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

[alsa-devel] add Steinberg UR44 USB Audio Interface support to quirks-table
by Vlastimil Wagner 27 Jan '15

27 Jan '15

hi In what state is the driver for Steinberg UR44 thanks Bukowski

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[alsa-devel] [PATCH 00/10] ASoC: sta32x: Add device tree binding
by Thomas Niederprüm 27 Jan '15

27 Jan '15

This patchset is the result of comparing the sta32x driver with the sta350 driver with the goal to make the sta32x driver device tree capable. In the merging process the coding style of the sta350 driver was adopted since it seemed more up to date. I did my best to break up the changes in logical chunks. Additionally some small fixes were added. The comparison with the sta350 lead to the following changes: - Switch to direct usage of the regmap API - Use the kernel gpio framework to control the reset pin of the sta32x - adopt calculation of mlck divider and extrapolation ratio from sta350 - add device tree bindings - Change the Codecs DAI name to be coherent with the sta350 driver Moreover some minor fixes were added: - Add the status register - Fix bit shift value for the IDE bit in the CONFF register - Add description for the driver and I2C dependency in Kconfig Thomas Niederprüm (10): ASoC: sta32x: Convert to direct regmap API usage. ASoC: sta32x: make sta32x a gpio consumer for the reset GPIO ASoC: sta32x: use DECLARE_TLV_DB_RANGE macro. ASoC: sta32x: add status register. ASoC: sta32x: move code to calculate mclk divider and extrapolation ratio to sta32x_hw_params() ASoC: sta32x: add device tree binding. ASoC: sta32x: use dev_dbg() for debug output. ASoC: sta32x: minor Kconfig update. ASoC: sta32x: correct bit shift value for IDE register. ASoC: sta32x: change dai name to be in line with the sta350 driver. .../devicetree/bindings/sound/st,sta32x.txt | 92 ++++ include/sound/sta32x.h | 18 +- sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/sta32x.c | 537 +++++++++++++-------- sound/soc/codecs/sta32x.h | 4 +- 5 files changed, 450 insertions(+), 205 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/st,sta32x.txt -- 2.1.1

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[alsa-devel] [PATCH_V3 0/4] sound:soc: jz4740: DT, dynamic sampling, enable clocks
by Zubair Lutfullah Kakakhel 27 Jan '15

27 Jan '15

Hi, Here are a few simple patches for the jz4740. First adds dynamic sampling support to jz4740-i2s. Then two to add a simple binding and DT support. Then a patch to set the clock rate for i2s. These are in preparation for jz4780 and ci20 later on. Patches are based on 3.19-rc6. Quite disjoint and stay within jz4740 so should apply easily on other trees. If you would like to have them rebased to a different tree, please tell. Thank-you V3 changes: Added dma binding to example but forgot to put it in required properties. V2 changes: Removed a redundant #define Added #ifdef CONFIG_OF Added dma in binding document Removed a redundant call to clk_prepare_enable Zubair Lutfullah Kakakhel (4): sound: soc: jz4740: Add dynamic sampling rate support to jz4740-i2s dt: sound: jz4740: Add binding documentation for jz4740-i2s sound: soc: jz4740: Add DT support to jz4740-i2s driver sound: jz4740: set i2s clk rate to 12MHz .../bindings/sound/ingenic,jz4740-i2s.txt | 23 ++++++++++++++++++++++ sound/soc/jz4740/jz4740-i2s.c | 22 ++++++++++++++++++++- 2 files changed, 44 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt -- 1.9.1

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Re: [alsa-devel] [PATCH v2 00/16] ASoC: samsung: Add clk provider for I2S internal clocks
by Sylwester Nawrocki 27 Jan '15

27 Jan '15

Hello Tobias, On 24/01/15 02:53, Tobias Jakobi wrote: > Hello, > > I've tested the series on my X2 and so far I haven't encountered any > obvious issues with it. > > I have a small question though. With the move to simple-audio-card the > old driver (selected by SND_SOC_ODROIDX2) is probably going away after > some time. > Currently SND_SOC_ODROIDX2 also selects SND_SOC_MAX98090 and > SND_SAMSUNG_I2S, which I believe (correct me if I'm wrong) are necessary > to use the sound subsystem even with simple-audio-card (since the > max98090 codec has to be built at least). I've found no way of selecting > these two manually, so at the moment I need both CONFIG_SND_SIMPLE_CARD > and SND_SOC_ODROIDX2 enabled to use sound. Is this intended behaviour? Thanks for the feedback. It seems I finally forgot to update Kconfig, presumably we should make SND_SOC_ODROIDX2 selecting SND_SIMPLE_CARD, i.e. diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 58c797a..7fb728c 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -247,6 +247,7 @@ config SND_SOC_ODROIDX2 depends on SND_SOC_SAMSUNG select SND_SOC_MAX98090 select SND_SAMSUNG_I2S + select SND_SIMPLE_CARD help Say Y here to enable audio support for the Odroidx2. Making SND_SOC_SAMSUNG selecting SND_SIMPLE_CARD is probably not a good idea and still we would be missing the codec part. I'm going to post a patch as above. > And while I'm at it. I'm booting the board with an upstream u-boot > version which sets the MPLL to 800MHz, which AFAIK is the clocking rate > that makes less 'trouble' (but at the expense of performance). > I was wondering if anyone can comment on whether the recent rework has > any influence on the behaviour when booting with the vendor u-boot > (which sets the MPLL to 880MHz). IIRC then clock rounding issues arose > there. > > It would be interesting to know if this series gets us closer to drive > the MPLL with the recommended (?) higher clocking rate. No, (unfortunately) this series has nothing to do with MPLL. The whole clock tree for sound is configured by the kernel, including the EPLL, which provides reference clock for the audio subsystem. MPLL and EPLL are normally used in different clock domains. So there should be no any clock rounding issues related to bootloader as far as audio is concerned. Please let me know if you find any related problems. -- Regards, Sylwester

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[alsa-devel] [PATCH v2 1/3] ASoC: codecs: wm8904: add dt ids table
by Bo Shen 27 Jan '15

27 Jan '15

From: Alexander Morozov <linux(a)meltdown.ru> Signed-off-by: Alexander Morozov <linux(a)meltdown.ru> [Add driver data to distinguish device type] Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v2: - Add driver data for distinguish the device capability. sound/soc/codecs/wm8904.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..6e3f175 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2255,10 +2255,32 @@ static const struct i2c_device_id wm8904_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); +#ifdef CONFIG_OF +static enum wm8904_type wm8904_data = WM8904; +static enum wm8904_type wm8912_data = WM8912; + +static const struct of_device_id wm8904_of_match[] = { + { + .compatible = "wlf,wm8904", + .data = &wm8904_data, + }, { + .compatible = "wlf,wm8912", + .data = &wm8912_data, + }, { + .compatible = "wlf,wm8918", + .data = &wm8904_data, + }, { + /* sentinel */ + } +}; +MODULE_DEVICE_TABLE(of, wm8904_of_match); +#endif + static struct i2c_driver wm8904_i2c_driver = { .driver = { .name = "wm8904", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(wm8904_of_match), }, .probe = wm8904_i2c_probe, .remove = wm8904_i2c_remove, -- 2.1.0.24.g4109c28

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[alsa-devel] Regression in kernel 3.19-rc5 for usb2 scarlett 18i20
by Guido Aulisi 26 Jan '15

26 Jan '15

Hi, I noticed a regression in 3.19-rc5 kernel (as shipped by fedora rawhide) with usb2 focusrite scarlett 18i20. I get a lot of xruns with jack, even at 1024 frame buffer and 2 periods. In 3.17.8 it works quite well without xruns at the same settings. I use a fedora 20 (64bit) laptop with 8GB ram, I only installed the rawhide kernel to try the new included scarlett mixer, no other software components changed. I can try to do some other test, but compiling the kernel on this laptop is time consuming. Thanks Guido

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[alsa-devel] [PATCH 0/7] ASoC: pcm512x: Clock master modes
by Peter Rosin 26 Jan '15

26 Jan '15

From: Peter Rosin <peda(a)axentia.se> Hi! This series implements BCLK master modes for the pcm512x driver. It has only been tested with the pcm5142 chip, but they are from the same family and should be compatible. I have mainly used the spec for the newer pcm5242 chip (also from the same family) as it fills in a lot of blanks in the pcm512x/pcm514x specs. The code has also seen most of its testing in a 3.10 environment, so there might be some forward-porting warts. But it is able to play sound in 3.18 as well, and most of the changes have little to do with anything but the clocking in the chip itself. Cheers, Peter Peter Rosin (7): ALSA: pcm: Add snd_interval_ranges() and snd_pcm_hw_constraint_ranges() ASoC: pcm512x: Fix spelling of register field names. ASoC: pcm512x: Change register default to match actual content after reset ASoC: pcm512x: Support mastering BCLK/LRCLK without using the PLL ASoC: pcm512x: Support mastering BCLK/LRCLK using the PLL ASoC: pcm512x: Avoid the PLL for the DAC clock, if possible ASoC: pcm512x: Support SND_SOC_DAIFMT_CBM_CFS .../devicetree/bindings/sound/pcm512x.txt | 25 +- include/sound/pcm.h | 12 + sound/core/pcm_lib.c | 85 ++ sound/soc/codecs/pcm512x.c | 955 +++++++++++++++++++- sound/soc/codecs/pcm512x.h | 109 ++- 5 files changed, 1161 insertions(+), 25 deletions(-) -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: rt5640: Add RT5642 ACPI ID for Intel Baytrail
by Jarkko Nikula 26 Jan '15

26 Jan '15

Asus T100TAF uses ACPI ID "10EC5642" for its audio codec. I suppose it is updated ACPI ID for the RT5642 codec since some earlier platforms are using "10EC5640" with the RT5642 too. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> --- ACPI ID found from the DSDT table of Asus T100TAF. This makes the rt5640 codec driver probing but the audio itself is not working due a bit different ACPI data and perhaps HW differences too compared to for instance Asus T100TA. --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b386958de042..178e55d4d481 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2125,6 +2125,7 @@ MODULE_DEVICE_TABLE(of, rt5640_of_match); static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { "10EC5640", 0 }, + { "10EC5642", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); -- 2.1.4

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[alsa-devel] [PATCH] ASoC: rt286: Fix capture volume setting issue
by Bard Liao 26 Jan '15

26 Jan '15

The purpose of rt286_adc_event is to mute/numnte the ADC mixer. However, it will also set the capture volume to default value. As a result, "ADC0 Capture Volume" is not working if it is set before capture start. This patch remove rt286_adc_event and add "ADC0 Capture Switch" to mute/unmute ADC mixer. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 38 ++++++-------------------------------- 1 file changed, 6 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 1d1c7f8..847cc4b 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -417,6 +417,8 @@ static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); static const struct snd_kcontrol_new rt286_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN, RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R("ADC0 Capture Switch", RT286_ADCL_GAIN, + RT286_ADCR_GAIN, 7, 1, 1), SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN, RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN, @@ -538,32 +540,6 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, return 0; } -static int rt286_adc_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - unsigned int nid; - - nid = (w->reg >> 20) & 0xff; - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - snd_soc_update_bits(codec, - VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), - 0x7080, 0x7000); - break; - case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(codec, - VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), - 0x7080, 0x7080); - break; - default: - return 0; - } - - return 0; -} - static int rt286_vref_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -667,12 +643,10 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), /* ADC Mux */ - SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, - &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | - SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, - &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | - SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, + &rt286_adc0_mux), + SND_SOC_DAPM_MUX("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, + &rt286_adc1_mux), /* Audio Interface */ SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 0/2] ASoC: Fix wm97xx touchscreen driver
by Lars-Peter Clausen 26 Jan '15

26 Jan '15

The wm97xx touchscreen driver will bind itself to the snd_ac97 device registered by CODEC drivers and expects that the device has already been reset by the CODEC driver. Previous to commit 6794f709b712 ("ASoC: ac97: Drop delayed device registration") the snd_ac97 device was only registered after the CODEC driver probe function had finished running, but starting with the commit it is registered within snd_soc_new_ac97_codec(). This breaks the touchscreen driver as the reset is no longer performed before the touchscreen driver probe function runs. This patch series introduces a new function snd_soc_alloc_ac97_codec() which allocates the snd_ac97 device, but does not yet register it yet. This allows the CODEC drivers to perform the reset before the device is registered. The series is meant to perform a minimum amount of changes while fixing the issue to prevent introducing other regressions. I have a few more patches that consolidate the reset handling in the drivers and puts it into the core in snd_soc_new_ac97_codec() which makes the split in the drivers unnecessary, but that's something for next and not for stable. Manuel can you give this another round of testing? Thanks, - Lars Lars-Peter Clausen (2): ASoC: Add support for allocating AC'97 device before registering it ASoC: wm97xx: Reset AC'97 device before registering it include/sound/soc.h | 1 + sound/soc/codecs/wm9705.c | 16 ++++++++++------ sound/soc/codecs/wm9712.c | 12 ++++++++---- sound/soc/codecs/wm9713.c | 12 ++++++++---- sound/soc/soc-ac97.c | 34 +++++++++++++++++++++++++++++----- 5 files changed, 56 insertions(+), 19 deletions(-) -- 1.8.0

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