing or performance critical and waking up a core for them would defeat some of the point. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-pcm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b6c6403..f4f68cb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -411,8 +411,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } else { /* start delayed pop wq here for playback streams */ rtd->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); + queue_delayed_work(system_power_efficient_wq, + &rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); } } else { /* capture streams can be powered down now */ -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: wm8904: fix the typo error for LINER Mux
by Bo Shen 18 Jul '13

18 Jul '13

fix the typo error, from "LINEL Mux" to "LINER Mux" Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 3ff195c..bdd575a 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1017,7 +1017,7 @@ static const struct soc_enum liner_enum = SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text); static const struct snd_kcontrol_new liner_mux = - SOC_DAPM_ENUM("LINEL Mux", liner_enum); + SOC_DAPM_ENUM("LINER Mux", liner_enum); static const char *sidetone_text[] = { "None", "Left", "Right" -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ALSA: hdmi - poll eld at resume time
by Wang Xingchao 18 Jul '13

18 Jul '13

Hdmi driver may not receive intrinsic event from gfx side when it's in runtime suspend mode. There's no ELD info when exit from runtime suspend. This patch avoid missing ELD info. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7803ddd..cb8ac66 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1868,12 +1868,33 @@ static void generic_hdmi_free(struct hda_codec *codec) kfree(spec); } +#ifdef CONFIG_PM +static int generic_hdmi_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + generic_hdmi_init(codec); + snd_hda_codec_resume_amp(codec); + snd_hda_codec_resume_cache(codec); + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + hdmi_present_sense(per_pin, 1); + } + return 0; +} +#endif + static const struct hda_codec_ops generic_hdmi_patch_ops = { .init = generic_hdmi_init, .free = generic_hdmi_free, .build_pcms = generic_hdmi_build_pcms, .build_controls = generic_hdmi_build_controls, .unsol_event = hdmi_unsol_event, +#ifdef CONFIG_PM + .resume = generic_hdmi_resume, +#endif }; -- 1.8.1.2

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[alsa-devel] [PATCH, RESEND] ALSA: ASoC: au1x: fix ac97c module build
by Manuel Lauss 17 Jul '13

17 Jul '13

The recent conversion to module_platform_driver left a stray ampersand in there, preventing a build. Cc: Mark Brown <broonie(a)linaro.org> Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- Against 3.11-rc1, added Mark to Cc. sound/soc/au1x/ac97c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index d6f7694..c8a2de1 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -341,7 +341,7 @@ static struct platform_driver au1xac97c_driver = { .remove = au1xac97c_drvremove, }; -module_platform_driver(&au1xac97c_driver); +module_platform_driver(au1xac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver"); -- 1.8.3.2

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[alsa-devel] Sound Card driver for audio codec, mux and dsp.
by Diego Sueiro 17 Jul '13

17 Jul '13

Hi there, I need to develop a solution to expose to userspace a single sound card with mixer controls for: Audio codec : sgtl5000 Audio mux: used to select an audio source on sgtl5000 LINE_IN. Audio dsp :sta309a How can I implement this drivers? Should I have to create a sound-card driver which is going to register and control the other drivers (codec, mux, dsp)? Is there some device driver or tutorial guide that I can use as an example? I'm using the kernel 3.0.15 from Freescale [1]. [1] - http://git.freescale.com/git/cgit.cgi/imx/linux-2.6-imx.git/log/?h=imx_3.0.… Kind Regards, -- *dS Diego Sueiro

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[alsa-devel] idea: a reserve alsa plugin
by David Henningsson 17 Jul '13

17 Jul '13

Just had an idea which I'll write down here before I forget it again...and I'm not saying I'll implement this anytime soon either, but here goes: There is a device reserve protocol between PulseAudio and JACK2 - when JACK needs the sound card, it'll send a dbus message to PulseAudio and grab a name in D-Bus. However, there are plenty of applications who like to access ALSA directly, without going through JACK2 or PulseAudio. By making a "reserve" plugin, we could have this functionality for those apps too. In practice, if the app usually opens "plughw:0" or "hw:0", it could instead open "reserve:plughw:0" or "reserve:hw:0" to also reserve the device from PulseAudio usage while the device is open. Meanwhile, PulseAudio is free to use other audio devices (which is not the case when using e g pasuspender). How does that sound? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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