July 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] AC97 on S3C6410
by Marco Sinigaglia 13 Jul '11

13 Jul '11

Hi All, I am still trying to implement the AC97, on S3C6410, using the WM9714. I have done some big step forward but I am now hitting another issue. The kernel can see the WM9714: <6>ALSA device list: <6> #0: SMDK (WM9713) Anyway, after that, i cannot see anymore activity on the AC97 bus. I got, indeed, these errors: <3>s3c-ac97: req addr = 12, rep addr = 00 [...] <3>s3c-ac97: req addr = 12, rep addr = 00 When I try to play something, i got this (I enabled the DEBUG on dma.c): <7>asoc: AC97 HiFi <-> s3c-ac97 info: <7>asoc: rate mask 0xd6 <7>asoc: min ch 2 max ch 2 <7>asoc: min rate 8000 max rate 48000 <7>dma22: s3c2410_request_dma: client=AC97 PCMOut, dev=(null) <7>DMA8: 00000000->00000000 L 00000000 C 00000000,00000000 S 00000000 <7>s3c2410_dma_request: channel initialised, c06779c8 <7>s3c2410_dma_devconfig: channel 22, source 1, dev 7f001018, chan c06779c8 <7>s3c2410_dma_devconfig: peripheral 6 <7>s3c2410_dma_devconfig: config 0000c980 <7>DMA8: 00000000->00000000 L 00000000 C 00000000,00000000 S 0000c980 <7>s3c64xx_dma_flush: flushing channel <7>s3c2410_dma_enqueue: buff c9dcfac0, dp 5e740000 lli (ffcd6000, 57241000) 8192 <7>enquing onto empty channel <7>s3c64xx_lli_to_regs: LLI ffcd6000 => regs <7>LLI[ffcd6000] 5e740000->7f001018, NL 00000000 C 96480000,00000800 <7>LLI[ffcd6000] 5e740000->7f001018, NL 00000000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 00000000 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfac0, curr c9dcfac0, end c9dcfac0 <7>s3c2410_dma_enqueue: buff c9dcfa60, dp 5e742000 lli (ffcd6020, 57241020) 8192 <7>enquing onto channel <7>LLI[ffcd6000] 5e740000->7f001018, NL 57241020 C 96480000,00000800 <7>LLI[ffcd6020] 5e742000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcfa60 <7>s3c2410_dma_enqueue: buff c9dcfa40, dp 5e744000 lli (ffcd6040, 57241040) 8192 <7>enquing onto channel <7>LLI[ffcd6020] 5e742000->7f001018, NL 57241040 C 96480000,00000800 <7>LLI[ffcd6040] 5e744000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcfa40 <7>s3c2410_dma_enqueue: buff c9dcfa20, dp 5e746000 lli (ffcd6060, 57241060) 8192 <7>enquing onto channel <7>LLI[ffcd6040] 5e744000->7f001018, NL 57241060 C 96480000,00000800 <7>LLI[ffcd6060] 5e746000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcfa20 <7>s3c2410_dma_enqueue: buff c9dcfa00, dp 5e748000 lli (ffcd6080, 57241080) 8192 <7>enquing onto channel <7>LLI[ffcd6060] 5e746000->7f001018, NL 57241080 C 96480000,00000800 <7>LLI[ffcd6080] 5e748000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcfa00 <7>s3c2410_dma_enqueue: buff c9dcf9e0, dp 5e74a000 lli (ffcd60a0, 572410a0) 8192 <7>enquing onto channel <7>LLI[ffcd6080] 5e748000->7f001018, NL 572410a0 C 96480000,00000800 <7>LLI[ffcd60a0] 5e74a000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcf9e0 <7>s3c2410_dma_enqueue: buff c9dcf9c0, dp 5e74c000 lli (ffcd60c0, 572410c0) 8192 <7>enquing onto channel <7>LLI[ffcd60a0] 5e74a000->7f001018, NL 572410c0 C 96480000,00000800 <7>LLI[ffcd60c0] 5e74c000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcf9c0 <7>s3c2410_dma_enqueue: buff c9dcf9a0, dp 5e74e000 lli (ffcd60e0, 572410e0) 8192 <7>enquing onto channel <7>LLI[ffcd60c0] 5e74c000->7f001018, NL 572410e0 C 96480000,00000800 <7>LLI[ffcd60e0] 5e74e000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcf9a0 <7>s3c2410_dma_enqueue: buff c9dcf980, dp 5e750000 lli (ffcd6100, 57241100) 8192 <7>enquing onto channel <7>LLI[ffcd60e0] 5e74e000->7f001018, NL 57241100 C 96480000,00000800 <7>LLI[ffcd6100] 5e750000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcf980 <7>s3c2410_dma_enqueue: buff c9dcf960, dp 5e752000 lli (ffcd6120, 57241120) 8192 <7>enquing onto channel <7>LLI[ffcd6100] 5e750000->7f001018, NL 57241120 C 96480000,00000800 <7>LLI[ffcd6120] 5e752000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcf960 <7>s3c2410_dma_enqueue: buff c7242940, dp 5e754000 lli (ffcd6140, 57241140) 8192 <7>enquing onto channel <7>LLI[ffcd6120] 5e752000->7f001018, NL 57241140 C 96480000,00000800 <7>LLI[ffcd6140] 5e754000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c7242940 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <3> clearing interrupts <7>s3c64xx_dma_start: clearing interrupts <7>s3c64xx_dma_start: starting channel <7>s3c64xx_dma_start: writing config 0000c981 <7>ALSA sound/core/pcm_lib.c:1755: playback write error (DMA or IRQ trouble?) <7>ALSA sound/core/pcm_lib.c:1755: playback write error (DMA or IRQ trouble?) <7>s3c64xx_dma_stop: stopping channel <7>DMA8: 5e740010->7f001018 L 57241020 C 96480000,00000800 S 0002c981 I have the feeling that, for some reasons, the DMA is not doing its job. I have added the following line on dma.c but it does not fix the error: s3c64xx_dma_init1(8, DMACH_AC97_PCMOUT, IRQ_DMA1, 0x75100000); Can someone, please, give me some feedback? I am using kernel 2.6.35 and the head of Android. Cheers Marco Member of the CSR plc group of companies. CSR plc registered in England and Wales, registered number 4187346, registered office Churchill House, Cambridge Business Park, Cowley Road, Cambridge, CB4 0WZ, United Kingdom More information can be found at www.csr.com. Follow CSR on Twitter at http://twitter.com/CSR_PLC and read our blog at www.csr.com/blog

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[alsa-devel] [PATCH 1/5] ALSA: snd-usb: don't fail hard if mixer creation fails
by Daniel Mack 13 Jul '11

13 Jul '11

There are cards that the mixer code fails to create controls for, and currently, the whole sound device creation fails in such cases. This patch changes the code so that it just spits out warnings and continues looking at other functions of the descriptor set which might work just fine. The worst thing that can happen now is that people see soundcards that don't have all funcions supported, but rejecting the whole device is not any better. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/card.c | 5 ++--- sound/usb/mixer.c | 9 ++++++--- 2 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 220c616..8990b49 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -512,9 +512,8 @@ static void *snd_usb_audio_probe(struct usb_device *dev, if (err > 0) { /* create normal USB audio interfaces */ if (snd_usb_create_streams(chip, ifnum) < 0 || - snd_usb_create_mixer(chip, ifnum, ignore_ctl_error) < 0) { - goto __error; - } + snd_usb_create_mixer(chip, ifnum, ignore_ctl_error) < 0) + snd_printk(KERN_WARNING "snd_usb_create_mixer() failed\n"); } /* we are allowed to call snd_card_register() many times */ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c22fa76..c43dbea 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1967,7 +1967,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) state.oterm.name = desc->iTerminal; err = parse_audio_unit(&state, desc->bSourceID); if (err < 0) - return err; + snd_printk(KERN_WARNING "Unable to parse terminal with bSourceID %d\n", + desc->bSourceID); } else { /* UAC_VERSION_2 */ struct uac2_output_terminal_descriptor *desc = p; @@ -1979,12 +1980,14 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) state.oterm.name = desc->iTerminal; err = parse_audio_unit(&state, desc->bSourceID); if (err < 0) - return err; + snd_printk(KERN_WARNING "Unable to parse terminal with bSourceID %d\n", + desc->bSourceID); /* for UAC2, use the same approach to also add the clock selectors */ err = parse_audio_unit(&state, desc->bCSourceID); if (err < 0) - return err; + snd_printk(KERN_WARNING "Unable to parse clock terminal with bCSourceID %d\n", + desc->bCSourceID); } } -- 1.7.5.4

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Re: [alsa-devel] Drop 2.2/2.4 kernel support in alsa-driver
by Tom Watson 13 Jul '11

13 Jul '11

Jaroslav Kysela <perex(a)perex.cz> Wrote: >Date 12.7.2011 18:12, Takashi Iwai wrote: >> [This is resent since I haven't seen my original post due to >> alsa-project.org site down. If you have already read it, please >> disregard.] >> >> Hi, >> >> in upcoming 3.0-kernel era, I believe it's high time to drop the >> support of old 2.2/2.4 kernels in alsa-driver external builds. >> Recently I changed some PCI drivers to use KBUILD_MODNAME, and this >> isn't supported by the old kernels, i.e. it's currently broken for >> these kernels. But I have no more gut to patch alsa-driver/* codes. >> >> Is there any active user of such kernels with the latest alsa-driver? >> If not, let's clean up the whole things. >> 2.2/2.4 kernel users can still use 1.0.24, and we can release 1.0.24.x >> series if really needed. > > I agree. I believe that these kernels are using minimum users now. > Almost all embeded devices are using 2.6 kernels, too. > > Jaroslav While this IS a good idea, I would suggest that ALSA take a cue from the kernel people and bump the version up a major number. It would make lots of sense that ALSA 2.0 go with kernel version 3.0. An alternative might be to skip version 2.0 and go to ALSA 3.0 to match kernel version 3.0. This won't change any code, but it would be a little clearer. That way when you drop support for older kernels, you can easily tell what version should be used. "Just a suggestion" worth the paper it is written on. -- Tom Watson tsw(a)johana.com

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[alsa-devel] [PATCH] ASoC: Don't warn on low WM8994/58 AIFnCLKs
by Mark Brown 13 Jul '11

13 Jul '11

We can have valid but very low clocks in accessory detection modes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e5f3f50..5af1535 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -195,10 +195,6 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif) aif + 1, rate); } - if (rate && rate < 3000000) - dev_warn(codec->dev, "AIF%dCLK is %dHz, should be >=3MHz for optimal performance\n", - aif + 1, rate); - wm8994->aifclk[aif] = rate; snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1 + offset, -- 1.7.5.4

1 0

[alsa-devel] [PATCH 00/10] ARM: mxs: add audio support
by Dong Aisheng 13 Jul '11

13 Jul '11

This patch series is to add audio support based on saif for mxs platform. The driver only supports playback currently. For capturing, due to hardware limitation that we have to use two saif instances to implement full duplex (playback & recording), we need to figure out a good design to fit in ASoC as the following work. In addition, we simply just support master mode since saif tx can only work in master mode due to hw limitation(rx can work in both master & slave mode). Dong Aisheng (10): ASoc: mxs: add mxs-pcm driver ASoc: mxs: add mxs-saif driver ASoc: mxs: add mxs-sgtl5000 machine driver ASoc: mxs: add asoc configuration files ARM: mxs: add saif clock ARM: mxs: add saif device ARM: mxs: add sgtl5000 i2c device ARM: mxs: add mxs-sgtl5000 device ARM: mxs: correct the using of frac div for saif ARM: mxs-dma: include <linux/dmaengine.h> arch/arm/mach-mxs/Kconfig | 1 + arch/arm/mach-mxs/clock-mx28.c | 8 +- arch/arm/mach-mxs/devices-mx28.h | 7 + arch/arm/mach-mxs/devices/Kconfig | 3 + arch/arm/mach-mxs/devices/Makefile | 1 + arch/arm/mach-mxs/devices/platform-mxs-saif.c | 60 +++ arch/arm/mach-mxs/include/mach/audio.h | 18 + arch/arm/mach-mxs/include/mach/devices-common.h | 12 + arch/arm/mach-mxs/include/mach/dma.h | 2 + arch/arm/mach-mxs/mach-mx28evk.c | 39 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 20 + sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 377 ++++++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 ++ sound/soc/mxs/mxs-saif.c | 523 +++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 132 ++++++ sound/soc/mxs/mxs-sgtl5000.c | 181 ++++++++ 19 files changed, 1437 insertions(+), 2 deletions(-) create mode 100644 arch/arm/mach-mxs/devices/platform-mxs-saif.c create mode 100644 arch/arm/mach-mxs/include/mach/audio.h create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

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Re: [alsa-devel] Bug#631963: linux-image-3.0.0-rc4-686-pae: no sound out the rear plugs
by Ben Hutchings 13 Jul '11

13 Jul '11

On Fri, 2011-07-01 at 23:41 +0200, Toni Mueller wrote: > Hi Ben, > > On Fri, 01.07.2011 at 22:22:01 +0100, Ben Hutchings <ben(a)decadent.org.uk> wrote: > > On Fri, Jul 01, 2011 at 11:06:56PM +0200, Toni Mueller wrote: > > > On Thu, 30.06.2011 at 07:59:19 +0200, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > I guess this must be the new feature, "Auto-Mute Mode". > > > it turns out that you're right - I just had the opportunity to verify > > > that. > > > > Can we consider this closed, then? Or should that feature be > > disabled by default? > > good question - I didn't see it. Don't know offhand where to easily > find it: Upstream's changelog isn't included in the package, so... > > I'd be fine either way, but think that the feature deserves more > advertising, or, alternatively (?), being disabled by default, to > prevent bogus bug reports like mine. What do you think of this? Ben. -- Ben Hutchings All extremists should be taken out and shot.

1 0

[alsa-devel] [PATCH 0/3] OMAP4/ASoC: New McPDM driver
by Peter Ujfalusi 13 Jul '11

13 Jul '11

Hello, The following series replaces the current driver for McPDM port. The current McPDM driver design is not suitable to support both the ABE and Legacy DMA operating modes. Therefore remove most of the old driver (and mcpdm.c) and replace it with a re-designed driver to support both Legacy and ABE modes of operation. As part of the work the McPDM driver has been converted to omap device, and from now on it is going to use hwmod. Patch 2, and 3 was originated from Jorge, and Margarita, but their @ti.com mail address does not work anymore, so I took those two patch under my name, but added them as s-o-b without e-mail address. For the McPDM driver replacement, I forced git to create a complete rewrite patch to help with the review. Regards, Peter --- Misael Lopez Cruz (1): ASoC: omap-mcpdm: Replace legacy driver Peter Ujfalusi (2): OMAP: McPDM: Convert McPDM device to omap_device OMAP4: hwmod: enable mcpdm hwmod device. arch/arm/mach-omap2/devices.c | 33 ++ arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 +- arch/arm/plat-omap/devices.c | 36 -- sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcpdm.c | 470 ---------------- sound/soc/omap/mcpdm.h | 153 ----- sound/soc/omap/omap-mcpdm.c | 828 +++++++++++++++++++--------- sound/soc/omap/omap-mcpdm.h | 109 ++++ sound/soc/omap/sdp4430.c | 4 +- 9 files changed, 702 insertions(+), 935 deletions(-) delete mode 100644 sound/soc/omap/mcpdm.c delete mode 100644 sound/soc/omap/mcpdm.h rewrite sound/soc/omap/omap-mcpdm.c (42%) create mode 100644 sound/soc/omap/omap-mcpdm.h -- 1.7.6

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Re: [alsa-devel] Sound BoF in LinuxCon Europe?
by David Henningsson 13 Jul '11

13 Jul '11

On 2011-07-08 12:36, Takashi Iwai wrote: > Hi, > > just wondering how about holding a sound-BoF in LinuxCon Europe in > October? > > I know ELCE will be held at the very same time, so it may need the > adjustment of schedule. But in general what do you think? > > I know some guys will attend LPC in September, too. If the topic is > redundant again after one month, it's fine, no much need for that. > OTOH, it's a conference in Europe, so people there may have more > chance to attend (like me :-) > > An alternative is a workshop in KS2011 day 1, but I think KS suits > better for more broader topics, e.g. the discussions on the media- > notification support or the jack implementation like a thread in the > last month. I like the initiative and I'm all for having a sound BoF! Unfortunately, the timing is bad. Ubuntu's biggest event, the Ubuntu Developer Summit, is in Florida 24-28 October and my attendance is required... -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 2

[alsa-devel] Drop 2.2/2.4 kernel support in alsa-driver build?
by Takashi Iwai 12 Jul '11

12 Jul '11

[This is resent since I haven't seen my original post due to alsa-project.org site down. If you have already read it, please disregard.] Hi, in upcoming 3.0-kernel era, I believe it's high time to drop the support of old 2.2/2.4 kernels in alsa-driver external builds. Recently I changed some PCI drivers to use KBUILD_MODNAME, and this isn't supported by the old kernels, i.e. it's currently broken for these kernels. But I have no more gut to patch alsa-driver/* codes. Is there any active user of such kernels with the latest alsa-driver? If not, let's clean up the whole things. 2.2/2.4 kernel users can still use 1.0.24, and we can release 1.0.24.x series if really needed. thanks, Takashi

3 2

[alsa-devel] Playback through softvol locks mixer
by Martin Bailey 12 Jul '11

12 Jul '11

Hi, I setup an ALSA plugin chain to allow individual volume-controlled mono access to a 4-channel sound card. I'm including the asound.conf only for the first channel. Whenever I open "plug:playback1" and play audio, it appears the "playback1" softvol mixer goes locked and any attempt to change its value within the same or another application results in "Operation not permitted". Can you think of something I might be missing? pcm.playback8k { type dmix ipc_key 666 ipc_perm 0666 slave { pcm "hw:0" format S16_LE channels 4 rate 8000 buffer_size 8192 period_size 1024 } bindings.0 0 bindings.1 1 bindings.2 2 bindings.3 3 } pcm.playback8k1 { type route slave { pcm "playback8k" format S16_LE channels 4 } ttable.0.0 1 } pcm.playback1 { type softvol slave.pcm "playback8k1" control.name "playback1" control.card 0 control.count 1 min_dB -51.0 max_dB 0.0 resolution 256 } ctl.playback1 { type hw card 0 }

1 0

[alsa-devel] Specifying the default buffer-size for snd-hda-intel
by Takashi Iwai 12 Jul '11

12 Jul '11

Hi, this was a feature requested some time ago: changing the default buffer size of driver, specifically HD-audio. Now I applied the patch below as a start, found in topic/hda and master branches of sound git tree. In future, we may increase the default size once when all distros set larger values than now. Together with this change, I changed alsa-driver-build.git for allowing an integer kconfig option via --with-card-options. You can pass like --with-card-options=hda-prealloc-size=2048 Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Add Kconfig for the default buffer size Add a Kconfig entry to specify the default buffer size. Distros using PulseAudio can choose a larger value here. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 13 +++++++++++++ sound/pci/hda/hda_intel.c | 8 +++++++- 2 files changed, 20 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 70762fc..1f1a4ae 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -14,6 +14,19 @@ menuconfig SND_HDA_INTEL if SND_HDA_INTEL +config SND_HDA_PREALLOC_SIZE + int "Pre-allocated buffer size for HD-audio driver" + range 0 32768 + default 64 + help + Speficies the default pre-allocated buffer-size in kB for + HD-audio driver. A larger buffer (e.g. 2048) is preferred + for systems with PulseAudio. The default 64 is chosen just + from the compatibility reason. + + Note that the pre-allocation size can be changed dynamically + via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. + config SND_HDA_HWDEP bool "Build hwdep interface for HD-audio driver" select SND_HWDEP diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 25619cd..5ce9531 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2075,6 +2075,8 @@ static void azx_pcm_free(struct snd_pcm *pcm) } } +#define MAX_PREALLOC_SIZE (32 * 1024 * 1024) + static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, struct hda_pcm *cpcm) @@ -2083,6 +2085,7 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, struct snd_pcm *pcm; struct azx_pcm *apcm; int pcm_dev = cpcm->device; + unsigned int size; int s, err; if (pcm_dev >= HDA_MAX_PCMS) { @@ -2118,9 +2121,12 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, snd_pcm_set_ops(pcm, s, &azx_pcm_ops); } /* buffer pre-allocation */ + size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024; + if (size > MAX_PREALLOC_SIZE) + size = MAX_PREALLOC_SIZE; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(chip->pci), - 1024 * 64, 32 * 1024 * 1024); + size, MAX_PREALLOC_SIZE); return 0; } -- 1.7.6

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[alsa-devel] [PATCH][INFO]: linux/sound/usb/ : add special settings for M-Audio Fast Track Pro and Quattro (24 bits, etc...)
by Guillaume Pellerin 12 Jul '11

12 Jul '11

!! ALREADY APPLIED BY TAKASHI IWAI ON LINUX 3.1 !! This patch built against Linux 2.6.39 gives M-Audio Fast Track Pro and M-Audio Quattro quirks and endpoints to boot and setup those devices with special options (digital inputs and outputs, 24 bits mode, etc...). M-Audio Audiophile quirks are just adapted to match the new global M-Audio parameters. Special configurations can be then loaded through a modprobe conf file. For example, to set the 24 bits mode on the Fast Track Pro add /etc/modprobe.d/fast_track_pro.conf : options snd_usb_audio vid=0x763 pid=0x2012 device_setup=0x08 index=1 enable=1 Here is a list of the possibilities in this example : http://files.parisson.com/debian/fast-track-pro.conf Enjoy, Cheers Alsa Team ;) Guillaume

1 0

[alsa-devel] [PATCH] ASoC: Use WM8994 FLL lock interrupt
by Mark Brown 12 Jul '11

12 Jul '11

If we have interrupts then wait for the FLL lock interrupt rather than using dead reckoning when waiting for the FLL to start. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 40 +++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm8994.h | 3 +++ 2 files changed, 42 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d2dcaa2..f89ae47 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1635,6 +1635,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, int reg_offset, ret; struct fll_div fll; u16 reg, aif1, aif2; + unsigned long timeout; aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) & WM8994_AIF1CLK_ENA; @@ -1726,7 +1727,15 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, WM8994_FLL1_ENA | WM8994_FLL1_FRAC, reg); - msleep(5); + if (wm8994->fll_locked_irq) { + timeout = wait_for_completion_timeout(&wm8994->fll_locked[id], + msecs_to_jiffies(10)); + if (timeout == 0) + dev_warn(codec->dev, + "Timed out waiting for FLL lock\n"); + } else { + msleep(5); + } } wm8994->fll[id].in = freq_in; @@ -1744,6 +1753,14 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, return 0; } +static irqreturn_t wm8994_fll_locked_irq(int irq, void *data) +{ + struct completion *completion = data; + + complete(completion); + + return IRQ_HANDLED; +} static int opclk_divs[] = { 10, 20, 30, 40, 55, 60, 80, 120, 160 }; @@ -2879,6 +2896,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->pdata = dev_get_platdata(codec->dev->parent); wm8994->codec = codec; + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) + init_completion(&wm8994->fll_locked[i]); + if (wm8994->pdata && wm8994->pdata->micdet_irq) wm8994->micdet_irq = wm8994->pdata->micdet_irq; else if (wm8994->pdata && wm8994->pdata->irq_base) @@ -2994,6 +3014,16 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } } + wm8994->fll_locked_irq = true; + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) { + ret = wm8994_request_irq(codec->control_data, + WM8994_IRQ_FLL1_LOCK + i, + wm8994_fll_locked_irq, "FLL lock", + &wm8994->fll_locked[i]); + if (ret != 0) + wm8994->fll_locked_irq = false; + } + /* Remember if AIFnLRCLK is configured as a GPIO. This should be * configured on init - if a system wants to do this dynamically * at runtime we can deal with that then. @@ -3179,6 +3209,9 @@ err_irq: wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994); if (wm8994->micdet_irq) free_irq(wm8994->micdet_irq, wm8994); + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) + wm8994_free_irq(codec->control_data, WM8994_IRQ_FLL1_LOCK + i, + &wm8994->fll_locked[i]); wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); err: @@ -3190,11 +3223,16 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = codec->control_data; + int i; wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); pm_runtime_disable(codec->dev); + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) + wm8994_free_irq(codec->control_data, WM8994_IRQ_FLL1_LOCK + i, + &wm8994->fll_locked[i]); + wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 0a1db04..1ab2266 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -11,6 +11,7 @@ #include <sound/soc.h> #include <linux/firmware.h> +#include <linux/completion.h> #include "wm_hubs.h" @@ -79,6 +80,8 @@ struct wm8994_priv { int mclk[2]; int aifclk[2]; struct wm8994_fll_config fll[2], fll_suspend[2]; + struct completion fll_locked[2]; + bool fll_locked_irq; int dac_rates[2]; int lrclk_shared[2]; -- 1.7.5.4

1 0

[alsa-devel] [PATCH v4 3/4] ASoC: SAMSUNG: Modify I2S driver to support idma
by Sangbeom Kim 12 Jul '11

12 Jul '11

Previously, I2S driver only can support system dma. In this patch, i2s driver can support internal dma too. IDMA h/w configuration is initialized on idma.c Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/soc/samsung/i2s.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 1568eea..c086b78 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -21,6 +21,7 @@ #include <plat/audio.h> #include "dma.h" +#include "idma.h" #include "i2s.h" #include "i2s-regs.h" @@ -60,6 +61,7 @@ struct i2s_dai { /* DMA parameters */ struct s3c_dma_params dma_playback; struct s3c_dma_params dma_capture; + struct s3c_dma_params idma_playback; u32 quirks; u32 suspend_i2smod; u32 suspend_i2scon; @@ -877,6 +879,10 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) if (i2s->quirks & QUIRK_NEED_RSTCLR) writel(CON_RSTCLR, i2s->addr + I2SCON); + if (i2s->quirks & QUIRK_SEC_DAI) + idma_reg_addr_init((void *)i2s->addr, + i2s->sec_dai->idma_playback.dma_addr); + probe_exit: /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; @@ -1077,6 +1083,7 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) sec_dai->dma_playback.dma_size = 4; sec_dai->base = regs_base; sec_dai->quirks = quirks; + sec_dai->idma_playback.dma_addr = i2s_cfg->idma_addr; sec_dai->pri_dai = pri_dai; pri_dai->sec_dai = sec_dai; } -- 1.7.1

4 7

[alsa-devel] [PATCH 1/2] ASoC: Implement DC servo completion IRQ handling for wm_hubs devices
by Mark Brown 12 Jul '11

12 Jul '11

The individual devices should set the flag dcs_done_irq in the hubs shared data structure to indicate that they will flag the interrupt by calling wm_hubs_dcs_done(). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 34 +++++++++++++++++++++++++++++----- sound/soc/codecs/wm_hubs.h | 8 ++++++++ 2 files changed, 37 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 7e60e22..5c2d565 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -63,9 +63,11 @@ static const struct soc_enum speaker_mode = static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) { + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); unsigned int reg; int count = 0; unsigned int val; + unsigned long timeout; val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1; @@ -74,18 +76,37 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) dev_dbg(codec->dev, "Waiting for DC servo...\n"); - do { - count++; - msleep(1); + if (hubs->dcs_done_irq) { + timeout = wait_for_completion_timeout(&hubs->dcs_done, + msecs_to_jiffies(500)); + if (timeout == 0) + dev_warn(codec->dev, "No DC servo interrupt\n"); + reg = snd_soc_read(codec, WM8993_DC_SERVO_0); - dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & op && count < 400); + } else { + do { + count++; + msleep(1); + reg = snd_soc_read(codec, WM8993_DC_SERVO_0); + dev_dbg(codec->dev, "DC servo: %x\n", reg); + } while (reg & op && count < 400); + } if (reg & op) dev_err(codec->dev, "Timed out waiting for DC Servo %x\n", op); } +irqreturn_t wm_hubs_dcs_done(int irq, void *data) +{ + struct wm_hubs_data *hubs = data; + + complete(&hubs->dcs_done); + + return IRQ_HANDLED; +} +EXPORT_SYMBOL_GPL(wm_hubs_dcs_done); + /* * Startup calibration of the DC servo */ @@ -863,8 +884,11 @@ EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_controls); int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, int lineout1_diff, int lineout2_diff) { + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; + init_completion(&hubs->dcs_done); + snd_soc_dapm_add_routes(dapm, analogue_routes, ARRAY_SIZE(analogue_routes)); diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 0d290d2..676b125 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -14,6 +14,9 @@ #ifndef _WM_HUBS_H #define _WM_HUBS_H +#include <linux/completion.h> +#include <linux/interrupt.h> + struct snd_soc_codec; extern const unsigned int wm_hubs_spkmix_tlv[]; @@ -28,6 +31,9 @@ struct wm_hubs_data { bool class_w; u16 class_w_dcs; + + bool dcs_done_irq; + struct completion dcs_done; }; extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *); @@ -38,4 +44,6 @@ extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, int jd_scthr, int jd_thr, int micbias1_lvl, int micbias2_lvl); +extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); + #endif -- 1.7.5.4

1 1

Re: [alsa-devel] Sound BoF in LinuxCon Europe?
by Liam Girdwood 12 Jul '11

12 Jul '11

On Fri, 2011-07-08 at 12:36 +0200, Takashi Iwai wrote: > Hi, > > just wondering how about holding a sound-BoF in LinuxCon Europe in > October? > > I know ELCE will be held at the very same time, so it may need the > adjustment of schedule. But in general what do you think? > I don't think I'll be at ELCE but Peter will be attending from TI. > I know some guys will attend LPC in September, too. If the topic is > redundant again after one month, it's fine, no much need for that. > OTOH, it's a conference in Europe, so people there may have more > chance to attend (like me :-) > Mark and I are looking at another ASoC/Audio workshop at LPC, to go over progress and planning for the next development phase. Imo, this would likely complement a BOF at LinuxCon since a good proportion of the BOF audience will probably be different from LPC with different hardware, requirements, users, etc. > An alternative is a workshop in KS2011 day 1, but I think KS suits > better for more broader topics, e.g. the discussions on the media- > notification support or the jack implementation like a thread in the > last month. > Regards Liam

3 2

[alsa-devel] What is correct way to put conditional stuff in ASoC codec driver?
by Ashish Chavan 11 Jul '11

11 Jul '11

Hi, I am working on updating ASoC codec driver for Dialog's DA7210 codec (sound/soc/codec/da7210.c). This update would be a major functionality update in nature. In fact we already have a feature complete driver for this codec but there are two main issues with that driver, (1) It was developed(and tested) for a quite old kernel version (2.6.28) (2) It was written in custom(non standard) way and not suitable for direct submission to kernel Our final goal is to pull in all missing features from this custom driver in to existing mainline driver. I have already created few patches for basic functions but I am bit confused at one point. Some of the features supported by this codec have inter dependency, e.g. ALC can be used only if NOISE SUPRESSION is disabled. So ideally, all controls related to ALC should be either disabled or not added at all, if NOISE SUPRESSION to be used. There are few other features having similar kind of dependency. I just want to know what is the correct way to handle this in ASoC codec driver? Looking at the existing codec drivers, it seems that having conditional defines is not common here. As We just want to support static configuration of such features, is it a good idea to add sub menu options in driver's Kconfig to enable/disable such features and use them within code? Any pointer(s) to existing code are most welcome. Thanks, -- Ashish ---------------------------------------------------------------------------- || Linux Is User Friendly, It's Just Selective About Who It's Friends Are || ----------------------------------------------------------------------------

2 11

[alsa-devel] [PATCH] Acer aspire 3830TG and Conexant 506c/20588
by David Henningsson 11 Jul '11

11 Jul '11

The new Conexant 5066 auto-parser is barely finished and already we might need a quirk system for it... Anyway, in this bug, the internal speaker is not working (and was not working before the auto-parser either). I'm attaching two patches that are quite simple, and at least the first one (add ID 506c) I think should be applied right away. The problem: The internal speaker is at node 0x1f and you need to turn on EAPD on node 0x1b for the speaker to sound. Now I'm not sure how to fix this in the best way. An "init verb" quirk, or a "use EAPD from this pin" quirk? Try to copy-paste realtek's quirk system, and if so, refactor it into hda_codec.c, or write something new? What are your ideas? Alsa-info reference: https://launchpadlibrarian.net/73966502/alsa-info Bug reference: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/783582 -- David Henningsson http://launchpad.net/~diwic

2 10

[alsa-devel] [RFC PATCH] ARM ASoC: add sound driver for imx27pdk using mc13783 codec
by Jürgen Lambrecht 11 Jul '11

11 Jul '11

- Started from sound/soc/imx/phycore-mc13783.c - Renamed to general "imx" because also valid for imx27pdk (3ds) and other 3ds platforms. Or should there be a file per platform? - Fixed some issues (add names, add read/write functions, changed late_initcall to module_init, configure clock input pin) - Add debug printing and comments. - Result: * soundcard detection; * after creating /dev entries, aplay runs; * both 'aplay song.wav' and 'madplay song.mp3' give the same error: a time-out waiting for DMA (ALSA pcm_lib.c:1802: playback write error (DMA or IRQ trouble?)) * this driver uses the DMA method; the FIQ method gives the same error Signed-off-by: J�rgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/imx/Kconfig | 20 +++-- sound/soc/imx/Makefile | 2 + sound/soc/imx/imx-mc13783.c | 175 +++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 190 insertions(+), 7 deletions(-) create mode 100644 sound/soc/imx/imx-mc13783.c diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index d8f130d..2f57f12 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -11,20 +11,19 @@ menuconfig SND_IMX_SOC if SND_IMX_SOC -config SND_MXC_SOC_SSI - tristate - config SND_MXC_SOC_FIQ tristate config SND_MXC_SOC_MX2 tristate +config SND_SOC_IMX_MC13783 + tristate + config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL select SND_SOC_WM8350 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Enable support for audio on the i.MX31ADS with the WM1133-EV1 @@ -34,7 +33,6 @@ config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 select SND_SOC_TVL320AIC32X4 - select SND_MXC_SOC_SSI select SND_MXC_SOC_MX2 help Say Y if you want to add support for SoC audio on Visstrim SM10 @@ -44,7 +42,6 @@ config SND_SOC_PHYCORE_AC97 tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" depends on MACH_PCM043 || MACH_PCA100 select SND_SOC_WM9712 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Say Y if you want to add support for SoC audio on Phytec phyCORE @@ -57,10 +54,19 @@ config SND_SOC_EUKREA_TLV320 || MACH_EUKREA_MBIMXSD35_BASEBOARD \ || MACH_EUKREA_MBIMXSD51_BASEBOARD select SND_SOC_TLV320AIC23 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX27_MC13783 + tristate "SoC Audio support for i.MX27 platforms with a MC13783 codec" + depends on MACH_MX27_3DS + select SND_SOC_MC13783 + select SND_MXC_SOC_MX2 + select SND_SOC_IMX_MC13783 + help + Say Y if you want to add support for SoC audio on i.MX27 platforms + with a MC13783 codec (based on Freescale's imx27pdk kit) + endif # SND_IMX_SOC diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index d6d609b..d86e5b8 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -12,8 +12,10 @@ snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o +snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o +obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/imx/imx-mc13783.c b/sound/soc/imx/imx-mc13783.c new file mode 100644 index 0000000..4274f84 --- /dev/null +++ b/sound/soc/imx/imx-mc13783.c @@ -0,0 +1,175 @@ +/* + * imx-mc13783.c -- SoC audio for imx processor with mx13783 co-processor + * + * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer(a)pengutronix.de> + * Copyright 2011 J�rgen Lambrecht, Televic <J.Lambrecht(a)televic.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. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <asm/mach-types.h> +#include "../codecs/mc13783.h" +#include "imx-ssi.h" + +static struct snd_soc_card imx_mc13783_audio; + +#define FMT_PLAYBACK (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) +#define FMT_CAPTURE (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + +static int imx_mc13783_audio_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* set codec and cpu DAI configuration */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = snd_soc_dai_set_fmt(codec_dai, FMT_PLAYBACK); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + ret = snd_soc_dai_set_fmt(cpu_dai, FMT_PLAYBACK); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + } else { + ret = snd_soc_dai_set_fmt(codec_dai, FMT_CAPTURE); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + ret = snd_soc_dai_set_fmt(cpu_dai, FMT_CAPTURE); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0, 0, 4, 32); /* args are not used in mc13783 */ + if (ret) + return ret; + } +//ok: checked; we use clia input pin; last arg is not used + ret = snd_soc_dai_set_sysclk(codec_dai, MC13783_CLK_CLIA, 26000000, SND_SOC_CLOCK_OUT); + if (ret) { + pr_err("%s: failed setting codec sysclk\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x0, 0xfffffffc, 4, 32); + if (ret) + return ret; + +//from eukrea-tlv320.c and mx27vis-aic32x4.c + ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret) { + pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + return ret; + } + + return 0; +} + +static int imx_mc13783_audio_hifi_hw_free(struct snd_pcm_substream *substream) +{ + return 0; +} + +static struct snd_soc_ops imx_mc13783_audio_hifi_ops = { + .hw_params = imx_mc13783_audio_hifi_hw_params, + .hw_free = imx_mc13783_audio_hifi_hw_free, +}; + +static int imx_mc13783_audio_probe(struct platform_device *pdev) +{ + return 0; +} + +static int imx_mc13783_audio_remove(struct platform_device *pdev) +{ + return 0; +} + +static struct snd_soc_dai_link imx_dai_mc13783[] = { + { + .name = "MC13783 Playback", + .stream_name = "Playback", + .codec_dai_name = "mc13783-hifi", + .platform_name = "imx-pcm-audio.0", + .codec_name = "mc13783-codec", + .cpu_dai_name = "imx-ssi.0", + .ops = &imx_mc13783_audio_hifi_ops, + }, { + .name = "MC13783 Capture", + .stream_name = "Capture", + .codec_dai_name = "mc13783-hifi", + .platform_name = "imx-pcm-audio.0", + .codec_name = "mc13783-codec", + .cpu_dai_name = "imx-ssi.0", + .ops = &imx_mc13783_audio_hifi_ops, + }, +}; + +static struct snd_soc_card imx_mc13783_audio = { + .name = "iMX-mc13783-audio", + .probe = imx_mc13783_audio_probe, + .remove = imx_mc13783_audio_remove, + .dai_link = imx_dai_mc13783, + .num_links = ARRAY_SIZE(imx_dai_mc13783), +}; + +static struct platform_device *imx_mc13783_snd_device; + +static int __init imx_mc13783_audio_init(void) +{ + int ret; + + pr_info("IMX MC13783 Sound Card\n"); +/* Possibly add a check for IMX and MC13783 */ +/* if (!machine_is_pcm038() && !machine_is_pcm037()) */ +/* * return happy. We might run on a totally different machine * */ +/* return 0; */ + + imx_mc13783_snd_device = platform_device_alloc("soc-audio", -1); + if (!imx_mc13783_snd_device) + return -ENOMEM; + + platform_set_drvdata(imx_mc13783_snd_device, &imx_mc13783_audio); + ret = platform_device_add(imx_mc13783_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(imx_mc13783_snd_device); + } + + return ret; +} + +static void __exit imx_mc13783_audio_exit(void) +{ + platform_device_unregister(imx_mc13783_snd_device); +} + +//late_initcall(imx_mc13783_audio_init); +module_init(imx_mc13783_audio_init); +module_exit(imx_mc13783_audio_exit); + +MODULE_AUTHOR("Sascha Hauer <s.hauer(a)pengutronix.de>"); +MODULE_DESCRIPTION("iMX ALSA SoC driver"); +MODULE_LICENSE("GPL"); -- 1.7.1

4 4

[PATCH 1/1] sgtl5000: fix i2c r/w issue add neglect VDDIO and VDDI=
by Dong Aisheng 11 Jul '11

11 Jul '11

O This is caused by sgtl5000 using register address step is 2, and snd-soc-core can't handle this as we expect, so we have to fill the register cache by reading register out when initialization instead of providing a default value array. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 159 ++++++++++++++-------------------------= ---- 1 files changed, 52 insertions(+), 107 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index ff29380..c29f153 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -33,87 +33,18 @@ #define SGTL5000_DAP_REG_OFFSET 0x0100 #define SGTL5000_MAX_REG_OFFSET 0x013A =20 -/* default value of sgtl5000 registers except DAP */ -static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] =3D { - 0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */ - 0x0000, /* 0x0002, CHIP_DIG_POWER. */ - 0x0008, /* 0x0004, CHIP_CKL_CTRL */ - 0x0010, /* 0x0006, CHIP_I2S_CTRL */ - 0x0000, /* 0x0008, reserved */ - 0x0008, /* 0x000A, CHIP_SSS_CTRL */ - 0x0000, /* 0x000C, reserved */ - 0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */ - 0x3c3c, /* 0x0010, CHIP_DAC_VOL */ - 0x0000, /* 0x0012, reserved */ - 0x015f, /* 0x0014, CHIP_PAD_STRENGTH */ - 0x0000, /* 0x0016, reserved */ - 0x0000, /* 0x0018, reserved */ - 0x0000, /* 0x001A, reserved */ - 0x0000, /* 0x001E, reserved */ - 0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */ - 0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */ - 0x0111, /* 0x0024, CHIP_ANN_CTRL */ - 0x0000, /* 0x0026, CHIP_LINREG_CTRL */ - 0x0000, /* 0x0028, CHIP_REF_CTRL */ - 0x0000, /* 0x002A, CHIP_MIC_CTRL */ - 0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */ - 0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */ - 0x7060, /* 0x0030, CHIP_ANA_POWER */ - 0x5000, /* 0x0032, CHIP_PLL_CTRL */ - 0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */ - 0x0000, /* 0x0036, CHIP_ANA_STATUS */ - 0x0000, /* 0x0038, reserved */ - 0x0000, /* 0x003A, CHIP_ANA_TEST2 */ - 0x0000, /* 0x003C, CHIP_SHORT_CTRL */ - 0x0000, /* reserved */ -}; - -/* default value of dap registers */ -static const u16 sgtl5000_dap_regs[] =3D { - 0x0000, /* 0x0100, DAP_CONTROL */ - 0x0000, /* 0x0102, DAP_PEQ */ - 0x0040, /* 0x0104, DAP_BASS_ENHANCE */ - 0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */ - 0x0000, /* 0x0108, DAP_AUDIO_EQ */ - 0x0040, /* 0x010A, DAP_SGTL_SURROUND */ - 0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */ - 0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */ - 0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */ - 0x0000, /* 0x0112, reserved */ - 0x0000, /* 0x0114, reserved */ - 0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */ - 0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */ - 0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */ - 0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */ - 0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */ - 0x8000, /* 0x0120, DAP_MAIN_CHAN */ - 0x0000, /* 0x0122, DAP_MIX_CHAN */ - 0x0510, /* 0x0124, DAP_AVC_CTRL */ - 0x1473, /* 0x0126, DAP_AVC_THRESHOLD */ - 0x0028, /* 0x0128, DAP_AVC_ATTACK */ - 0x0050, /* 0x012A, DAP_AVC_DECAY */ - 0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */ - 0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */ - 0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */ - 0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */ - 0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */ - 0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */ - 0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */ - 0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */ -}; - /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { - VDDA, - VDDIO, +// VDDA, +// VDDIO, VDDD, SGTL5000_SUPPLY_NUM }; =20 /* vddd is optional supply */ static const char *supply_names[SGTL5000_SUPPLY_NUM] =3D { - "VDDA", - "VDDIO", +// "VDDA", +// "VDDIO", "VDDD" }; =20 @@ -780,6 +711,11 @@ static int ldo_regulator_is_enabled(struct regulator_d= ev *dev) return ldo->enabled; } =20 +/* + * enable internal VDDD power supply. Since register + * cache not fill yet, we have to use hw_read and write + * instead of snd_soc_read and snd_soc_write. + */ static int ldo_regulator_enable(struct regulator_dev *dev) { struct ldo_regulator *ldo =3D rdev_get_drvdata(dev); @@ -797,17 +733,17 @@ static int ldo_regulator_enable(struct regulator_dev = *dev) ldo->voltage =3D (1600 - reg * 50) * 1000; =20 /* set voltage to register */ - snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - (0x1 << 4) - 1, reg); + codec->write(codec, SGTL5000_CHIP_LINREG_CTRL, reg); =20 - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_LINEREG_D_POWERUP, - SGTL5000_LINEREG_D_POWERUP); + reg =3D codec->hw_read(codec, SGTL5000_CHIP_ANA_POWER); + reg |=3D SGTL5000_LINEREG_D_POWERUP; + codec->write(codec, SGTL5000_CHIP_ANA_POWER, reg); =20 + reg &=3D ~SGTL5000_LINREG_SIMPLE_POWERUP; /* when internal ldo enabled, simple digital power can be disabled */ - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_LINREG_SIMPLE_POWERUP, - 0); + codec->write(codec, SGTL5000_CHIP_ANA_POWER, reg); + + udelay(10); =20 ldo->enabled =3D 1; return 0; @@ -1022,13 +958,11 @@ static int sgtl5000_suspend(struct snd_soc_codec *co= dec, pm_message_t state) static int sgtl5000_restore_regs(struct snd_soc_codec *codec) { u16 *cache =3D codec->reg_cache; - int i; - int regular_regs =3D SGTL5000_CHIP_SHORT_CTRL >> 1; + int reg; + int step =3D codec->driver->reg_cache_step; =20 /* restore regular registers */ - for (i =3D 0; i < regular_regs; i++) { - int reg =3D i << 1; - + for (reg =3D 0; reg < SGTL5000_CHIP_SHORT_CTRL; reg +=3D step) { /* this regs depends on the others */ if (reg =3D=3D SGTL5000_CHIP_ANA_POWER || reg =3D=3D SGTL5000_CHIP_CLK_CTRL || @@ -1037,35 +971,32 @@ static int sgtl5000_restore_regs(struct snd_soc_code= c *codec) reg =3D=3D SGTL5000_CHIP_CLK_CTRL) continue; =20 - snd_soc_write(codec, reg, cache[i]); + snd_soc_write(codec, reg, cache[reg]); } =20 /* restore dap registers */ - for (i =3D SGTL5000_DAP_REG_OFFSET >> 1; - i < SGTL5000_MAX_REG_OFFSET >> 1; i++) { - int reg =3D i << 1; - - snd_soc_write(codec, reg, cache[i]); - } + for (reg =3D SGTL5000_DAP_REG_OFFSET; + reg < SGTL5000_MAX_REG_OFFSET; reg +=3D step) + snd_soc_write(codec, reg, cache[reg]); =20 /* * restore power and other regs according * to set_power() and set_clock() */ snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, - cache[SGTL5000_CHIP_LINREG_CTRL >> 1]); + cache[SGTL5000_CHIP_LINREG_CTRL]); =20 snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, - cache[SGTL5000_CHIP_ANA_POWER >> 1]); + cache[SGTL5000_CHIP_ANA_POWER]); =20 snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, - cache[SGTL5000_CHIP_CLK_CTRL >> 1]); + cache[SGTL5000_CHIP_CLK_CTRL]); =20 snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, - cache[SGTL5000_CHIP_REF_CTRL >> 1]); + cache[SGTL5000_CHIP_REF_CTRL]); =20 snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]); + cache[SGTL5000_CHIP_LINE_OUT_CTRL]); return 0; } =20 @@ -1105,12 +1036,12 @@ static int sgtl5000_set_power_regs(struct snd_soc_c= odec *codec) int vag; struct sgtl5000_priv *sgtl5000 =3D snd_soc_codec_get_drvdata(codec); =20 - vdda =3D regulator_get_voltage(sgtl5000->supplies[VDDA].consumer); - vddio =3D regulator_get_voltage(sgtl5000->supplies[VDDIO].consumer); +// vdda =3D regulator_get_voltage(sgtl5000->supplies[VDDA].consumer); +// vddio =3D regulator_get_voltage(sgtl5000->supplies[VDDIO].consumer); vddd =3D regulator_get_voltage(sgtl5000->supplies[VDDD].consumer); =20 - vdda =3D vdda / 1000; - vddio =3D vddio / 1000; + vdda =3D 3300; + vddio =3D 3300; vddd =3D vddd / 1000; =20 if (vdda <=3D 0 || vddio <=3D 0 || vddd < 0) { @@ -1335,6 +1266,22 @@ err_regulator_free: =20 } =20 +static int sgtl5000_fill_reg_cache(struct snd_soc_codec *codec) +{ + int reg; + int step =3D codec->driver->reg_cache_step; + u16 *cache =3D codec->reg_cache; + + for (reg =3D SGTL5000_DAP_REG_OFFSET; + reg <=3D SGTL5000_MAX_REG_OFFSET; reg +=3D step) + cache[reg] =3D codec->hw_read(codec, reg); + + for (reg =3D 0; reg <=3D SGTL5000_CHIP_SHORT_CTRL; reg +=3D step) + cache[reg] =3D codec->hw_read(codec, reg); + + return 0; +} + static int sgtl5000_probe(struct snd_soc_codec *codec) { int ret; @@ -1351,6 +1298,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec= ) if (ret) return ret; =20 + sgtl5000_fill_reg_cache(codec); + /* power up sgtl5000 */ ret =3D sgtl5000_set_power_regs(codec); if (ret) @@ -1443,10 +1392,9 @@ static struct snd_soc_codec_driver sgtl5000_driver = =3D { .suspend =3D sgtl5000_suspend, .resume =3D sgtl5000_resume, .set_bias_level =3D sgtl5000_set_bias_level, - .reg_cache_size =3D ARRAY_SIZE(sgtl5000_regs), + .reg_cache_size =3D SGTL5000_MAX_REG_OFFSET, .reg_word_size =3D sizeof(u16), .reg_cache_step =3D 2, - .reg_cache_default =3D sgtl5000_regs, .volatile_register =3D sgtl5000_volatile_register, }; =20 @@ -1466,9 +1414,6 @@ static __devinit int sgtl5000_i2c_probe(struct i2c_cl= ient *client, * at init phase makes life easy. * FIXME: should we drop 'const' of sgtl5000_regs? */ - memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)), - sgtl5000_dap_regs, - SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET); =20 i2c_set_clientdata(client, sgtl5000); =20 --=20 1.7.0.4

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[alsa-devel] [PATCH] ASoC: Use late enable handling for direct voice, speaker and headphone
by Mark Brown 11 Jul '11

11 Jul '11

This ensures appropriate clocking for bypass paths to speaker and headphone and direct voice paths on affected revisions. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 30 +++++++++++++++++++++--------- sound/soc/codecs/wm_hubs.c | 3 --- 3 files changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 03af7ad..6e85b88 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -876,7 +876,7 @@ SND_SOC_DAPM_MIXER("SPKL", WM8993_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), SND_SOC_DAPM_MIXER("SPKR", WM8993_POWER_MANAGEMENT_3, 9, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), - +SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), }; static const struct snd_soc_dapm_route routes[] = { diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 8d1258c..96ea69b 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1146,13 +1146,33 @@ SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, late_enable_ev, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), + +SND_SOC_DAPM_MIXER_E("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, + left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer), + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MIXER_E("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, + right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer), + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) }; static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0) +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, + left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), +SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, + right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), }; static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = { @@ -1283,14 +1303,6 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), -SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), - -SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, - left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), -SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, - right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), - SND_SOC_DAPM_POST("Debug log", post_ev), }; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2d6c88b..7e60e22 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -598,9 +598,6 @@ SND_SOC_DAPM_MIXER("IN2L PGA", WM8993_POWER_MANAGEMENT_2, 7, 0, SND_SOC_DAPM_MIXER("IN2R PGA", WM8993_POWER_MANAGEMENT_2, 5, 0, in2r_pga, ARRAY_SIZE(in2r_pga)), -/* Dummy widgets to represent differential paths */ -SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("MIXINL", WM8993_POWER_MANAGEMENT_2, 9, 0, mixinl, ARRAY_SIZE(mixinl)), SND_SOC_DAPM_MIXER("MIXINR", WM8993_POWER_MANAGEMENT_2, 8, 0, -- 1.7.5.4

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[alsa-devel] [PATCH 0/5] treewide: Add missing breaks in switch/case
by Joe Perches 10 Jul '11

10 Jul '11

Found by grep. Uncompiled/untested. Joe Perches (5): gpio-tps65910.c: Add missing breaks in switch/case tps65910-irq: Add missing breaks in switch/case ath5k: Add missing breaks in switch/case fusb300_udc: Add missing breaks in switch/case dbri: Add missing breaks in switch/case drivers/gpio/gpio-tps65910.c | 2 ++ drivers/mfd/tps65910-irq.c | 2 ++ drivers/net/wireless/ath/ath5k/desc.c | 3 +++ drivers/usb/gadget/fusb300_udc.c | 3 +++ sound/sparc/dbri.c | 5 +++++ 5 files changed, 15 insertions(+), 0 deletions(-) -- 1.7.6.131.g99019

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[alsa-devel] max channels of 3 jacks 6 channels motherboard using model=auto realtek codec
by Raymond Yau 10 Jul '11

10 Jul '11

Refer to commit ce764ab22e40a046065c9417ee5f04ad2a816ac1 ALSA: hda - Add channel-mode support to Realtek auto-parser It seem that max channel is six instead of two when the channel mode is 2ch for those 3 jacks motherboard using hda-emu and alsa-info from http://thread.gmane.org/gmane.linux.alsa.devel/86816 > PCM 0 p 44100 2 16 Open PCM ALC887 Analog for play Available PCM parameters: channels: 2/6 formats: S16_LE S32_LE rates: 44100 48000 96000 192000 Prepare PCM, rate=44100, channels=2, format=16 bits because spec->multiout.max_channels = spec->multiout.num_dacs * 2; is assigned after alc_auto_add_multi_channel_mode() increase spec->multiout.num_dacs from 1 to 3 spec->multiout.num_dacs = num_pins + 1;

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[alsa-devel] [PATCH] ASoC: OMAP4: HDMI: Make stricter dependency on OMAP2_DSS
by Ricardo Neri 09 Jul '11

09 Jul '11

When building DSS HDMI driver with audio support, there is dependency on SND_SOC to register the ASoC audio codec. Also, SND_OMAP_SOC_OMAP4_HDMI has dependency on OMAP2_DSS. Then, to avoid build breakages, SND_OMAP_SOC_OMAP4_HDMI, is restricted to be available only when OMAP2_DSS AND SND_SOC have the same build type. Also, dependency on ARCH_OMAP4 is removed, as it is checked indirectly by OMAP4_DSS_HDMI. Reported-by: Liam Girdwood <lrg(a)ti.com> Signed-off-by: Ricardo Neri <ricardo.neri(a)ti.com> --- sound/soc/omap/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index e8ca5be..ffbd83f 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -113,7 +113,7 @@ config SND_OMAP_SOC_SDP4430 config SND_OMAP_SOC_OMAP4_HDMI tristate "SoC Audio support for Texas Instruments OMAP4 HDMI" - depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS && ARCH_OMAP4 + depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && (OMAP2_DSS = SND_SOC) select SND_OMAP_SOC_HDMI help Say Y if you want to add support for SoC HDMI audio on Texas Instruments -- 1.7.4.1

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[alsa-devel] ALSA become "Segmentation fault" on current linus tree
by kuninori.morimoto.gx＠renesas.com 09 Jul '11

09 Jul '11

Dear Liam, Mark Currect linus tree become "Segmentation fault" when I use aplay. # aplay xxx.wav Segmentation fault I used "git bisect" and 1st bad commit was below. I'm using sh7372 mackerel board. commit 22de71ba03311cdc1063757c50a1488cb90a1fca Author: Liam Girdwood <lrg(a)ti.com> Date: Thu May 12 16:14:04 2011 +0100 ASoC: core - allow ASoC more flexible machine name Allow ASoC machine drivers to register a driver name and a longname. This allows user space to determine the flavour of machine driver. Signed-off-by: Liam Girdwood <lrg(a)ti.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Best regards --- Kuninori Morimoto

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