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

Re: [alsa-devel] plans for fireworks driver
by Doug McLain 10 Mar '13

10 Mar '13

After realizing that the ffado-mixer also didn't work on my AF12, I found that echo firmware > 4.8 broke ffado for the AF12 & AF8: http://subversion.ffado.org/ticket/360 Interestingly though, I have been running 5.7 for a couple years using the snd-fireworks driver with no problems. I have no use for the mixer on the AF12 after I initially set it up way before the firmware problem came about, so I never noticed anything was wrong. Your driver now discovers the AF12 fine: $ cat /proc/asound/card1/efc_hardware flags: 0x11 guid_hi: 0x14860F guid_lo: 0xC44A84A6 type: 0xAF12 version: 0x0 vendor_name: Echo Digital Audio model_name: AudioFire12 supported_clock: 0x5 nb_1394_playback_channels: 0xC nb_1394_capture_channels: 0xC nb_phys_audio_out: 0xC nb_phys_audio_in: 0xC nb_out_groups: 0x1 out_group[0x0]: type 0x0, count 0x12 out_group[0x1]: type 0x0, count 0x0 out_group[0x2]: type 0x0, count 0x0 out_group[0x3]: type 0x0, count 0x0 out_group[0x4]: type 0x0, count 0x0 out_group[0x5]: type 0x0, count 0x0 out_group[0x6]: type 0x0, count 0x0 out_group[0x7]: type 0x0, count 0x0 nb_in_groups: 0x1 out_group[0x0]: type 0x0, count 0x12 out_group[0x1]: type 0x0, count 0x0 out_group[0x2]: type 0x0, count 0x0 out_group[0x3]: type 0x0, count 0x0 out_group[0x4]: type 0x0, count 0x0 out_group[0x5]: type 0x0, count 0x0 out_group[0x6]: type 0x0, count 0x0 out_group[0x7]: type 0x0, count 0x0 nb_midi_out: 0x1 nb_midi_in: 0x1 max_sample_rate: 0x2EE00 min_sample_rate: 0x7D00 dsp_version: 0x4080000 arm_version: 0x4080000 mixer_playback_channels: 0xC mixer_capture_channels: 0xC fpga_version: 0x0 nb_1394_playback_channels_2x: 0xC nb_1394_capture_channels_2x: 0xC nb_1394_playback_channels_4x: 0xC nb_1394_capture_channels_4x: 0xC Interestingly though, I've been running On 03/09/2013 09:11 PM, Takashi Sakamoto wrote: > Hi Doug, > > Thanks for your checking my source. > > >> The latest change, which prompted me to write this email in the >> first place, was the removal of __devinit and friends from kernel 3.8. > > Oh, it's my fault... __devXXX macros are removed from Linux tree since > Dec 2012. > For Linux 3.8-rc4 or later my module needs your patch but i's inconvenient. > I realized it but have been forgeting... > > I remove them with this commit. Now you can build my module without your > patch. > https://github.com/takaswie/snd-firewire-improve/commit/77742d3f68e98e94ff9… > > > > > ...and the AF2 > > Thank you. I add your log in the end of README. > https://github.com/takaswie/snd-firewire-improve/commit/52beb73fa1e248a8eec… > > > >> It discovers the AF2, but not the AF12. Here is the log from the AF12: > > Mar 9 15:47:11 iggy kernel: [63134.769274] ALSA fireworks_command.c:313 > > EFC Failed [0/1]: 0 > > H'm... It looks to fail to get the number of hardware metering channels > when proving my module. > To investigating this, anyway, I need the hardware information of AF12... > > Would you try this patch for AF12? This patch disable to get hardware > metering channels during module proving process. > > --- a/fireworks/fireworks.c > +++ b/fireworks/fireworks.c > @@ -329,11 +329,6 @@ static int snd_efw_probe(struct device *dev) > if (err < 0) > goto error_card; > > - /* get the number of hardware meters */ > - err = snd_efw_get_hardware_meters_count(efw); > - if (err < 0) > - goto error_card; > - > efw->syt_interval = 8; > > for (i = 0; i < MAX_MIDI_OUTPUTS; i += 1) > > > Thanks > > > Takashi Sakamoto > o-takashi(a)sakamocchi.jp > > (Mar 10 2013 05:52), Doug McLain wrote: >> Attached is a patch of my hack to build your source of Fedora 18 with >> kernel 3.8. As I mentioned in the last email, I have had to do various >> little hacks to the other snd-fireworks module source over the past few >> years as chnages were made to the kernel API. The latest change, which >> prompted me to write this email in the first place, was the removal of >> __devinit and friends from kernel 3.8. So now its built, installed, and >> running. It discovers the AF2, but not the AF12. Here is the log from >> the AF12: >> >> Mar 9 15:47:06 iggy kernel: [63129.761604] firewire_core 0000:01:0b.0: >> phy config: new root=ffc2, gap_count=7 >> Mar 9 15:47:11 iggy kernel: [63134.769274] ALSA fireworks_command.c:313 >> EFC Failed [0/1]: 0 >> Mar 9 15:47:11 iggy kernel: [63134.769289] snd-fireworks: probe of >> fw2.0 failed with error -5 >> Mar 9 15:47:11 iggy kernel: [63134.769293] firewire_core 0000:01:0b.0: >> created device fw2: GUID 0014860fc44a84a6, S400 >> >> >> ...and the AF2 >> $ cat /proc/asound/card0/efc_hardware >> flags: 0xA1 >> guid_hi: 0x148606 >> guid_lo: 0x9D5BD6CA >> type: 0xAF2 >> version: 0x0 >> vendor_name: Echo Digital Audio >> model_name: AudioFire2 >> supported_clock: 0x9 >> nb_1394_playback_channels: 0x6 >> nb_1394_capture_channels: 0x4 >> nb_phys_audio_out: 0x6 >> nb_phys_audio_in: 0x4 >> nb_out_groups: 0x3 >> out_group[0x0]: type 0x0, count 0x2 >> out_group[0x1]: type 0x5, count 0x2 >> out_group[0x2]: type 0x1, count 0x2 >> out_group[0x3]: type 0x0, count 0x0 >> out_group[0x4]: type 0x0, count 0x0 >> out_group[0x5]: type 0x0, count 0x0 >> out_group[0x6]: type 0x0, count 0x0 >> out_group[0x7]: type 0x0, count 0x0 >> nb_in_groups: 0x2 >> out_group[0x0]: type 0x0, count 0x2 >> out_group[0x1]: type 0x5, count 0x2 >> out_group[0x2]: type 0x1, count 0x2 >> out_group[0x3]: type 0x0, count 0x0 >> out_group[0x4]: type 0x0, count 0x0 >> out_group[0x5]: type 0x0, count 0x0 >> out_group[0x6]: type 0x0, count 0x0 >> out_group[0x7]: type 0x0, count 0x0 >> nb_midi_out: 0x1 >> nb_midi_in: 0x1 >> max_sample_rate: 0x17700 >> min_sample_rate: 0x7D00 >> dsp_version: 0x0 >> arm_version: 0x4080000 >> mixer_playback_channels: 0x6 >> mixer_capture_channels: 0x4 >> fpga_version: 0x3000200 >> nb_1394_playback_channels_2x: 0x6 >> nb_1394_capture_channels_2x: 0x4 >> nb_1394_playback_channels_4x: 0x6 >> nb_1394_capture_channels_4x: 0x4 >> >> -D >> >> On 03/09/2013 07:30 AM, Takashi Sakamoto wrote: >>> Dear Doug, >>> >>> I'm a newcomer of ALSA project and currently working for snd-fireworks. >>> Then I have a favour of you. >>> (My first name is the same one that ALSA core developer has. But I'm >>> newcomer Takashi) >>> >>> Unfortunatelly my post isn't delivered by mailman of >>> alsa-devel(a)alsa-project.org. But I'm not a professional for mailman or >>> SMTP so it takes a bit time that I debut in alsa-devel... >>> >>> Well, my source is here. >>> https://github.com/takaswie/snd-firewire-improve >>> >>> It's a bit worse to you because current source has a lack of some >>> functions that Clemens' snd-fireworks has. >>> >>> - playback is available, capture is not. Additionally 2ch are >>> available (under investigating...) >>> - MIDI is not available >>> >>> The reason is snd-firewire-lib.ko has these functions. Currently I'm >>> working for them. >>> >>> But the implementation of hardware control is almost done: >>> - Hardware metering/monitoring/mixing are visible via procfs, but not >>> configurable >>> - sampling rate, digital mode, digital format and clock source are >>> changeable via alsamixer >>> >>> Hardware metering/monitoring/mixing will also be available via alsamixer >>> after I've investigated the suitable way in ALSA's control component >>> API. >>> >>> My favor of you is "building and installing my module and give me >>> hardware information of your AF2 and AF12". I'm working with Ubuntu and >>> utilize DKMS for debugging my module. You can see an instruction in >>> README. >>> >>> If your Fireworks is identified by your system, there is >>> /proc/asound/cardX/efc_hardware. If you "cat" this file, my module >>> submit a command to your device to get hardware information and print >>> the value. My module decide available functions refering it so it's very >>> important. >>> >>> This is the output of my device, for your information: >>> $ cat /proc/asound/card2/efc_hardware >>> flags: 0x625 >>> guid_hi: 0x14860A >>> guid_lo: 0x5B6BDB9B >>> type: 0xAF9 >>> version: 0x0 >>> vendor_name: Echo Digital Audio >>> model_name: AudioFirePre8 >>> supported_clock: 0x1D >>> nb_1394_playback_channels: 0x10 >>> nb_1394_capture_channels: 0x10 >>> nb_phys_audio_out: 0x10 >>> nb_phys_audio_in: 0x10 >>> nb_out_groups: 0x2 >>> out_group[0x0]: type 0x0, count 0x8 >>> out_group[0x1]: type 0x3, count 0x8 >>> out_group[0x2]: type 0x0, count 0x0 >>> out_group[0x3]: type 0x0, count 0x0 >>> out_group[0x4]: type 0x0, count 0x0 >>> out_group[0x5]: type 0x0, count 0x0 >>> out_group[0x6]: type 0x0, count 0x0 >>> out_group[0x7]: type 0x0, count 0x0 >>> nb_in_groups: 0x2 >>> out_group[0x0]: type 0x0, count 0x8 >>> out_group[0x1]: type 0x3, count 0x8 >>> out_group[0x2]: type 0x0, count 0x0 >>> out_group[0x3]: type 0x0, count 0x0 >>> out_group[0x4]: type 0x0, count 0x0 >>> out_group[0x5]: type 0x0, count 0x0 >>> out_group[0x6]: type 0x0, count 0x0 >>> out_group[0x7]: type 0x0, count 0x0 >>> nb_midi_out: 0x1 >>> nb_midi_in: 0x1 >>> max_sample_rate: 0x17700 >>> min_sample_rate: 0x7D00 >>> dsp_version: 0x0 >>> arm_version: 0x5050000 >>> mixer_playback_channels: 0x10 >>> mixer_capture_channels: 0x10 >>> fpga_version: 0x4010100 >>> nb_1394_playback_channels_2x: 0xC >>> nb_1394_capture_channels_2x: 0xC >>> nb_1394_playback_channels_4x: 0xA >>> nb_1394_capture_channels_4x: 0xA >>> >>> Kind Regards >>> >>> >>> Takashi Sakamoto >>> o-takashi(a)sakamocchi.jp >>> >>> (2013年03月09日 06:49), Doug McLain wrote: >>>> I have been building and using the fireworks driver from >>>> alsa-kprivate/fireworks for a few years now with my AF2 and AF12. The >>>> driver works great for me, although every so often after a kernel >>>> update, compilation breaks and I have to hack at it a little bit to get >>>> it to build again. >>>> >>>> Are there any plans to include this driver into the kernel? >>> >

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Re: [alsa-devel] plans for fireworks driver
by Doug McLain 09 Mar '13

09 Mar '13

Attached is a patch of my hack to build your source of Fedora 18 with kernel 3.8. As I mentioned in the last email, I have had to do various little hacks to the other snd-fireworks module source over the past few years as chnages were made to the kernel API. The latest change, which prompted me to write this email in the first place, was the removal of __devinit and friends from kernel 3.8. So now its built, installed, and running. It discovers the AF2, but not the AF12. Here is the log from the AF12: Mar 9 15:47:06 iggy kernel: [63129.761604] firewire_core 0000:01:0b.0: phy config: new root=ffc2, gap_count=7 Mar 9 15:47:11 iggy kernel: [63134.769274] ALSA fireworks_command.c:313 EFC Failed [0/1]: 0 Mar 9 15:47:11 iggy kernel: [63134.769289] snd-fireworks: probe of fw2.0 failed with error -5 Mar 9 15:47:11 iggy kernel: [63134.769293] firewire_core 0000:01:0b.0: created device fw2: GUID 0014860fc44a84a6, S400 ...and the AF2 $ cat /proc/asound/card0/efc_hardware flags: 0xA1 guid_hi: 0x148606 guid_lo: 0x9D5BD6CA type: 0xAF2 version: 0x0 vendor_name: Echo Digital Audio model_name: AudioFire2 supported_clock: 0x9 nb_1394_playback_channels: 0x6 nb_1394_capture_channels: 0x4 nb_phys_audio_out: 0x6 nb_phys_audio_in: 0x4 nb_out_groups: 0x3 out_group[0x0]: type 0x0, count 0x2 out_group[0x1]: type 0x5, count 0x2 out_group[0x2]: type 0x1, count 0x2 out_group[0x3]: type 0x0, count 0x0 out_group[0x4]: type 0x0, count 0x0 out_group[0x5]: type 0x0, count 0x0 out_group[0x6]: type 0x0, count 0x0 out_group[0x7]: type 0x0, count 0x0 nb_in_groups: 0x2 out_group[0x0]: type 0x0, count 0x2 out_group[0x1]: type 0x5, count 0x2 out_group[0x2]: type 0x1, count 0x2 out_group[0x3]: type 0x0, count 0x0 out_group[0x4]: type 0x0, count 0x0 out_group[0x5]: type 0x0, count 0x0 out_group[0x6]: type 0x0, count 0x0 out_group[0x7]: type 0x0, count 0x0 nb_midi_out: 0x1 nb_midi_in: 0x1 max_sample_rate: 0x17700 min_sample_rate: 0x7D00 dsp_version: 0x0 arm_version: 0x4080000 mixer_playback_channels: 0x6 mixer_capture_channels: 0x4 fpga_version: 0x3000200 nb_1394_playback_channels_2x: 0x6 nb_1394_capture_channels_2x: 0x4 nb_1394_playback_channels_4x: 0x6 nb_1394_capture_channels_4x: 0x4 -D On 03/09/2013 07:30 AM, Takashi Sakamoto wrote: > Dear Doug, > > I'm a newcomer of ALSA project and currently working for snd-fireworks. > Then I have a favour of you. > (My first name is the same one that ALSA core developer has. But I'm > newcomer Takashi) > > Unfortunatelly my post isn't delivered by mailman of > alsa-devel(a)alsa-project.org. But I'm not a professional for mailman or > SMTP so it takes a bit time that I debut in alsa-devel... > > Well, my source is here. > https://github.com/takaswie/snd-firewire-improve > > It's a bit worse to you because current source has a lack of some > functions that Clemens' snd-fireworks has. > > - playback is available, capture is not. Additionally 2ch are > available (under investigating...) > - MIDI is not available > > The reason is snd-firewire-lib.ko has these functions. Currently I'm > working for them. > > But the implementation of hardware control is almost done: > - Hardware metering/monitoring/mixing are visible via procfs, but not > configurable > - sampling rate, digital mode, digital format and clock source are > changeable via alsamixer > > Hardware metering/monitoring/mixing will also be available via alsamixer > after I've investigated the suitable way in ALSA's control component API. > > My favor of you is "building and installing my module and give me > hardware information of your AF2 and AF12". I'm working with Ubuntu and > utilize DKMS for debugging my module. You can see an instruction in README. > > If your Fireworks is identified by your system, there is > /proc/asound/cardX/efc_hardware. If you "cat" this file, my module > submit a command to your device to get hardware information and print > the value. My module decide available functions refering it so it's very > important. > > This is the output of my device, for your information: > $ cat /proc/asound/card2/efc_hardware > flags: 0x625 > guid_hi: 0x14860A > guid_lo: 0x5B6BDB9B > type: 0xAF9 > version: 0x0 > vendor_name: Echo Digital Audio > model_name: AudioFirePre8 > supported_clock: 0x1D > nb_1394_playback_channels: 0x10 > nb_1394_capture_channels: 0x10 > nb_phys_audio_out: 0x10 > nb_phys_audio_in: 0x10 > nb_out_groups: 0x2 > out_group[0x0]: type 0x0, count 0x8 > out_group[0x1]: type 0x3, count 0x8 > out_group[0x2]: type 0x0, count 0x0 > out_group[0x3]: type 0x0, count 0x0 > out_group[0x4]: type 0x0, count 0x0 > out_group[0x5]: type 0x0, count 0x0 > out_group[0x6]: type 0x0, count 0x0 > out_group[0x7]: type 0x0, count 0x0 > nb_in_groups: 0x2 > out_group[0x0]: type 0x0, count 0x8 > out_group[0x1]: type 0x3, count 0x8 > out_group[0x2]: type 0x0, count 0x0 > out_group[0x3]: type 0x0, count 0x0 > out_group[0x4]: type 0x0, count 0x0 > out_group[0x5]: type 0x0, count 0x0 > out_group[0x6]: type 0x0, count 0x0 > out_group[0x7]: type 0x0, count 0x0 > nb_midi_out: 0x1 > nb_midi_in: 0x1 > max_sample_rate: 0x17700 > min_sample_rate: 0x7D00 > dsp_version: 0x0 > arm_version: 0x5050000 > mixer_playback_channels: 0x10 > mixer_capture_channels: 0x10 > fpga_version: 0x4010100 > nb_1394_playback_channels_2x: 0xC > nb_1394_capture_channels_2x: 0xC > nb_1394_playback_channels_4x: 0xA > nb_1394_capture_channels_4x: 0xA > > Kind Regards > > > Takashi Sakamoto > o-takashi(a)sakamocchi.jp > > (2013年03月09日 06:49), Doug McLain wrote: >> I have been building and using the fireworks driver from >> alsa-kprivate/fireworks for a few years now with my AF2 and AF12. The >> driver works great for me, although every so often after a kernel >> update, compilation breaks and I have to hack at it a little bit to get >> it to build again. >> >> Are there any plans to include this driver into the kernel? >

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[alsa-devel] plans for fireworks driver
by Doug McLain 08 Mar '13

08 Mar '13

I have been building and using the fireworks driver from alsa-kprivate/fireworks for a few years now with my AF2 and AF12. The driver works great for me, although every so often after a kernel update, compilation breaks and I have to hack at it a little bit to get it to build again. Are there any plans to include this driver into the kernel?

1 0

[alsa-devel] [PATCH 0/5] ASoC: OMAP2+: Update Audio IP with sDMA binding for DT boot
by Sebastien Guiriec 08 Mar '13

08 Mar '13

This patch serie is converting OMAP Audio IPs in order to use OMAP sDMA DT binding in case of DT boot on OMAP2+ platforms. It is depending on the next patches: [1] http://www.spinics.net/lists/arm-kernel/msg227836.html [2] http://www.mail-archive.com/linux-omap@vger.kernel.org/msg84668.html It has been tested on: - OMAP5 (sEVM/panda)in DT boot (McPDM/DMIC). - OMAP4 (SDP4430/Panda) in both DT/non DT boot (McPDM/DMIC) - OMAP3 (Beagle XM) in both DT/non DT boot (McBSP) Sebastien Guiriec (5): ASoC: omap-pcm: Move to generic DMA for DT binding ASoC: omap-dmic: Update driver for DMA DT binding. ASoC: omap-mcpdm: Update driver for DMA DT binding. ASoC: omap-mcbsp:Update driver for DMA DT binding. ARM: dts: OMAP2+: Add SDMA Audio IPs bindings arch/arm/boot/dts/omap2420.dtsi | 6 ++++++ arch/arm/boot/dts/omap2430.dtsi | 15 +++++++++++++++ arch/arm/boot/dts/omap3.dtsi | 15 +++++++++++++++ arch/arm/boot/dts/omap4.dtsi | 17 +++++++++++++++++ arch/arm/boot/dts/omap5.dtsi | 14 ++++++++++++++ sound/soc/omap/mcbsp.c | 26 ++++++++++++++++---------- sound/soc/omap/omap-dmic.c | 15 +++++++++------ sound/soc/omap/omap-mcpdm.c | 22 ++++++++++++++-------- sound/soc/omap/omap-pcm.c | 12 ++++++++++-- sound/soc/omap/omap-pcm.h | 1 + 10 files changed, 117 insertions(+), 26 deletions(-) -- 1.7.10.4

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[alsa-devel] [PATCH v2] ALSA: ASoC: add codec driver for TI TAS5086
by Daniel Mack 08 Mar '13

08 Mar '13

This patch adds a driver for TI's TA5086 6-channel PWM processor. This chip has a very unusual register layout, specifically because the registers are of unequal size, and multi-byte registers require bulk writes to take effect. Regmap does not support these kind of mappings. Currently, the driver does not touch any of the registers >= 0x20, so it doesn't matter, because the register map is mapped to an 8-bit array. In case more features will be added in the future that require access to higher registers, the entire regmap H/W I/O routines have to be open-coded. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- v2: * define clock indices in a header * switched to switch-case in some places * dropped empty suspend function * moved device id checks and gpio allocations to i2c level * switched to mute_stream() .../devicetree/bindings/sound/ti,tas5086.txt | 32 ++ include/sound/tas5086.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5086.c | 601 +++++++++++++++++++++ 5 files changed, 646 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas5086.txt create mode 100644 include/sound/tas5086.h create mode 100644 sound/soc/codecs/tas5086.c diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt new file mode 100644 index 0000000..8ea4f5b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -0,0 +1,32 @@ +Texas Instruments TAS5086 6-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,tas5086". + - reg: The i2c address. Should contain <0x1b>. + +Optional properties: + + - reset-gpio: A GPIO spec to define which pin is connected to the + chip's !RESET pin. If specified, the driver will + assert a hardware reset at probe time. + + - ti,charge-period: This property should contain the time in microseconds + that closely matches the external single-ended + split-capacitor charge period. The hardware chip + waits for this period of time before starting the + PWM signals. This helps reduce pops and clicks. + + When not specified, the hardware default of 1300ms + is retained. + +Examples: + + i2c_bus { + tas5086@1b { + compatible = "ti,tas5086"; + reg = <0x1b>; + reset-gpio = <&gpio 23 0>; + ti,charge-period = <156000>; + }; + }; diff --git a/include/sound/tas5086.h b/include/sound/tas5086.h new file mode 100644 index 0000000..aac481b --- /dev/null +++ b/include/sound/tas5086.h @@ -0,0 +1,7 @@ +#ifndef _SND_SOC_CODEC_TAS5086_H_ +#define _SND_SOC_CODEC_TAS5086_H_ + +#define TAS5086_CLK_IDX_MCLK 0 +#define TAS5086_CLK_IDX_SCLK 1 + +#endif /* _SND_SOC_CODEC_TAS5086_H_ */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 500f666..18fea10 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -64,6 +64,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS + select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C @@ -324,6 +325,9 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate +config SND_SOC_TAS5086 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3a7ec1c..b9e41c9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -56,6 +56,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o +snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -179,6 +180,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o +obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c new file mode 100644 index 0000000..008bea4 --- /dev/null +++ b/sound/soc/codecs/tas5086.c @@ -0,0 +1,601 @@ +/* + * TAS5086 ASoC codec driver + * + * Copyright (c) 2013 Daniel Mack <zonque(a)gmail.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * TODO: + * - implement DAPM and input muxing + * - implement modulation limit + * - implement non-default PWM start + * + * Note that this chip has a very unusual register layout, specifically + * because the registers are of unequal size, and multi-byte registers + * require bulk writes to take effect. Regmap does not support that kind + * of devices. + * + * Currently, the driver does not touch any of the registers >= 0x20, so + * it doesn't matter because the entire map can be accessed as 8-bit + * array. In case more features will be added in the future + * that require access to higher registers, the entire regmap H/W I/O + * routines have to be open-coded. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/tas5086.h> + +#define TAS5086_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE) + +#define TAS5086_PCM_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +/* + * TAS5086 registers + */ +#define TAS5086_CLOCK_CONTROL 0x00 /* Clock control register */ +#define TAS5086_CLOCK_RATE(val) (val << 5) +#define TAS5086_CLOCK_RATE_MASK (0x7 << 5) +#define TAS5086_CLOCK_RATIO(val) (val << 2) +#define TAS5086_CLOCK_RATIO_MASK (0x7 << 2) +#define TAS5086_CLOCK_SCLK_RATIO_48 (1 << 1) +#define TAS5086_CLOCK_VALID (1 << 0) + +#define TAS5086_DEEMPH_MASK 0x03 +#define TAS5086_SOFT_MUTE_ALL 0x3f + +#define TAS5086_DEV_ID 0x01 /* Device ID register */ +#define TAS5086_ERROR_STATUS 0x02 /* Error status register */ +#define TAS5086_SYS_CONTROL_1 0x03 /* System control register 1 */ +#define TAS5086_SERIAL_DATA_IF 0x04 /* Serial data interface register */ +#define TAS5086_SYS_CONTROL_2 0x05 /* System control register 2 */ +#define TAS5086_SOFT_MUTE 0x06 /* Soft mute register */ +#define TAS5086_MASTER_VOL 0x07 /* Master volume */ +#define TAS5086_CHANNEL_VOL(X) (0x08 + (X)) /* Channel 1-6 volume */ +#define TAS5086_VOLUME_CONTROL 0x09 /* Volume control register */ +#define TAS5086_MOD_LIMIT 0x10 /* Modulation limit register */ +#define TAS5086_PWM_START 0x18 /* PWM start register */ +#define TAS5086_SURROUND 0x19 /* Surround register */ +#define TAS5086_SPLIT_CAP_CHARGE 0x1a /* Split cap charge period register */ +#define TAS5086_OSC_TRIM 0x1b /* Oscillator trim register */ +#define TAS5086_BKNDERR 0x1c + +/* + * Default TAS5086 power-up configuration + */ +static const struct reg_default tas5086_reg_defaults[] = { + { 0x00, 0x6c }, + { 0x01, 0x03 }, + { 0x02, 0x00 }, + { 0x03, 0xa0 }, + { 0x04, 0x05 }, + { 0x05, 0x60 }, + { 0x06, 0x00 }, + { 0x07, 0xff }, + { 0x08, 0x30 }, + { 0x09, 0x30 }, + { 0x0a, 0x30 }, + { 0x0b, 0x30 }, + { 0x0c, 0x30 }, + { 0x0d, 0x30 }, + { 0x0e, 0xb1 }, + { 0x0f, 0x00 }, + { 0x10, 0x02 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x3f }, + { 0x19, 0x00 }, + { 0x1a, 0x18 }, + { 0x1b, 0x82 }, + { 0x1c, 0x05 }, +}; + +static bool tas5086_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x0f) || (reg >= 0x11 && reg <= 0x17)); +} + +static bool tas5086_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS5086_DEV_ID: + case TAS5086_ERROR_STATUS: + return true; + } + + return false; +} + +static bool tas5086_writeable_reg(struct device *dev, unsigned int reg) +{ + return tas5086_accessible_reg(dev, reg) && (reg != TAS5086_DEV_ID); +} + +struct tas5086_private { + struct regmap *regmap; + unsigned int mclk, sclk; + unsigned int format; + bool deemph; + /* Current sample rate for de-emphasis control */ + int rate; + /* GPIO driving Reset pin, if any */ + int gpio_nreset; +}; + +static int tas5086_deemph[] = { 0, 32000, 44100, 48000 }; + +static int tas5086_set_deemph(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int i, val = 0; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(tas5086_deemph); i++) + if (tas5086_deemph[i] == priv->rate) + val = i; + + return regmap_update_bits(priv->regmap, TAS5086_SYS_CONTROL_1, + TAS5086_DEEMPH_MASK, val); +} + +static int tas5086_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int tas5086_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return tas5086_set_deemph(codec); +} + + +static int tas5086_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case TAS5086_CLK_IDX_MCLK: + priv->mclk = freq; + break; + case TAS5086_CLK_IDX_SCLK: + priv->sclk = freq; + break; + } + + return 0; +} + +static int tas5086_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The TAS5086 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + /* we need to refer to the data format from hw_params() */ + priv->format = format; + + return 0; +} + +static const int tas5086_sample_rates[] = { + 32000, 38000, 44100, 48000, 88200, 96000, 176400, 192000 +}; + +static const int tas5086_ratios[] = { + 64, 128, 192, 256, 384, 512 +}; + +static int index_in_array(const int *array, int len, int needle) +{ + int i; + + for (i = 0; i < len; i++) + if (array[i] == needle) + return i; + + return -ENOENT; +} + +static int tas5086_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int ret; + + priv->rate = params_rate(params); + + /* Look up the sample rate and refer to the offset in the list */ + val = index_in_array(tas5086_sample_rates, + ARRAY_SIZE(tas5086_sample_rates), priv->rate); + + if (val < 0) { + dev_err(codec->dev, "Invalid sample rate\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATE_MASK, + TAS5086_CLOCK_RATE(val)); + if (ret < 0) + return ret; + + /* MCLK / Fs ratio */ + val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios), + priv->mclk / priv->rate); + if (val < 0) { + dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATIO_MASK, + TAS5086_CLOCK_RATIO(val)); + if (ret < 0) + return ret; + + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_SCLK_RATIO_48, + (priv->sclk == 48 * priv->rate) ? + TAS5086_CLOCK_SCLK_RATIO_48 : 0); + if (ret < 0) + return ret; + + /* + * The chip has a very unituitive register mapping and muxes information + * about data format and sample depth into the same register, but not on + * a logical bit-boundary. Hence, we have to refer to the format passed + * in the set_dai_fmt() callback and set up everything from here. + * + * First, determine the 'base' value, using the format ... + */ + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x03; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x06; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + /* ... then add the offset for the sample bit depth. */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val += 0; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val += 1; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + val += 2; + break; + default: + dev_err(codec->dev, "Invalid bit width\n"); + return -EINVAL; + }; + + ret = regmap_write(priv->regmap, TAS5086_SERIAL_DATA_IF, val); + if (ret < 0) + return ret; + + /* clock is considered valid now */ + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_VALID, TAS5086_CLOCK_VALID); + if (ret < 0) + return ret; + + return tas5086_set_deemph(codec); +} + +static int tas5086_mute_stream(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + if (mute) + val = TAS5086_SOFT_MUTE_ALL; + + return regmap_write(priv->regmap, TAS5086_SOFT_MUTE, val); +} + +/* TAS5086 controls */ +static const DECLARE_TLV_DB_SCALE(tas5086_dac_tlv, -10350, 50, 1); + +static const struct snd_kcontrol_new tas5086_controls[] = { + SOC_SINGLE_TLV("Master Playback Volume", TAS5086_MASTER_VOL, + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume", + TAS5086_CHANNEL_VOL(0), TAS5086_CHANNEL_VOL(1), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume", + TAS5086_CHANNEL_VOL(2), TAS5086_CHANNEL_VOL(3), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume", + TAS5086_CHANNEL_VOL(4), TAS5086_CHANNEL_VOL(5), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + tas5086_get_deemph, tas5086_put_deemph), +}; + +static const struct snd_soc_dai_ops tas5086_dai_ops = { + .hw_params = tas5086_hw_params, + .set_sysclk = tas5086_set_dai_sysclk, + .set_fmt = tas5086_set_dai_fmt, + .mute_stream = tas5086_mute_stream, +}; + +static struct snd_soc_dai_driver tas5086_dai = { + .name = "tas5086-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 6, + .rates = TAS5086_PCM_RATES, + .formats = TAS5086_PCM_FORMATS, + }, + .ops = &tas5086_dai_ops, +}; + +#ifdef CONFIG_PM +static int tas5086_soc_resume(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* Restore codec state */ + return regcache_sync(priv->regmap); +} +#else +#define tas5086_soc_resume NULL +#endif /* CONFIG_PM */ + +#ifdef CONFIG_OF +static const struct of_device_id tas5086_dt_ids[] = { + { .compatible = "ti,tas5086", }, + { } +}; +MODULE_DEVICE_TABLE(of, tas5086_dt_ids); +#endif + +/* charge period values in microseconds */ +static const int tas5086_charge_period[] = { + 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200, + 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000, + 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000, +}; + +static int tas5086_probe(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int charge_period = 1300000; /* hardware default is 1300 ms */ + int i, ret; + + if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { + struct device_node *of_node = codec->dev->of_node; + of_property_read_u32(of_node, "ti,charge-period", &charge_period); + } + + /* lookup and set split-capacitor charge period */ + if (charge_period == 0) { + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); + } else { + i = index_in_array(tas5086_charge_period, + ARRAY_SIZE(tas5086_charge_period), + charge_period); + if (i >= 0) + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, + i + 0x08); + else + dev_warn(codec->dev, + "Invalid split-cap charge period of %d ns.\n", + charge_period); + } + + /* enable factory trim */ + ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00); + if (ret < 0) + return ret; + + /* start all channels */ + ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20); + if (ret < 0) + return ret; + + /* set master volume to 0 dB */ + ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); + if (ret < 0) + return ret; + + /* mute all channels for now */ + ret = regmap_write(priv->regmap, TAS5086_SOFT_MUTE, + TAS5086_SOFT_MUTE_ALL); + if (ret < 0) + return ret; + + return 0; +} + +static int tas5086_remove(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(priv->gpio_nreset)) + /* Set codec to the reset state */ + gpio_set_value(priv->gpio_nreset, 0); + + return 0; +}; + +static struct snd_soc_codec_driver soc_codec_dev_tas5086 = { + .probe = tas5086_probe, + .remove = tas5086_remove, + .resume = tas5086_soc_resume, + .controls = tas5086_controls, + .num_controls = ARRAY_SIZE(tas5086_controls), +}; + +static const struct i2c_device_id tas5086_i2c_id[] = { + { "tas5086", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id); + +static const struct regmap_config tas5086_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(tas5086_reg_defaults), + .reg_defaults = tas5086_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5086_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5086_volatile_reg, + .writeable_reg = tas5086_writeable_reg, + .readable_reg = tas5086_accessible_reg, +}; + +static int tas5086_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct tas5086_private *priv; + struct device *dev = &i2c->dev; + int gpio_nreset = -EINVAL; + int i, ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(i2c, &tas5086_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&i2c->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, priv); + + if (of_match_device(of_match_ptr(tas5086_dt_ids), dev)) { + struct device_node *of_node = dev->of_node; + gpio_nreset = of_get_named_gpio(of_node, "reset-gpio", 0); + } + + if (gpio_is_valid(gpio_nreset)) + if (devm_gpio_request(dev, gpio_nreset, "TAS5086 Reset")) + gpio_nreset = -EINVAL; + + if (gpio_is_valid(gpio_nreset)) { + /* Reset codec - minimum assertion time is 400ns */ + gpio_direction_output(gpio_nreset, 0); + udelay(1); + gpio_set_value(gpio_nreset, 1); + + /* Codec needs ~15ms to wake up */ + msleep(15); + } + + priv->gpio_nreset = gpio_nreset; + + /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ + ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); + if (ret < 0) + return ret; + + if (i != 0x3) { + dev_err(dev, + "Failed to identify TAS5086 codec (got %02x)\n", i); + return -ENODEV; + } + + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, + &tas5086_dai, 1); +} + +static int tas5086_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static struct i2c_driver tas5086_i2c_driver = { + .driver = { + .name = "tas5086", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas5086_dt_ids), + }, + .id_table = tas5086_i2c_id, + .probe = tas5086_i2c_probe, + .remove = tas5086_i2c_remove, +}; + +static int __init tas5086_modinit(void) +{ + return i2c_add_driver(&tas5086_i2c_driver); +} +module_init(tas5086_modinit); + +static void __exit tas5086_modexit(void) +{ + i2c_del_driver(&tas5086_i2c_driver); +} +module_exit(tas5086_modexit); + +MODULE_AUTHOR("Daniel Mack <zonque(a)gmail.com>"); +MODULE_DESCRIPTION("Texas Instruments TAS5086 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); -- 1.8.1.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: Drop unused "dapm" field form soc_enum struct
by Lars-Peter Clausen 08 Mar '13

08 Mar '13

This field was added in commit 2e72f8e ("ASoC: New enum type: value_enum"), but has never been used since. Considering that the soc_enum struct is usually shared between all instances of a CODEC, it also doesn't make much sense to have a pointer to DAPM specific data in it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a6a059c..c84062b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1086,7 +1086,6 @@ struct soc_enum { unsigned int mask; const char * const *texts; const unsigned int *values; - void *dapm; }; /* codec IO */ -- 1.8.0

2 2

[alsa-devel] [PATCH] ALSA: ASoC: Add codec driver for AK5386
by Daniel Mack 08 Mar '13

08 Mar '13

Adds a driver for Asahi Kasei's AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC. The device has no control port interface but an optional RESET/PDN GPIO pin. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Documentation/devicetree/bindings/sound/ak5386.txt | 19 +++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak5386.c | 152 +++++++++++++++++++++ 4 files changed, 177 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak5386.txt create mode 100644 sound/soc/codecs/ak5386.c diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt new file mode 100644 index 0000000..dc3914f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak5386.txt @@ -0,0 +1,19 @@ +AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC + +This device has no control interface. + +Required properties: + + - compatible : "asahi-kasei,ak5386" + +Optional properties: + + - reset-gpio : a GPIO spec for the reset/power down pin. + If specified, it will be deasserted at probe time. + +Example: + +spdif: ak5386@0 { + compatible = "asahi-kasei,ak5386"; + reset-gpio = <&gpio0 23>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 86b3524..18fea10 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_AK5386 select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC @@ -204,6 +205,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_AK5386 + tristate + config SND_SOC_ALC5623 tristate config SND_SOC_ALC5632 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8077bc2..b9e41c9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -14,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o @@ -138,6 +139,7 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_AK5386) += snd-soc-ak5386.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c new file mode 100644 index 0000000..1f30398 --- /dev/null +++ b/sound/soc/codecs/ak5386.c @@ -0,0 +1,152 @@ +/* + * ALSA SoC driver for + * Asahi Kasei AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC + * + * (c) 2013 Daniel Mack <zonque(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_device.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/initval.h> + +struct ak5386_priv { + int reset_gpio; +}; + +static struct snd_soc_codec_driver soc_codec_ak5386; + +static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + format &= SND_SOC_DAIFMT_FORMAT_MASK; + if (format != SND_SOC_DAIFMT_LEFT_J && + format != SND_SOC_DAIFMT_I2S) { + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + return 0; +} + +static int ak5386_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + + /* + * From the datasheet: + * + * All external clocks (MCLK, SCLK and LRCK) must be present unless + * PDN pin = “L”. If these clocks are not provided, the AK5386 may + * draw excess current due to its use of internal dynamically + * refreshed logic. If the external clocks are not present, place + * the AK5386 in power-down mode (PDN pin = “L”). + */ + + if (gpio_is_valid(priv->reset_gpio)) + gpio_set_value(priv->reset_gpio, 1); + + return 0; +} + +static int ak5386_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(priv->reset_gpio)) + gpio_set_value(priv->reset_gpio, 0); + + return 0; +} + +static const struct snd_soc_dai_ops ak5386_dai_ops = { + .set_fmt = ak5386_set_dai_fmt, + .hw_params = ak5386_hw_params, + .hw_free = ak5386_hw_free, +}; + +static struct snd_soc_dai_driver ak5386_dai = { + .name = "ak5386-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + }, + .ops = &ak5386_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id ak5386_dt_ids[] = { + { .compatible = "asahi-kasei,ak5386", }, + { } +}; +MODULE_DEVICE_TABLE(of, ak5386_dt_ids); +#endif + +static int ak5386_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct ak5386_priv *priv; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->reset_gpio = -EINVAL; + dev_set_drvdata(dev, priv); + + if (of_match_device(of_match_ptr(ak5386_dt_ids), dev)) + priv->reset_gpio = of_get_named_gpio(dev->of_node, + "reset-gpio", 0); + + if (gpio_is_valid(priv->reset_gpio)) + if (devm_gpio_request_one(dev, priv->reset_gpio, + GPIOF_OUT_INIT_LOW, + "AK5386 Reset")) + priv->reset_gpio = -EINVAL; + + return snd_soc_register_codec(dev, &soc_codec_ak5386, + &ak5386_dai, 1); +} + +static int ak5386_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver ak5386_driver = { + .probe = ak5386_probe, + .remove = ak5386_remove, + .driver = { + .name = "ak5386", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ak5386_dt_ids), + }, +}; + +module_platform_driver(ak5386_driver); + +MODULE_DESCRIPTION("ASoC driver for AK5386 ADC"); +MODULE_AUTHOR("Daniel Mack <zonque(a)gmail.com>"); +MODULE_LICENSE("GPL"); -- 1.8.1.4

2 1

[alsa-devel] [PATCH v2 00/12] ARM: mxs: move to generic DMA device tree binding
by Shawn Guo 08 Mar '13

08 Mar '13

The series converts mxs-dma and its clients to generic DMA device tree binding/helper. I published a branch below to ease people who is willing to help testing the series. git://git.linaro.org/people/shawnguo/linux-2.6.git devel/mxs-dma-dt Changes since v1: - Remove the use of of_dma_filter_info which should not be needed by a customized xlate function - Correct dma-names to have it mean the channel function in client devices - Add a new dmaengine_pcm API snd_dmaengine_generic_pcm_open() for users that support generic DMA device tree binding - Remove the old fsl,<module>-dma-channel properties from device tree Subsystem maintainers, I need your ACKs to have the series go via arm-soc as a whole if the patches look good to you. Thanks, Shawn Shawn Guo (12): ARM: dts: add generic DMA device tree binding for mxs-dma dma: mxs-dma: use devm_* managed functions dma: mxs-dma: move to generic device tree binding mmc: mxs-mmc: move to use generic DMA helper spi: mxs-spi: move to use generic DMA helper i2c: i2c-mxs: move to use generic DMA helper mtd: gpmi: move to use generic DMA helper serial: mxs-auart: move to use generic DMA helper ASoC: dmaengine_pcm: add snd_dmaengine_generic_pcm_open() ASoC: mxs: move to use generic DMA helper dma: mxs-dma: remove code left from generic DMA binding conversion ARM: dts: remove old DMA binding data from client nodes .../devicetree/bindings/dma/fsl-mxs-dma.txt | 49 +++++++- Documentation/devicetree/bindings/i2c/i2c-mxs.txt | 12 +- Documentation/devicetree/bindings/mmc/mxs-mmc.txt | 12 +- .../devicetree/bindings/mtd/gpmi-nand.txt | 17 +-- .../devicetree/bindings/sound/mxs-saif.txt | 18 +-- Documentation/devicetree/bindings/spi/mxs-spi.txt | 12 +- .../bindings/tty/serial/fsl-mxs-auart.txt | 16 ++- arch/arm/boot/dts/imx23.dtsi | 57 +++++++-- arch/arm/boot/dts/imx28.dtsi | 104 +++++++++++----- arch/arm/boot/dts/imx6qdl.dtsi | 12 +- drivers/dma/mxs-dma.c | 128 +++++++++++--------- drivers/i2c/busses/i2c-mxs.c | 40 +----- drivers/mmc/host/mxs-mmc.c | 48 +------- drivers/mtd/nand/gpmi-nand/gpmi-nand.c | 51 +------- drivers/mtd/nand/gpmi-nand/gpmi-nand.h | 3 +- drivers/spi/spi-mxs.c | 60 ++------- drivers/tty/serial/mxs-auart.c | 52 +------- include/linux/fsl/mxs-dma.h | 20 --- include/linux/spi/mxs-spi.h | 4 +- include/sound/dmaengine_pcm.h | 2 + sound/soc/mxs/mxs-pcm.c | 42 +------ sound/soc/mxs/mxs-pcm.h | 5 - sound/soc/mxs/mxs-saif.c | 28 +---- sound/soc/mxs/mxs-saif.h | 1 - sound/soc/soc-dmaengine-pcm.c | 39 ++++++ 25 files changed, 368 insertions(+), 464 deletions(-) delete mode 100644 include/linux/fsl/mxs-dma.h -- 1.7.9.5

4 10

[alsa-devel] HDA regression on Fujitsu S7020 laptop (ALC260 codec)
by Jonathan Woithe 07 Mar '13

07 Mar '13

Having been running kernel 3.3.5 for a while, I decided to upgrade to 3.7.1 tonight. Unfortunately there have been a number of HDA-related regressions between these two kernels when running on a Fujitsu S7020 laptop which are clearly related to the HDA rewrite which has been going on recently. This laptop utilises the Realtek ALC260 codec. Testing was initially done with alsa-lib 1.0.24 and alsa-utils 1.0.24. I upgraded to version 1.0.26 of both but the behaviour remained the same under 3.3.5 and 3.7.1. Under 3.3.5, sound works perfectly. Under 3.7.1 there are a number of regressions: - there is no alsamixer control for the internal speaker. I have a "PCM" volume control which appears to do nothing. Previously there was a speaker control, the PCM control affected every output (from memory), as did the master volume. - no audio comes out of the internal speaker regardless of the "auto mute" setting. - there are no pin mode (re-tasking) controls for the mic or headphone jack in alsamixer. On this laptop the mic jack should have various bias options as well as line-in. The headphone jack should include headphone (aka output amplifier on), line-out (output amplifier off), line-in and mic options. The second ADC does show up in alsamixer as a second capture device, but probably due to the lack of retasking support it is not possible to select the headphone jack as an input option. Audio does come out of the headphone jack. However, as noted above it is not possible to switch this from headphone to line-out. The volume of the headphone output is affected by the "master" and "headphone" volume controls. The inability to set the pin modes is a showstopper for me - I need to be able to do this and thus will be stuck on older kernels until this is fixed. At the end of this post I have included the /proc/asound/card0/codec#0 output under 3.3.5 (which works well) and 3.7.1 (which is broken as described above). What needs to be done to restore the functionality for the S7020 that has been working for years? It seems that some form of model quirk needs to be ported into the new HDA framework from earlier kernels, but since this sort of thing appears to have been removed with the rewrite (at least for realtek codecs) it's difficult to see where to begin. Regards jonathan Output from /proc/asound/card0/codec#0 under kernel 3.3.5: Codec: Realtek ALC260 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0260 Subsystem Id: 0x10cf0000 Revision Id: 0x100400 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC260 Analog", type="Audio", device=0 Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x04 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Control: name="Capture Switch", index=0, device=0 Control: name="Capture Volume", index=0, device=0 Device: name="ALC260 Analog", type="Audio", device=0 Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 7 0x12* 0x13 0x14 0x15 0x16 0x0f 0x10 Node 0x05 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Input Source", index=1, device=0 Control: name="Capture Switch", index=1, device=0 Control: name="Capture Volume", index=1, device=0 Device: name="ALC260 Analog", type="Audio", device=2 Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 8 0x12* 0x13 0x14 0x15 0x16 0x07 0x0f 0x10 Node 0x06 [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x19 Node 0x07 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="CD Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="CD Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Mic/Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic/Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x03 0x03] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 8 0x12 0x13 0x14 0x15 0x16 0x17 0x0f 0x10 Node 0x08 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x27 0x27] Connection: 2 0x02 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x02 0x07 Node 0x0a [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80] [0x80] Amp-Out caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=0 Amp-Out vals: [0x00] Connection: 2 0x02 0x07 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0e [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0f [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003f: IN OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x08 Node 0x10 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003f: IN OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0xe4011110: [Both] Line Out at Sep Right Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Connection: 1 0x09 Node 0x11 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 1 0x0a Node 0x12 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Mic/Line Jack Mode", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x03a11820: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0b Node 0x13 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x24811121: [Jack] Line In at Sep Right Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Headphone Jack Mode", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x16 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x88331122: [Fixed] CD at Ext Drive Bar Conn = ATAPI, Color = Black DefAssociation = 0x2, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x17 [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0xb7931123: [Fixed] Aux at Oth Mobile-In Conn = ATAPI, Color = Black DefAssociation = 0x2, Sequence = 0x3 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x18 [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x19 [Pin Complex] wcaps 0x400280: Mono Digital Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Node 0x1a [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=13 Node 0x1b [Volume Knob Widget] wcaps 0x600080: Mono Volume-Knob: delta=0, steps=64, direct=0, val=0 Unsolicited: tag=00, enabled=0 Connection: 0 Output from /proc/asound/card0/codec#0 under kernel 3.7.1: Codec: Realtek ALC260 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0260 Subsystem Id: 0x10cf0000 Revision Id: 0x100400 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC260 Analog", type="Audio", device=0 Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x04 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Control: name="Capture Switch", index=0, device=0 Control: name="Capture Volume", index=0, device=0 Device: name="ALC260 Analog", type="Audio", device=0 Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] [0x0c 0x0c] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 7 0x12* 0x13 0x14 0x15 0x16 0x0f 0x10 Node 0x05 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Input Source", index=1, device=0 Control: name="Capture Switch", index=1, device=0 Control: name="Capture Volume", index=1, device=0 Device: name="ALC260 Analog", type="Audio", device=2 Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 8 0x12* 0x13 0x14 0x15 0x16 0x07 0x0f 0x10 Node 0x06 [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x19 Node 0x07 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="CD Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="CD Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Amp-In caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x03 0x03] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 8 0x12 0x13 0x14 0x15 0x16 0x17 0x0f 0x10 Node 0x08 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x27 0x27] Connection: 2 0x02 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="PCM Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x33 0x33] Connection: 2 0x02 0x07 Node 0x0a [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] [0x80] Amp-Out caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=0 Amp-Out vals: [0x23] Connection: 2 0x02 0x07 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0e [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x08* 0x09 Node 0x0f [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003f: IN OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x08 Node 0x10 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="PCM Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003f: IN OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0xe4011110: [Both] Line Out at Sep Right Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x09 Node 0x11 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x0a Node 0x12 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Mic Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x03a11820: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=03, enabled=1 Connection: 1 0x0b Node 0x13 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Line Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x24811121: [Jack] Line In at Sep Right Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 1 0x0d Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x16 [Pin Complex] wcaps 0x400001: Stereo Control: name="CD Phantom Jack", index=0, device=0 Pincap 0x00000020: IN Pin Default 0x88331122: [Fixed] CD at Ext Drive Bar Conn = ATAPI, Color = Black DefAssociation = 0x2, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x17 [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0xb7931123: [Fixed] Aux at Oth Mobile-In Conn = ATAPI, Color = Black DefAssociation = 0x2, Sequence = 0x3 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x18 [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x19 [Pin Complex] wcaps 0x400280: Mono Digital Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Node 0x1a [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=13 Node 0x1b [Volume Knob Widget] wcaps 0x600080: Mono Volume-Knob: delta=0, steps=64, direct=0, val=0 Unsolicited: tag=00, enabled=0 Connection: 0

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[alsa-devel] [PATCH] ALSA: vmaster: Fix slave change notification
by Takashi Iwai 07 Mar '13

07 Mar '13

When a value of a vmaster slave control is changed, the ctl change notification is sometimes ignored. This happens when the master control overrides, e.g. when the corresponding master control is muted. The reason is that slave_put() returns the value of the actual slave put callback, and it doesn't reflect the virtual slave value change. This patch fixes the function just to return 1 whenever a slave value is changed. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/vmaster.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 8575861..0097f36 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -213,7 +213,10 @@ static int slave_put(struct snd_kcontrol *kcontrol, } if (!changed) return 0; - return slave_put_val(slave, ucontrol); + err = slave_put_val(slave, ucontrol); + if (err < 0) + return err; + return 1; } static int slave_tlv_cmd(struct snd_kcontrol *kcontrol, -- 1.8.1.4

1 0