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

1 0

[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

1 0

[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

1 2

[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;

2 2

[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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[alsa-devel] ASoC updates for 3.1
by Mark Brown 09 Jul '11

09 Jul '11

The following changes since commit f187700c2d769bd3160b52f99cc4e747d5c5014b: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-07-05 08:20:19 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Axel Lin (1): ASoC: pxa2xx-pcm: remove unused variable 'dai' Kuninori Morimoto (1): ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2 Liam Girdwood (4): ASoC: core - Add API call to register platform kcontrols. ASoC: core - Add platform widget IO ASoC: core - Add convenience register for platform kcontrol and DAPM ASoC: core - Add platform IO tracing Mark Brown (3): ASoC: Manage WM8731 ACTIVE bit as a supply widget ASoC: Manage WM8731 ACTIVE bit as a supply widget Merge branch 'for-3.0' into for-3.1 Stephen Warren (1): ASoC: Tegra: Implement SPDIF CPU DAI include/sound/soc-dapm.h | 2 + include/sound/soc.h | 12 + include/trace/events/asoc.h | 45 ++++ sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/wm8731.c | 29 +--- sound/soc/pxa/pxa2xx-pcm.c | 1 - sound/soc/soc-core.c | 47 ++++ sound/soc/soc-dapm.c | 13 +- sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_spdif.c | 371 ++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_spdif.h | 473 +++++++++++++++++++++++++++++++++++++++++ 12 files changed, 976 insertions(+), 30 deletions(-) create mode 100644 sound/soc/tegra/tegra_spdif.c create mode 100644 sound/soc/tegra/tegra_spdif.h

1 0

[alsa-devel] Jack event API - decision needed
by David Henningsson 09 Jul '11

09 Jul '11

Hi, I'm still new in the community in that sense that I'm not sure how decisions are made. But I could use the outcome of such a decision. Background: I'm trying to pull together the missing pieces of jack detection on both kernel / plumbing / application layers, so that when you plug something in (headset, microphone etc), userspace is notified and can take appropriate actions (e g routing decisions). As part of that I wrote a udev patch a few days ago, which nobody commented on in alsa-devel [1], but was somewhat disliked by at least Kay Sievers who maintains udev [2], who preferred we would rewrite our input layer to do something else within ALSA. So before I proceed further I'd like to know if 1) We're continuing the path with /dev/input devices 2a) We'll rewrite these devices to be read-only ALSA mixer controls 2b) We'll rewrite these devices to be something within ALSA, but not exactly mixer controls. For options 2a) and 2b) I guess the existing /dev/input thing should be deprecated and/or removed. So part of decision should maybe be based on information about how widespread the usage of these devices are currently...? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-June/040916.html [2] e g http://www.spinics.net/lists/hotplug/msg04949.html

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Re: [alsa-devel] Sound BoF in LinuxCon Europe?
by Mark Brown 09 Jul '11

09 Jul '11

On Fri, Jul 08, 2011 at 12:36:34PM +0200, Takashi Iwai wrote: > 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 have no current plans to attend these conferences, I'm planning to go to LPC instead. Not sure if I'd be able to arrange it unless I get a KS invite. > 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. Yes, I don't think it's good for covering at KS as it's far too detailed. If we're going to do something probably LinuxCon/ELC Europe make most sense.

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[alsa-devel] ASoC updates for 3.0
by Mark Brown 09 Jul '11

09 Jul '11

The following changes since commit 873bd4cb4fbba6a3e99f750e17ef2ba6ef96e9d3: ASoC: Don't set invalid name string to snd_card->driver field (2011-07-05 14:39:27 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 The WM8731 patch fixes resume when we suspend with an active playback or record. Kuninori Morimoto (1): ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2 Mark Brown (1): ASoC: Manage WM8731 ACTIVE bit as a supply widget sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/wm8731.c | 29 +++-------------------------- 2 files changed, 4 insertions(+), 27 deletions(-)

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Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Takashi Iwai 09 Jul '11

09 Jul '11

At Thu, 7 Jul 2011 21:35:17 +0900, Toshiharu Okada wrote: > > Date: Thu, 07 Jul 2011 11:00:47 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > >> > >> Therefore, I am not considering about ASoC. > >> Do I need to fit this driver to an ASoC framework for an upstream? > > > >Yes, I guess this would be a better choice in your case. > > Thank you for your comment. > >From now on, We study about ASoC. > If there are a document, a website, a driver, etc. which are referred, > please tell us. Check Documentation/sound/alsa/soc/* files. > Why is choosing ASoS better? > Is it better to correspond to ASoC, when Audio Codec is unfixed? It's not clear what you mean exactly as "unfixed", but in general, the decision rather depends on the usage of the device. If it's designed for a use as a desktop PC component (such as a PCI card or onboard-audio on PC), it'd make sense to implement as a stand-alone, self-contained driver. OTOH, if it's targeted mainly for embedded area, ASoC is the right answer. It's more modularized, and can be more flexibly configured in the end. Takashi

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Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 08 Jul '11

08 Jul '11

On Thu, Jul 7, 2011 at 3:08 PM, Oliver Neukum <oliver(a)neukum.org> wrote: > Am Donnerstag, 7. Juli 2011, 14:38:51 schrieb Daniel Mack: >> On Thu, Jul 7, 2011 at 2:33 PM, Oliver Neukum <oliver(a)neukum.org> wrote: >> >> > PS: Do you still see this if you enable 64bit DMA for EHCI? >> >> The problem is that I personally don't see that issue at all. I even >> installed 4GB of RAM to my development machine last year to be able to >> reproduce this, but I can't, even when the memory allocator is under >> heavy load. The only people who see this effect are Pedro Ribeiro and >> William Light (both in Cc:), and both have been very helpful in trying >> patches and reporting back in detail. Which instructions could we >> probably give to these people to finally hunt this issue down? > > Do they have an IOMMU or are they using bounce buffers? > If the latter, you might be able to reproduce it if you disable your > IOMMU. Ok, Pedro and William - can I ask you to dump the following information to some paste bin: - your .config - the dmesg output of a booted kernel that shows the effect - the output of "sudo lspci -v" Also, the attached patch might give you at least one warning to your kernel logs once the audio stream starts and outputs noise. I'm curious whether it really does. Thanks, Daniel

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[alsa-devel] [RFC PATCH] ARM: imx27pdk (3ds): enable m13783 codec over SSI
by Jürgen Lambrecht 08 Jul '11

08 Jul '11

- add SSI in Kconfig - enable audio in MFD mc13783 - configure audmux-v1 - configure SSI4 Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- arch/arm/mach-imx/Kconfig | 1 + arch/arm/mach-imx/mach-mx27_3ds.c | 40 ++++++++++++++++++++++++++++++++++++- 2 files changed, 40 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-imx/Kconfig b/arch/arm/mach-imx/Kconfig index 0a3512b..2fe5e70 100644 --- a/arch/arm/mach-imx/Kconfig +++ b/arch/arm/mach-imx/Kconfig @@ -286,6 +286,7 @@ config MACH_MX27_3DS select IMX_HAVE_PLATFORM_MXC_MMC select IMX_HAVE_PLATFORM_MXC_NAND select IMX_HAVE_PLATFORM_SPI_IMX + select IMX_HAVE_PLATFORM_IMX_SSI select MXC_DEBUG_BOARD select MXC_ULPI if USB_ULPI help diff --git a/arch/arm/mach-imx/mach-mx27_3ds.c b/arch/arm/mach-imx/mach-mx27_3ds.c index 2787cf3..c0f7dcc 100644 --- a/arch/arm/mach-imx/mach-mx27_3ds.c +++ b/arch/arm/mach-imx/mach-mx27_3ds.c @@ -41,6 +41,7 @@ #include <mach/irqs.h> #include <mach/3ds_debugboard.h> #include <mach/mxc_nand.h> +#include <mach/audmux.h> #include "devices-imx27.h" @@ -101,6 +102,11 @@ static const int mx27pdk_pins[] __initconst = { PE2_PF_USBOTG_DIR, PE24_PF_USBOTG_CLK, PE25_PF_USBOTG_DATA7, + /* SSI4 for audio to MC13783 */ + PC16_PF_SSI4_FS, + PC17_PF_SSI4_RXD, + PC18_PF_SSI4_TXD, + PC19_PF_SSI4_CLK, /* CSPI1 */ PD31_PF_CSPI1_MOSI, PD30_PF_CSPI1_MISO, @@ -271,6 +277,13 @@ static struct regulator_init_data vgen_init = { .consumer_supplies = vgen_consumers, }; +static struct regulator_init_data vaudio_init = { + .constraints = { + .boot_on = 1, + .always_on = 1, + }, +}; + static struct mc13xxx_regulator_init_data mx27_3ds_regulators[] = { { .id = MC13783_REG_VMMC1, @@ -284,6 +297,9 @@ static struct mc13xxx_regulator_init_data mx27_3ds_regulators[] = { }, { .id = MC13783_REG_GPO3, /* Turn on 3.3V */ .init_data = &gpo_init, + }, { + .id = MC13783_REG_VAUDIO, /* Power codec */ + .init_data = &vaudio_init, }, }; @@ -295,7 +311,7 @@ static struct mc13xxx_platform_data mc13783_pdata = { }, .flags = MC13783_USE_REGULATOR | MC13783_USE_TOUCHSCREEN | - MC13783_USE_RTC, + MC13783_USE_RTC | MC13783_USE_CODEC, }; /* SPI */ @@ -385,6 +401,11 @@ static const struct imxi2c_platform_data mx27_3ds_i2c0_data __initconst = { .bitrate = 100000, }; +/* 4-wire is synchronous; SSI-slave and MC13783-master; DMA instead of FIQ */ +static const struct imx_ssi_platform_data mx27pdk_ssi4_pdata __initconst = { + .flags = IMX_SSI_SYN | IMX_SSI_DMA | IMX_SSI_USE_I2S_SLAVE, +}; + static void __init mx27pdk_init(void) { imx27_soc_init(); @@ -420,6 +441,23 @@ static void __init mx27pdk_init(void) pr_warn("Init of the debugboard failed, all devices on the debugboard are unusable.\n"); imx27_add_imx_i2c(0, &mx27_3ds_i2c0_data); imx27_add_imx_fb(&mx27_3ds_fb_data); + + /* SSI unit master I2S codec connected to SSI_PINS_4*/ + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + MXC_AUDMUX_V1_PCR_SYN | /* 4wire mode */ + MXC_AUDMUX_V1_PCR_TFSDIR | /* Transmit Frame Sync is output */ + MXC_AUDMUX_V1_PCR_TCLKDIR | /* Transmit Clock is output */ + MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | /*Transmit Frame Sync and Clock select source port to be HPCR3 or external SSI4*/ + MXC_AUDMUX_V1_PCR_RFSDIR | /* Receive Frame Sync is output */ + MXC_AUDMUX_V1_PCR_RCLKDIR | /* Receive Clock is output */ + MXC_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | /*Receive Frame Sync and Clock select source port to be HPCR3 or external SSI4*/ + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) /*Receive Data select source port to be HPCR3 or external SSI4*/ + ); + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4, + MXC_AUDMUX_V1_PCR_SYN | /* 4wire mode */ + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) /*Receive Data select source port to be HPCR1 or internal SSI1*/ + ); + imx27_add_imx_ssi(0, &mx27pdk_ssi4_pdata); } static void __init mx27pdk_timer_init(void) -- 1.7.1

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Re: [alsa-devel] [PATCH] ASoC: dapm - Add API calls to query valid DAPM paths.
by Mark Brown 07 Jul '11

07 Jul '11

Funny formatting due to phone, sorry... Indeed, but I'd have expected that anything that needed to do that would be able to (and want to) ask by name rather than type? Liam Girdwood <lrg(a)ti.com> wrote: >On 07/07/11 17:38, Mark Brown wrote: >> On Thu, Jul 07, 2011 at 05:18:47PM +0100, Liam Girdwood wrote: >> >>> The reason for using a separate walk here was because this code is >>> taken from my old auto-router and it was to avoid the extra logic >>> introduced by the auto-router. The auto-router did need to check for >>> loops and always select the shortest path when > 1 path was viable >>> (interesting as WM9713 has both loops and multiple paths). Let me see >>> how cleanly I can merge it into the current walk. >> >> Alternatively could we replace the current walk? It's O(n^2) or so in >> the number of widgets so it's not the greatest treasure ever. So long >> as we only have one walk and it works I'm happy. >> >>>> It does also occur to me that perhaps what we want to do here is allow >>>> widgets to find out about paths when they're being activated anyway? >> >>> For dynamic PCM, we need to also work out which DAIs are active based >>> on the graph. Hence for playback we can supply the DAC/AIF and find >>> out all the connected output AIF/Speakers/Pins/etc. This is required >>> at the start of open() in order that we can call the other PCM ops for >>> each DAI, codec and platform. >> >> Right, that's not really what I'm saying though - I'm saying that >> widgets in general could be interested in this information. For >> example, some charge pumps can save additional power with some input >> signal paths (which is currently handled but this would make that code >> more general). > >Ok, but I'm not stopping this sort of functionality from being added >later. Right now this is not a priority (for me anyway). > >> >>>>> + * snd_soc_dapm_get_connected_widgets_type - query audio path and it's widgets. >>>>> + * @dapm: the dapm context. >>>>> + * @stream_name: stream name. >>>>> + * @list: list of active widgets for this stream. >>>>> + * @stream: stream direction. >>>>> + * @type: Initial widget type. >>>>> + * >>>>> + * Queries DAPM graph as to whether an valid audio stream path exists for >>>>> + * the DAPM stream and initial widget type specified. This takes into account >>>>> + * current mixer and mux kcontrol settings. Creates list of valid widgets. >>>> >>>> Why would someone want to query by type? That seems surprising. Name >>>> and/or direction yes but type seems hard to find a use for. >> >>> We query by type so that we can qualify the root widget for a shared >>> common stream name. i.e. Dynamic DSP needs to find the AIF rather than >>> a DAC for stream X. >> >> It does? Hrm. Perhaps it would make life simpler if we just mandated >> the use of explicit AIF widgets? > >But we may also want to know what was after a DAC. I'm thinking back >from the conference discussions where we could use this information to >selectively enable/disable kcontrols. However, this is not going to be >used at the moment either so I'll stick by the AIF widgets. > >Liam > >

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[alsa-devel] [PATCH] ASoC: dapm - Add API calls to query valid DAPM paths.
by Liam Girdwood 07 Jul '11

07 Jul '11

In preparation for ASoC Dynamic PCM (AKA DSP) support. Add DAPM API calls to determine whether a DAPM audio path is valid between source and sink widgets and return a lits of active widgets in that path. This also takes into account all kcontrol mux and mixer settings in between the source and sink widgets to validate the audio path. This will be used by the DSP core to determine the runtime DAI mappings between FE and BE DAIs in order to run PCM operations. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 14 ++ sound/soc/soc-dapm.c | 371 ++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 376 insertions(+), 9 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e09505c..5a7dae6 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -310,6 +310,7 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct snd_soc_dapm_widget_list; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -375,6 +376,13 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); +/* dapm path query */ +int snd_soc_dapm_get_connected_widgets_type(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream, enum snd_soc_dapm_type type); +int snd_soc_dapm_get_connected_widgets_name(struct snd_soc_dapm_context *dapm, + const char *name, struct snd_soc_dapm_widget_list **list, int stream); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ @@ -432,6 +440,8 @@ struct snd_soc_dapm_path { u32 connect:1; /* source and sink widgets are connected */ u32 walked:1; /* path has been walked */ u32 weak:1; /* path ignored for power management */ + u32 length:6; /* path length - used by route checker */ + int (*connected)(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -456,6 +466,8 @@ struct snd_soc_dapm_widget { unsigned char shift; /* bits to shift */ unsigned int saved_value; /* widget saved value */ unsigned int value; /* widget current value */ + unsigned int path_idx; + unsigned int hops; unsigned int mask; /* non-shifted mask */ unsigned int on_val; /* on state value */ unsigned int off_val; /* off state value */ @@ -518,6 +530,8 @@ struct snd_soc_dapm_context { enum snd_soc_bias_level target_bias_level; struct list_head list; + int num_valid_paths; /* valid paths for route checker */ + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0a78482..800af10 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -48,6 +48,8 @@ #include <trace/events/asoc.h> +#define DAPM_MAX_HOPS 16 + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -169,6 +171,366 @@ static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, return change; } +/* get snd_card from DAPM context */ +static inline struct snd_card *dapm_get_snd_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card->snd_card; + else if (dapm->platform) + return dapm->platform->card->snd_card; + + dev_err(dapm->dev, "no snd_card for dapm context\n"); + return NULL; +} + +/* get soc_card from DAPM context */ +static inline struct snd_soc_card *dapm_get_soc_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card; + else if (dapm->platform) + return dapm->platform->card; + + dev_err(dapm->dev, "no soc_card for dapm context\n"); + return NULL; +} + +/* reset 'walked' bit for each dapm path */ +static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm) +{ + struct snd_soc_dapm_path *p; + + list_for_each_entry(p, &dapm->card->paths, list) + p->walked = 0; +} + +/* clear all 'walked' path data for each widget */ +static void dapm_clear_paths(struct snd_soc_dapm_context *dapm) +{ + struct snd_soc_dapm_path *p; + struct snd_soc_dapm_widget *w; + struct list_head *l; + + list_for_each(l, &dapm->card->paths) { + p = list_entry(l, struct snd_soc_dapm_path, list); + p->length = 0; + } + list_for_each(l, &dapm->card->widgets) { + w = list_entry(l, struct snd_soc_dapm_widget, list); + w->hops = 0; + } + dapm_clear_walk(dapm); +} + +/* add widget to list if it's not already in the list */ +static int dapm_add_unique_widget(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget_list **list, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget_list *wlist; + int wlistsize, wlistentries, i; + + /* is the list empty ? */ + if (*list == NULL) { + + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + sizeof(struct snd_soc_dapm_widget *); + *list = kzalloc(wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + } else { + wlist = *list; + + /* is this widget already in the list */ + for (i = 0; i < wlist->num_widgets; i++) { + if (wlist->widgets[i] == w) + return 0; + } + + wlistentries = wlist->num_widgets + 1; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + *list = krealloc(wlist, wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + } + wlist = *list; + + /* insert the widget */ + dev_dbg(dapm->dev, "added %s in widget list pos %d\n", + w->name, wlist->num_widgets); + wlist->widgets[wlist->num_widgets] = w; + wlist->num_widgets++; + return 1; +} + +/* is widget an output endpoint */ +static int is_output_widget_ep(struct snd_soc_dapm_widget *widget) +{ + switch (widget->id) { + case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: + return 1; + case snd_soc_dapm_output: + if (widget->connected && !widget->ext) + return 1; + else + return 0; + case snd_soc_dapm_hp: + case snd_soc_dapm_spk: + case snd_soc_dapm_line: + return !list_empty(&widget->sources); + default: + return 0; + } +} + +/* is widget an input endpoint */ +static int is_input_widget_ep(struct snd_soc_dapm_widget *widget) +{ + switch (widget->id) { + case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: + return 1; + case snd_soc_dapm_input: + if (widget->connected && !widget->ext) + return 1; + else + return 0; + case snd_soc_dapm_mic: + return !list_empty(&widget->sources); + default: + return 0; + } +} + +/* + * Find all the playback paths between the root source widget and + * sink widgets. Store all path widgets in the list. + */ +static int dapm_find_playback_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list, int hops) +{ + struct list_head *lp; + struct snd_soc_dapm_path *path; + int dist = 0; + + if (hops > DAPM_MAX_HOPS) + return 0; + + /* are we at an endpoint widget */ + if (is_output_widget_ep(root)) { + dev_dbg(dapm->dev," ! %d: valid playback route found\n", hops); + dapm->num_valid_paths++; + return 1; + } + + /* have we visited this widget before */ + if (root->hops && root->hops <= hops) + return 0; + root->hops = hops; + + /* check all the output paths on this source widget by walking + * from source to sink */ + list_for_each(lp, &root->sinks) { + path = list_entry(lp, struct snd_soc_dapm_path, list_source); + + dev_dbg(dapm->dev," %c %d: %s -> %s -> %s\n", + path->connect ? '*' : ' ', hops, + root->name, path->name, path->sink->name); + + /* have we been down this path before ? */ + if (path->length && path->length <= hops) + continue; + + /* check down the next path if connected */ + if (path->sink && path->connect && + dapm_find_playback_paths(dapm, path->sink, list, hops + 1)) { + path->length = hops; + + /* add widget to list */ + dapm_add_unique_widget(dapm, list, path->sink); + + if (!dist || dist > path->length) + dist = path->length; + } + } + + return dist; +} + +/* + * Find all the capture paths between the root sink widget and + * source widgets. Store all path widgets in the list. + */ +static int dapm_find_capture_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list, int hops) +{ + struct list_head *lp; + struct snd_soc_dapm_path *path; + int dist = 0; + + if (hops > DAPM_MAX_HOPS) + return 0; + + /* are we at an endpoint widget */ + if (is_input_widget_ep(root)) { + dev_dbg(dapm->dev," ! %d: valid capture route found\n", hops); + dapm->num_valid_paths++; + return 1; + } + + /* have we visited this widget before */ + if (root->hops && root->hops <= hops) + return 0; + root->hops = hops; + + /* check all the output paths on this source widget by walking from + * sink to source */ + list_for_each(lp, &root->sources) { + path = list_entry(lp, struct snd_soc_dapm_path, list_sink); + + dev_dbg(dapm->dev," %c %d: %s <- %s <- %s\n", + path->connect ? '*' : ' ', hops, + root->name, path->name, path->source->name); + + /* have we been here before ? */ + if (path->length && path->length <= hops) + continue; + + /* check down the next path if connected */ + if (path->source && path->connect && + dapm_find_capture_paths(dapm, path->source, list, hops + 1)) { + path->length = hops; + + /* add widget to list */ + dapm_add_unique_widget(dapm, list, path->source); + + if (!dist || dist > path->length) + dist = path->length; + } + } + + return dist; +} + +static int dapm_get_playback_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + dev_dbg(dapm->dev, "Playback: checking paths from %s\n",root->name); + dapm_find_playback_paths(dapm, root, list, 0); + return dapm->num_valid_paths; +} + +static int dapm_get_capture_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + dev_dbg(dapm->dev, "Capture: checking paths to %s\n", root->name); + dapm_find_capture_paths(dapm, root, list, 0); + return dapm->num_valid_paths; +} + +/** + * snd_soc_dapm_get_connected_widgets_type - query audio path and it's widgets. + * @dapm: the dapm context. + * @stream_name: stream name. + * @list: list of active widgets for this stream. + * @stream: stream direction. + * @type: Initial widget type. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the DAPM stream and initial widget type specified. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_get_connected_widgets_type( + struct snd_soc_dapm_context *dapm, const char *stream_name, + struct snd_soc_dapm_widget_list **list, int stream, + enum snd_soc_dapm_type type) +{ + struct snd_soc_dapm_widget *w; + int paths; + + /* get stream root widget AIF, DAC or ADC from stream string + * and direction */ + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (!w->sname) + continue; + + if (w->id != type) + continue; + + if (strstr(w->sname, stream_name)) + goto found; + } + dev_err(dapm->dev, "root widget not found\n"); + return 0; + +found: + dapm->num_valid_paths = 0; + *list = NULL; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = dapm_get_playback_paths(dapm, w, list); + else + paths = dapm_get_capture_paths(dapm, w, list); + + dapm_clear_paths(dapm); + return paths; +} + +/** + * snd_soc_dapm_get_connected_widgets_name - query audio path and it's widgets. + * @dapm: the dapm context. + * @name: initial widget name. + * @list: list of active widgets for this stream. + * @stream: stream direction. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial widget specified by name. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_get_connected_widgets_name(struct snd_soc_dapm_context *dapm, + const char *name, struct snd_soc_dapm_widget_list **list, int stream) +{ + struct snd_soc_dapm_widget *w; + int paths; + + /* get stream root widget AIF, DAC or ADC from stream string + * and direction */ + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (strstr(w->name, name)) + goto found; + } + dev_err(dapm->dev, "root widget %s not found\n", name); + return 0; + +found: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = dapm_get_playback_paths(dapm, w, list); + else + paths = dapm_get_capture_paths(dapm, w, list); + + dapm_clear_paths(dapm); + return paths; +} + /** * snd_soc_dapm_set_bias_level - set the bias level for the system * @dapm: DAPM context @@ -576,15 +938,6 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } -/* reset 'walked' bit for each dapm path */ -static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm) -{ - struct snd_soc_dapm_path *p; - - list_for_each_entry(p, &dapm->card->paths, list) - p->walked = 0; -} - /* We implement power down on suspend by checking the power state of * the ALSA card - when we are suspending the ALSA state for the card * is set to D3. -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2
by Kuninori Morimoto 07 Jul '11

07 Jul '11

mask didn't cover update-data Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch based on asoc/for-3.1 sound/soc/codecs/ak4642.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 4be0570..65f4604 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -357,7 +357,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) default: return -EINVAL; } - snd_soc_update_bits(codec, PW_MGMT2, MS, data); + snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); /* format type */ -- 1.7.4.1

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Re: [alsa-devel] [PATCH] ASoC: pxa2xx-pcm: remove unused variable 'dai'
by Mark Brown 07 Jul '11

07 Jul '11

On Wed, Jul 06, 2011 at 09:20:42PM +0800, Axel Lin wrote: > Remove unused variable 'dai' to eliminate below warning. Applied, thanks.

1 0

[alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 07 Jul '11

07 Jul '11

I have to revisit an issue we've discussed in length around a year ago and which still remains unsolved. I've been getting feedback from more users of my driver snd-usb-caiaq who report the issue found in bug #15580 in the kernel bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=15580 Let me quickly summarize the current state of this issue as I see it. The problem seems to be that certain 64bit chipsets can't deal with the fact that an URB's transfer_buffer is allocated with kmalloc(GFP_KERNEL). The effect is that kmalloc() is very likely to hand out memory which is not addressable by devices that are connected via 32bit PCI busses, such as EHCI controllers. In theory, DMA bounce buffers should be installed in such cases, or the IOMMU would be in charge to re-map these buffers to suitable locations, but for at least two people who have reported the issue, this obviously fails. So my approach was to force the driver using memory that is DMA coherent, which leads to 32bit addressable memory if the device's DMA_MASK is set accordingly. Other drivers do the same, and the patch for doing this is found here: http://lkml.org/lkml/2010/5/7/61. Users that see the bug all report that this patch fixes their problem with my driver. However, it was rejected back then as other developers said it was fixing the wrong end, and that the driver doesn't necessarily need coherent memory. Takashi recently posted a patch to the bugzilla entry which uses a different approach: it introduces a function to determine suitable GFP flags for USB devices, and passes __DMA32 to kmalloc() eventually. However, using this flags directly with the SLUB allocator is illegal and causes a BUG() in mm/slub.c, cache_grow(). The question now is how to proceed. I see three possible ways. 1. Find a way to allocate 32bit-aware memory with kmalloc(), following Takashi's idea 2. Find out exactly why these machines fail to install bounce buffers or set up their IOMMU correctly 3. re-activate the currently disabled functions usb_buffer_{map,unmap,sync} functions and let the USB stack do the memory mapping I'm not really deep into all the memory management code and even less into how the IOMMU and bounce buffers work, so I need some help here. Can anyone elaborate what would be the best option? I really want to get this fixed now, and provide a driver that works for everyone. What really puzzles me is that this doesn't hit a lot more people with other drivers. Thanks, Daniel

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Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 07 Jul '11

07 Jul '11

On Thu, Jul 7, 2011 at 2:33 PM, Oliver Neukum <oliver(a)neukum.org> wrote: > PS: Do you still see this if you enable 64bit DMA for EHCI? The problem is that I personally don't see that issue at all. I even installed 4GB of RAM to my development machine last year to be able to reproduce this, but I can't, even when the memory allocator is under heavy load. The only people who see this effect are Pedro Ribeiro and William Light (both in Cc:), and both have been very helpful in trying patches and reporting back in detail. Which instructions could we probably give to these people to finally hunt this issue down? Daniel

1 0

Re: [alsa-devel] [PATCH] ASoC: pxa2xx-pcm: remove unused variable 'dai'
by Liam Girdwood 07 Jul '11

07 Jul '11

On 06/07/11 14:20, Axel Lin wrote: > Remove unused variable 'dai' to eliminate below warning. > > CC sound/soc/pxa/pxa2xx-pcm.o > sound/soc/pxa/pxa2xx-pcm.c: In function 'pxa2xx_soc_pcm_new': > sound/soc/pxa/pxa2xx-pcm.c:91: warning: unused variable 'dai' > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/pxa/pxa2xx-pcm.c | 1 - > 1 files changed, 0 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c > index da28394..c430600 100644 > --- a/sound/soc/pxa/pxa2xx-pcm.c > +++ b/sound/soc/pxa/pxa2xx-pcm.c > @@ -88,7 +88,6 @@ static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32); > static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > - struct snd_soc_dai *dai = rtd->cpu_dai; > struct snd_pcm *pcm = rtd->pcm; > int ret = 0; > Acked-by: Liam Girdwood <lrg(a)ti.com>

1 0

Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Takashi Iwai 07 Jul '11

07 Jul '11

At Thu, 7 Jul 2011 16:54:18 +0900, Toshiharu Okada wrote: > > Date: Wed, 06 Jul 2011 13:06:15 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > > > >I just took a quick glance over the code, and wonder whether this kind > >of driver would fit better with ASoC framework. > >Have you considered the implementation on ASoC? > > Thank you for your comment. > > I referred to the following files and implemented this driver. > > http://www.mirrorservice.org/sites/ftp.sourceforge.net/pub/sourceforge/s/pr… > sound/drivers/sta2x11i2s.c > > Therefore, I am not considering about ASoC. > Do I need to fit this driver to an ASoC framework for an upstream? Yes, I guess this would be a better choice in your case. Takashi

1 0

[alsa-devel] hda-intel front headphone (8086:27d8 1043:83bc)
by Alex 07 Jul '11

07 Jul '11

Hi, I run a Debian/sid on an Asus P5KPL-SE mainboard, the snd-hda-intel seem to work fine, ever if it run with guess method: Unknown hardware: "HDA-Intel" "Realtek ALC887" "HDA:10ec0887,104383bc,00100202" "0x1043" "0x83bc" Hardware is initialized using a guess method The only annoying thing is about the front headphone, every mixer application show the front headphone item but doesn't have adjustable level and I hear nothing on headphone. Any hint? http://www.alsa-project.org/db/?f=d6bc3005fab4683f2606e48f2bc909258a4ee3e3 Thanks in advance Regards

2 3

[alsa-devel] [PATCH] ASoC: twl6040: Add back support for legacy mode
by Peter Ujfalusi 07 Jul '11

07 Jul '11

The legacy mode has been accidentaly removed by commit: ASoC: twl6040: add all ABE DAIs Add back the twl6040-hifi dai. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Tony, Mark, Liam, this patch fixes the accidental removal of the twl6040-hifi dai from the codec driver. The series, which this patch depends on not yet pulled (I have the branch ready), since I'm still waiting for few acks for it. The MFD series is mostly OK, but the threaded interrupt related followup waits for acks, and also the PLL restructuring series is in waiting state. Is it OK, if I queue up this patch as well for the same pull request, and goes with the rest through l-o (since it is on top of those changes)? I could try to squash it down, if you prefer to the offending commit - I hope it is not breaking the rebase too much... Thank you, Peter sound/soc/codecs/twl6040.c | 18 ++++++++++++++++++ 1 files changed, 18 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index d4b944c..443032b 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1464,6 +1464,24 @@ static struct snd_soc_dai_ops twl6040_dai_ops = { static struct snd_soc_dai_driver twl6040_dai[] = { { + .name = "twl6040-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}, +{ .name = "twl6040-ul", .capture = { .stream_name = "Capture", -- 1.7.6

2 1

[alsa-devel] [PATCH v3 0/5] MFD/input/ASoC: twl6040: irq registration changes
by Peter Ujfalusi 06 Jul '11

06 Jul '11

Hello, Changes since v2: - Fix the check in the slaves driver for the validity of the received irq - Mark Brown's ack added to patch 3 Intro mail froim v1: As Samuel, and Felipe requested for the twl6040 MFD series: http://marc.info/?l=linux-kernel&m=130978512225430&w=2 http://marc.info/?l=linux-omap&m=130978403224218&w=2 This is the update to remove the obfuscation around the threaded irq request in the twl6040 driver and it's slaves. With this change the slaves uses directly the threaded irq request, and the core driver passes the irq number for the slaves to use (configured runtime, so machine drivers do not need to be updated). Regards, Peter PS: for the remaining comments from Samuel I have changed the series, since the requested change was small. I'm not intended to send the series again, unless it is requested (again the change from v6 is minimal - few lines has been removed/replaced). --- Peter Ujfalusi (5): MFD: twl6040: Use resource to provide irq number for slaves input: twl6040-vibra: Do not use wrapper for irq request ASoC: twl6040: Do not use wrapper for irq request MFD: twl6040: Demand valid interrupt configuration MFD: twl6040: Remove wrapper for threaded irq request drivers/input/misc/twl6040-vibra.c | 17 ++++++--- drivers/mfd/twl6040-core.c | 69 ++++++++++++++++++++++++----------- drivers/mfd/twl6040-irq.c | 16 +-------- include/linux/mfd/twl6040.h | 22 ----------- sound/soc/codecs/twl6040.c | 19 +++++++--- 5 files changed, 74 insertions(+), 69 deletions(-) -- 1.7.6

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Re: [alsa-devel] [RFC] ASoC: Add machine driver for i.MX SGTL500 based boards
by Mark Brown 06 Jul '11

06 Jul '11

On Mon, Jul 04, 2011 at 03:41:55PM -0300, Fabio Estevam wrote: > sgtl5000 1-000a: sgtl5000 revision 17 > sgtl5000 1-000a: asoc: failed to probe CODEC sgtl5000.1-000a: -22 > asoc: failed to instantiate card sgtl5000: -22 Have you looked to see where the error is coming from? > + ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_LRCLK, > + params_rate(params), 0); > + > + if (ret) { > + pr_err("%s: failed setting codec sysclk\n", __func__); > + return ret; > + } Fix this in the CODEC driver, copying this code into the machine drivers is clearly not useful. > +static int __init imx_sgtl5000_init(void) > +{ > + int ret; > + > + imx_sgtl5000_snd_device = platform_device_alloc("soc-audio", -1); > + if (!imx_sgtl5000_snd_device) > + return -ENOMEM; We're not accepting new drivers using soc-audio, you should use a regular platform device and register the card with snd_soc_register_card().

2 4

[alsa-devel] [PATCH] ARM i.MX dma: Fix burstsize settings
by Sascha Hauer 06 Jul '11

06 Jul '11

dmaengine expects the maxburst parameter in words, not bytes. The imxdma driver and its users do this wrong. Fix this. As a side note the imx-pcm-dma-mx2 driver was 'fixed' to work with imx-dma. This broke the driver with imx-sdma support which correctly takes the maxburst parameter in words. This patch puts the sdma based sound back to work. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- I made this as a cross subsystem patch to maintain bisectability. I suggest to push it either via my tree or the dmaengine tree. drivers/dma/imx-dma.c | 3 ++- drivers/mmc/host/mxcmmc.c | 8 ++++---- sound/soc/imx/imx-pcm-dma-mx2.c | 4 ++-- 3 files changed, 8 insertions(+), 7 deletions(-) diff --git a/drivers/dma/imx-dma.c b/drivers/dma/imx-dma.c index e18eaab..d99f71c 100644 --- a/drivers/dma/imx-dma.c +++ b/drivers/dma/imx-dma.c @@ -135,7 +135,8 @@ static int imxdma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, if (ret) return ret; - imx_dma_config_burstlen(imxdmac->imxdma_channel, imxdmac->watermark_level); + imx_dma_config_burstlen(imxdmac->imxdma_channel, + imxdmac->watermark_level * imxdmac->word_size); return 0; default: diff --git a/drivers/mmc/host/mxcmmc.c b/drivers/mmc/host/mxcmmc.c index cc20e02..14aa213 100644 --- a/drivers/mmc/host/mxcmmc.c +++ b/drivers/mmc/host/mxcmmc.c @@ -715,13 +715,13 @@ static void mxcmci_set_ios(struct mmc_host *mmc, struct mmc_ios *ios) int burstlen, ret; /* - * use burstlen of 64 in 4 bit mode (--> reg value 0) - * use burstlen of 16 in 1 bit mode (--> reg value 16) + * use burstlen of 64 (16 words) in 4 bit mode (--> reg value 0) + * use burstlen of 16 (4 words) in 1 bit mode (--> reg value 16) */ if (ios->bus_width == MMC_BUS_WIDTH_4) - burstlen = 64; - else burstlen = 16; + else + burstlen = 4; if (mxcmci_use_dma(host) && burstlen != host->burstlen) { host->burstlen = burstlen; diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 4173b3d..43fdc24f 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -110,12 +110,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, slave_config.direction = DMA_TO_DEVICE; slave_config.dst_addr = dma_params->dma_addr; slave_config.dst_addr_width = buswidth; - slave_config.dst_maxburst = dma_params->burstsize * buswidth; + slave_config.dst_maxburst = dma_params->burstsize; } else { slave_config.direction = DMA_FROM_DEVICE; slave_config.src_addr = dma_params->dma_addr; slave_config.src_addr_width = buswidth; - slave_config.src_maxburst = dma_params->burstsize * buswidth; + slave_config.src_maxburst = dma_params->burstsize; } ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config); -- 1.7.5.3 -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

2 1

[alsa-devel] [PATCH] ASoC: Conditionalize the enable of WM8994 ADC TDM mode
by Mark Brown 06 Jul '11

06 Jul '11

Future devices will not benefit from this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 16 ++++++++++++---- 1 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c0d8686..299f6eb 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3114,10 +3114,18 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT, 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT); - /* Unconditionally enable AIF1 ADC TDM mode; it only affects - * behaviour on idle TDM clock cycles. */ - snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1, - WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM); + /* Unconditionally enable AIF1 ADC TDM mode on chips which can + * use this; it only affects behaviour on idle TDM clock + * cycles. */ + switch (control->type) { + case WM8994: + case WM8958: + snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1, + WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM); + break; + default: + break; + } wm8994_update_class_w(codec); -- 1.7.5.4

2 1

Re: [alsa-devel] Wrong bitrate from usb microphone
by Takashi Iwai 06 Jul '11

06 Jul '11

At Tue, 05 Jul 2011 15:34:06 +0200, Pali Rohár wrote: > > Hello, > > I have USB webcam with integrated USB microphone. Microphone does not working > correctly because kernel module snd-usb-audio set wrong bitrate. > > Problem appered after this commit: > http://git.kernel.org/?p=linux/kernel/git/stable/linux-2.6.39.y.git;a=commi… > > When I reverted patch in above commit microphone works properly. > This commit set bitrate to 8000Hz for 041e:4068 but my webcam (propably) > produce 16000Hz. So what to do with my webcam? It needs to revert above patch! > > Here is output from lsusb of my webcam: > Bus 002 Device 003: ID 041e:4068 Creative Technology, Ltd Live! Cam Notebook > [VF0470] > Device Descriptor: > bLength 18 > bDescriptorType 1 > bcdUSB 1.10 > bDeviceClass 0 (Defined at Interface level) > bDeviceSubClass 0 > bDeviceProtocol 0 > bMaxPacketSize0 8 > idVendor 0x041e Creative Technology, Ltd > idProduct 0x4068 Live! Cam Notebook [VF0470] > bcdDevice 1.00 > iManufacturer 1 > iProduct 2 > iSerial 0 > bNumConfigurations 1 Could you give the lsusb -v output? Arseniy, also check whether the current code is really OK? thanks, Takashi

1 0

[alsa-devel] [PATCH] Cirrus Logic CS4210-rev-A1 support.
by Tim Howe 06 Jul '11

06 Jul '11

From: Tim Howe <timothyc.howe(a)gmail.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- sound/pci/hda/patch_cirrus.c | 841 ++++++++++++++++++++++++++++++++++++++++-- 1 files changed, 805 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index c7b5ca2..7e09722 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -51,7 +51,7 @@ struct cs_spec { unsigned int cur_adc_format; hda_nid_t dig_in; - const struct hda_bind_ctls *capture_bind[2]; + struct hda_bind_ctls *capture_bind[2]; unsigned int gpio_mask; unsigned int gpio_dir; @@ -61,6 +61,12 @@ struct cs_spec { unsigned int hp_detect:1; unsigned int mic_detect:1; + /* CS421x */ + unsigned int spdif_detect:1; + unsigned int sense_b:1; + hda_nid_t vendor_nid; + struct hda_input_mux input_mux; + unsigned int last_input; }; /* available models */ @@ -114,18 +120,20 @@ enum { static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0, + return snd_hda_codec_read(codec, spec->vendor_nid, 0, AC_VERB_GET_PROC_COEF, 0); } static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx, unsigned int coef) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_PROC_COEF, coef); } @@ -231,7 +239,7 @@ static int cs_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, /* */ -static const struct hda_pcm_stream cs_pcm_analog_playback = { +static struct hda_pcm_stream cs_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -242,7 +250,7 @@ static const struct hda_pcm_stream cs_pcm_analog_playback = { }, }; -static const struct hda_pcm_stream cs_pcm_analog_capture = { +static struct hda_pcm_stream cs_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -252,7 +260,7 @@ static const struct hda_pcm_stream cs_pcm_analog_capture = { }, }; -static const struct hda_pcm_stream cs_pcm_digital_playback = { +static struct hda_pcm_stream cs_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -264,7 +272,7 @@ static const struct hda_pcm_stream cs_pcm_digital_playback = { }, }; -static const struct hda_pcm_stream cs_pcm_digital_capture = { +static struct hda_pcm_stream cs_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -331,8 +339,8 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx) struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t pin = cfg->inputs[idx].pin; - unsigned int val; - if (!is_jack_detectable(codec, pin)) + unsigned int val = snd_hda_query_pin_caps(codec, pin); + if (!(val & AC_PINCAP_PRES_DETECT)) return 0; val = snd_hda_codec_get_pincfg(codec, pin); return (snd_hda_get_input_pin_attr(val) != INPUT_PIN_ATTR_INT); @@ -346,10 +354,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { - hda_nid_t pins[2]; + hda_nid_t pins[HDA_MAX_NUM_INPUTS]; unsigned int type; int j, nums; - type = get_wcaps_type(get_wcaps(codec, nid)); + type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) + >> AC_WCAP_TYPE_SHIFT; if (type != AC_WID_AUD_IN) continue; nums = snd_hda_get_connections(codec, nid, pins, @@ -558,10 +567,10 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, const char *name; int err, index; struct snd_kcontrol *kctl; - static const char * const speakers[] = { + static char *speakers[] = { "Front Speaker", "Surround Speaker", "Bass Speaker" }; - static const char * const line_outs[] = { + static char *line_outs[] = { "Front Line-Out", "Surround Line-Out", "Bass Line-Out" }; @@ -641,7 +650,7 @@ static int build_output(struct hda_codec *codec) /* */ -static const struct snd_kcontrol_new cs_capture_ctls[] = { +static struct snd_kcontrol_new cs_capture_ctls[] = { HDA_BIND_SW("Capture Switch", 0), HDA_BIND_VOL("Capture Volume", 0), }; @@ -709,7 +718,7 @@ static int cs_capture_source_put(struct snd_kcontrol *kcontrol, return change_cur_input(codec, idx, 0); } -static const struct snd_kcontrol_new cs_capture_source = { +static struct snd_kcontrol_new cs_capture_source = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -718,7 +727,7 @@ static const struct snd_kcontrol_new cs_capture_source = { .put = cs_capture_source_put, }; -static const struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec, +static struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec, struct hda_ctl_ops *ops) { struct cs_spec *spec = codec->spec; @@ -821,8 +830,7 @@ static int build_digital_output(struct hda_codec *codec) if (!spec->multiout.dig_out_nid) return 0; - err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid, - spec->multiout.dig_out_nid); + err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); if (err < 0) return err; err = snd_hda_create_spdif_share_sw(codec, &spec->multiout); @@ -847,14 +855,15 @@ static void cs_automute(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int hp_present; + unsigned int caps, hp_present; hda_nid_t nid; int i; hp_present = 0; for (i = 0; i < cfg->hp_outs; i++) { nid = cfg->hp_pins[i]; - if (!is_jack_detectable(codec, nid)) + caps = snd_hda_query_pin_caps(codec, nid); + if (!(caps & AC_PINCAP_PRES_DETECT)) continue; hp_present = snd_hda_jack_detect(codec, nid); if (hp_present) @@ -923,7 +932,7 @@ static void init_output(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); if (!cfg->speaker_outs) continue; - if (is_jack_detectable(codec, nid)) { + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | HP_EVENT); @@ -982,7 +991,7 @@ static void init_input(struct hda_codec *codec) cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); } -static const struct hda_verb cs_coef_init_verbs[] = { +static struct hda_verb cs_coef_init_verbs[] = { {0x11, AC_VERB_SET_PROC_STATE, 1}, {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG}, {0x11, AC_VERB_SET_PROC_COEF, @@ -1016,7 +1025,7 @@ static const struct hda_verb cs_coef_init_verbs[] = { * blocks, which will alleviate the issue. */ -static const struct hda_verb cs_errata_init_verbs[] = { +static struct hda_verb cs_errata_init_verbs[] = { {0x01, AC_VERB_SET_POWER_STATE, 0x00}, /* AFG: D0 */ {0x11, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ @@ -1125,7 +1134,7 @@ static void cs_unsol_event(struct hda_codec *codec, unsigned int res) } } -static const struct hda_codec_ops cs_patch_ops = { +static struct hda_codec_ops cs_patch_ops = { .build_controls = cs_build_controls, .build_pcms = cs_build_pcms, .init = cs_init, @@ -1165,7 +1174,7 @@ static const char * const cs420x_models[CS420X_MODELS] = { }; -static const struct snd_pci_quirk cs420x_cfg_tbl[] = { +static struct snd_pci_quirk cs420x_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0x0ac0, "MacBookPro 5,3", CS420X_MBP53), SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), @@ -1179,7 +1188,7 @@ struct cs_pincfg { u32 val; }; -static const struct cs_pincfg mbp53_pincfgs[] = { +static struct cs_pincfg mbp53_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100141 }, { 0x0b, 0x90100140 }, @@ -1193,7 +1202,7 @@ static const struct cs_pincfg mbp53_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg mbp55_pincfgs[] = { +static struct cs_pincfg mbp55_pincfgs[] = { { 0x09, 0x012b4030 }, { 0x0a, 0x90100121 }, { 0x0b, 0x90100120 }, @@ -1207,7 +1216,7 @@ static const struct cs_pincfg mbp55_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg imac27_pincfgs[] = { +static struct cs_pincfg imac27_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100140 }, { 0x0b, 0x90100142 }, @@ -1221,22 +1230,22 @@ static const struct cs_pincfg imac27_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { +static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { [CS420X_MBP53] = mbp53_pincfgs, [CS420X_MBP55] = mbp55_pincfgs, [CS420X_IMAC27] = imac27_pincfgs, }; -static void fix_pincfg(struct hda_codec *codec, int model) +static void fix_pincfg(struct hda_codec *codec, int model, + struct cs_pincfg **pin_configs) { - const struct cs_pincfg *cfg = cs_pincfgs[model]; + const struct cs_pincfg *cfg = pin_configs[model]; if (!cfg) return; for (; cfg->nid; cfg++) snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } - static int patch_cs420x(struct hda_codec *codec) { struct cs_spec *spec; @@ -1247,11 +1256,13 @@ static int patch_cs420x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + spec->vendor_nid = CS420X_VENDOR_NID; + spec->board_config = snd_hda_check_board_config(codec, CS420X_MODELS, cs420x_models, cs420x_cfg_tbl); if (spec->board_config >= 0) - fix_pincfg(codec, spec->board_config); + fix_pincfg(codec, spec->board_config, cs_pincfgs); switch (spec->board_config) { case CS420X_IMAC27: @@ -1278,18 +1289,776 @@ static int patch_cs420x(struct hda_codec *codec) return err; } +/* + Cirrus Logic CS4210 + + 1 DAC => HP(sense) / Speakers, + 1 ADC <= LineIn(sense) / MicIn / DMicIn, + 1 SPDIF OUT => SPDIF Trasmitter(sense) +*/ +#define CS4210_DAC_NID 0x02 +#define CS4210_ADC_NID 0x03 +#define CS421X_VENDOR_NID 0x0B +#define CS421X_DMIC_PIN_NID 0x09 /* Port E */ +#define CS421X_SPDIF_PIN_NID 0x0A /* Port H */ + +#define CS421X_IDX_DEV_CFG 0x01 +#define CS421X_IDX_ADC_CFG 0x02 +#define CS421X_IDX_DAC_CFG 0x03 +#define CS421X_IDX_SPK_CTL 0x04 + +#define SPDIF_EVENT 0x04 + +/* CS4210 boards */ +enum { + CS421X_CDB4210, + CS421X_MODELS +}; + +/* CS4210 board names */ +static const char *cs421x_models[CS421X_MODELS] = { + [CS421X_CDB4210] = "cdb4210", +}; + +static struct snd_pci_quirk cs421x_cfg_tbl[] = { + /* Test Intel board + CDB2410 */ + SND_PCI_QUIRK(0x8086, 0x5001, "DP45SG/CDB4210", CS421X_CDB4210), + {} /* terminator */ +}; + +/* CS4210 board pinconfigs */ +/* Default CS4210 (CDB4210)*/ +static struct cs_pincfg cdb4210_pincfgs[] = { + { 0x05, 0x0321401f }, + { 0x06, 0x90170010 }, + { 0x07, 0x03813031 }, + { 0x08, 0xb7a70037 }, + { 0x09, 0xb7a6003e }, + { 0x0a, 0x034510f0 }, + {} /* terminator */ +}; + +static struct cs_pincfg *cs421x_pincfgs[CS421X_MODELS] = { + [CS421X_CDB4210] = cdb4210_pincfgs, +}; + +static struct hda_verb cs421x_coef_init_verbs[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 1}, + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DEV_CFG}, + /* + Disable Coefficient Index Auto-Increment(DAI)=1, + PDREF=0 + */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0001 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_ADC_CFG}, + /* ADC SZCMode = Digital Soft Ramp */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0002 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DAC_CFG}, + {0x0B, AC_VERB_SET_PROC_COEF, + (0x0002 /* DAC SZCMode = Digital Soft Ramp */ + | 0x0004 /* Mute DAC on FIFO error */ + | 0x0008 /* Enable DAC High Pass Filter */ + )}, + {} /* terminator */ +}; + +/* Errata: CS4210 rev A1 Silicon + * + * http://www.cirrus.com/en/pubs/errata/ + * + * Description: 1. Performance degredation is present in the ADC. + * 2. Speaker output is not completely muted upon HP detect. + * 3. Noise is present when clipping occurs on the amplified + * speaker outputs. + * + * Workaround: The following verb sequence written to the registers during + * initialization will correct the issues listed above. + */ + +static struct hda_verb cs421x_coef_init_verbs_A1_silicon_fixes[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0006}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x9999}, /* Test mode: on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x000A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x14CB}, /* Chop double */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0011}, + {0x0B, AC_VERB_SET_PROC_COEF, 0xA2D0}, /* Increase ADC current */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x02A9}, /* Mute speaker */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001B}, + {0x0B, AC_VERB_SET_PROC_COEF, 0X1006}, /* Remove noise */ + + {} /* terminator */ +}; + +#include <sound/tlv.h> +/* Speaker Amp Gain is controlled by the vendor widget's coef 4 */ +static const DECLARE_TLV_DB_SCALE(cs421x_speaker_boost_db_scale, 900, 300, 0); + +static int cs421x_boost_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 3; + return 0; +} + +static int cs421x_boost_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL) & 0x0003; + return 0; +} + +static int cs421x_boost_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + unsigned int vol = ucontrol->value.integer.value[0]; + unsigned int coef = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL); + + coef &= ~0x0003; + coef |= (vol & 0x0003); + cs_vendor_coef_set(codec, CS421X_IDX_SPK_CTL, coef); + + return 1; +} + +static const struct snd_kcontrol_new cs421x_speaker_bost_ctl = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Speaker Boost Playback Volume", + .info = cs421x_boost_vol_info, + .get = cs421x_boost_vol_get, + .put = cs421x_boost_vol_put, + .tlv = { .p = cs421x_speaker_boost_db_scale }, +}; + +/* + CS4210 auto-output redirect + HP/SPK/SPDIF +*/ +static void cs421x_auto_output(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int caps, hp_present, spdif_present; + hda_nid_t nid; + int i; + + spdif_present = 0; + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + caps = snd_hda_query_pin_caps(codec, nid); + if (caps & AC_PINCAP_PRES_DETECT) { + /* + TODO: SPDIF output redirect when SENSE_B is enabled. + Shared (SENSE_A) jack (e.g HP/mini-TOSLINK) + assumed. + */ + if (snd_hda_jack_detect(codec, nid) + /* && spec->sense_b */) + spdif_present = 1; + } + } + + hp_present = 0; + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + caps = snd_hda_query_pin_caps(codec, nid); + if (!(caps & AC_PINCAP_PRES_DETECT)) + continue; + hp_present = snd_hda_jack_detect(codec, nid); + if (hp_present) + break; + } + + /* mute speakers is spdif or hp jack is plugged in */ + for (i = 0; i < cfg->speaker_outs; i++) { + nid = cfg->speaker_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + (hp_present | spdif_present) ? 0 : PIN_OUT); + } + + /* mute HPs if spdif jack (SENSE_B) is present */ + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + (spdif_present & spec->sense_b)? 0 : PIN_HP); + } + + /* SPDIF TX on/off */ + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spdif_present ? PIN_OUT:0); + + } + /* Update board GPIOs if neccessary ... */ +} + +/* + CS4210 auto-input redirect + Switch max 3 inputs of a single ADC (nid 3) +*/ +static void cs421x_auto_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid; + unsigned int present; + + nid = cfg->inputs[spec->automic_idx].pin; + present = snd_hda_jack_detect(codec, nid); + + if (present) { + spec->last_input = spec->cur_input; + spec->cur_input = spec->automic_idx; + } else { + spec->cur_input = spec->last_input; + } + + snd_hda_codec_write_cache(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); +} + +static void cs421x_pinmux_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + unsigned int def_conf, coef; + + /* GPIO, DMIC_SCL, DMIC_SDA and SENSE_B are multiplexed */ + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + + if (spec->gpio_mask) + coef |= 0x0008; /* B1,B2 are GPIOs */ + else + coef &= ~0x0008; + + if (spec->sense_b) + coef |= 0x0010; /* B2 is SENSE_B, not inverted */ + else + coef &= ~0x0010; + + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + if ((spec->gpio_mask || spec->sense_b) && + is_active_pin(codec, CS421X_DMIC_PIN_NID)) + { + /* + GPIO or SENSE_B forced - disconnect the DMIC pin. + */ + def_conf = snd_hda_codec_get_pincfg(codec, CS421X_DMIC_PIN_NID); + def_conf &= ~AC_DEFCFG_PORT_CONN; + def_conf |= (AC_JACK_PORT_NONE << AC_DEFCFG_PORT_CONN_SHIFT); + snd_hda_codec_set_pincfg(codec, CS421X_DMIC_PIN_NID, def_conf); + } +} + +/* */ +static void init_cs421x_output(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + /* mute first */ + for (i = 0; i < spec->multiout.num_dacs; i++) + snd_hda_codec_write(codec, spec->multiout.dac_nids[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + if (spec->multiout.hp_nid) + snd_hda_codec_write(codec, spec->multiout.hp_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) { + if (!spec->multiout.extra_out_nid[i]) + break; + snd_hda_codec_write(codec, spec->multiout.extra_out_nid[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + } + + /* set appropriate pin controls */ + for (i = 0; i < cfg->line_outs; i++) + snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + /* HP */ + for (i = 0; i < cfg->hp_outs; i++) { + hda_nid_t nid = cfg->hp_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + if (!cfg->speaker_outs) + continue; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | HP_EVENT); + spec->hp_detect = 1; + } + } + + /* Speaker */ + for (i = 0; i < cfg->speaker_outs; i++) + snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + + /* SPDIF is enabled on presence detect */ + if (spec->hp_detect || spec->spdif_detect) + cs421x_auto_output(codec); +} + +static void init_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + unsigned int ctl; + hda_nid_t pin = cfg->inputs[i].pin; + + if (!spec->adc_nid[i]) + continue; + /* set appropriate pin control and mute first */ + ctl = PIN_IN; + if (cfg->inputs[i].type == AUTO_PIN_MIC) { + unsigned int caps = snd_hda_query_pin_caps(codec, pin); + caps >>= AC_PINCAP_VREF_SHIFT; + if (caps & AC_PINCAP_VREF_80) + ctl = PIN_VREF80; + } + + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, ctl); + snd_hda_codec_write(codec, spec->adc_nid[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(spec->adc_idx[i])); + + if (spec->mic_detect && spec->automic_idx == i) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | MIC_EVENT); + } + } + + if (spec->mic_detect) + cs421x_auto_input(codec); + else { + spec->cur_adc = spec->adc_nid[spec->cur_input]; + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); + } + +} + +static void init_cs421x_digital(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + + for (i = 0; i < cfg->dig_outs; i++) { + hda_nid_t nid = cfg->dig_out_pins[i]; + if (!cfg->speaker_outs) + continue; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) + { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | SPDIF_EVENT); + spec->spdif_detect = 1; + } + } +} + +static int cs421x_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + + snd_hda_sequence_write(codec, cs421x_coef_init_verbs); + snd_hda_sequence_write(codec, cs421x_coef_init_verbs_A1_silicon_fixes); + + cs421x_pinmux_init(codec); + + if (spec->gpio_mask) { + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, + spec->gpio_mask); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + spec->gpio_dir); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_data); + } + + init_cs421x_output(codec); + init_cs421x_input(codec); + init_cs421x_digital(codec); + + return 0; +} + +/* + * CS4210 Input MUX (1 ADC) + */ +static int cs421x_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_info(&spec->input_mux, uinfo); +} + +static int cs421x_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_input; + return 0; +} + +static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol, + spec->adc_nid[0], &spec->cur_input); + +} + +static struct snd_kcontrol_new cs421x_capture_source = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = cs421x_mux_enum_info, + .get = cs421x_mux_enum_get, + .put = cs421x_mux_enum_put, +}; + +static int cs421x_add_input_volume_control(struct hda_codec *codec, int item) +{ + struct cs_spec* spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + const struct hda_input_mux *imux = &spec->input_mux; + hda_nid_t pin = cfg->inputs[item].pin; + struct snd_kcontrol *kctl; + u32 caps; + + if (!(get_wcaps(codec, pin) & AC_WCAP_IN_AMP)) + return 0; + + caps = query_amp_caps(codec, pin, HDA_INPUT); + caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + if (caps <= 1) + return 0; + + return add_volume(codec, imux->items[item].label, 0, + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT), 1, &kctl); +} + +/* add a (input-boost) volume control to the given input pin */ +static int build_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct hda_input_mux *imux = &spec->input_mux; + int i, err, type_idx; + const char *label; + + if (!spec->num_inputs) + return 0; + + /* make bind-capture */ + spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw); + spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol); + for (i = 0; i < 2; i++) { + struct snd_kcontrol *kctl; + int n; + if (!spec->capture_bind[i]) + return -ENOMEM; + kctl = snd_ctl_new1(&cs_capture_ctls[i], codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = (long)spec->capture_bind[i]; + err = snd_hda_ctl_add(codec, 0, kctl); + if (err < 0) + return err; + for (n = 0; n < AUTO_PIN_LAST; n++) { + if (!spec->adc_nid[n]) + continue; + err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]); + if (err < 0) + return err; + } + } + + /* Add Input MUX Items + Capture Volume/Switch */ + for (i = 0; i < spec->num_inputs; i++) { + label = hda_get_autocfg_input_label(codec, cfg, i); + snd_hda_add_imux_item(imux, label, spec->adc_idx[i], &type_idx); + + err = cs421x_add_input_volume_control(codec, i); + if (err < 0) + return err; + } + + /* + Add 'Capture Source' Switch if + * 2 inputs and no mic detec + * 3 inputs + */ + if ((spec->num_inputs == 2 && !spec->mic_detect) || + (spec->num_inputs == 3) ) + { + err = snd_hda_ctl_add(codec, spec->adc_nid[0], + snd_ctl_new1(&cs421x_capture_source, codec)); + if (err < 0) + return err; + } + + return 0; +} + +/* Single DAC (Mute/Gain) */ +static int build_cs421x_output(struct hda_codec *codec) +{ + hda_nid_t dac = CS4210_DAC_NID; + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct snd_kcontrol *kctl; + int err; + char* name = "HP/Speakers"; + + fix_volume_caps(codec, dac); + if (!spec->vmaster_sw) { + err = add_vmaster(codec, dac); + if (err < 0) + return err; + } + + err = add_mute(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_sw, kctl); + if (err < 0) + return err; + + err = add_volume(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_vol, kctl); + if (err < 0) + return err; + + if (cfg->speaker_outs) { + err = snd_hda_ctl_add(codec, 0, + snd_ctl_new1(&cs421x_speaker_bost_ctl, codec)); + if (err < 0) + return err; + } + return err; +} + +static int cs421x_build_controls(struct hda_codec *codec) +{ + int err; + + err = build_cs421x_output(codec); + if (err < 0) + return err; + err = build_cs421x_input(codec); + if (err < 0) + return err; + err = build_digital_output(codec); + if (err < 0) + return err; + return cs421x_init(codec); +} + +static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res) +{ + switch ((res >> 26) & 0x3f) { + case HP_EVENT: + case SPDIF_EVENT: + cs421x_auto_output(codec); + break; + + case MIC_EVENT: + cs421x_auto_input(codec); + break; + } +} + +static int parse_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t pin = cfg->inputs[i].pin; + spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]); + spec->cur_input = spec->last_input = i; + spec->num_inputs ++; + + /* check whether the automatic mic switch is available */ + if (is_ext_mic(codec, i) && cfg->num_inputs >= 2) { + spec->mic_detect = 1; + spec->automic_idx = i; + } + } + return 0; +} + +static int cs421x_parse_auto_config(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = parse_output(codec); + if (err < 0) + return err; + err = parse_cs421x_input(codec); + if (err < 0) + return err; + err = parse_digital_output(codec); + if (err < 0) + return err; + return 0; +} + +#ifdef SND_HDA_NEEDS_RESUME +/* + Manage PDREF, when transitioning to D3hot + (DAC,ADC) -> D3, PDREF=1, AFG->D3 +*/ +static int cs421x_suspend(struct hda_codec *codec, pm_message_t state) +{ + unsigned int coef; + + snd_hda_shutup_pins(codec); + + snd_hda_codec_write(codec, CS4210_DAC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, CS4210_ADC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + coef |= 0x0004; /* PDREF */ + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + return 0; +} +#endif + +static struct hda_codec_ops cs4210_patch_ops = { + .build_controls = cs421x_build_controls, + .build_pcms = cs_build_pcms, + .init = cs421x_init, + .free = cs_free, + .unsol_event = cs421x_unsol_event, +#ifdef SND_HDA_NEEDS_RESUME + .suspend = cs421x_suspend, +#endif +}; + +static int patch_cs421x(struct hda_codec *codec) +{ + struct cs_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + spec->vendor_nid = CS421X_VENDOR_NID; + + spec->board_config = + snd_hda_check_board_config(codec, CS421X_MODELS, + cs421x_models, cs421x_cfg_tbl); + if (spec->board_config >= 0) + fix_pincfg(codec, spec->board_config, cs421x_pincfgs); + /* + Setup GPIO/SENSE for each board (if used) + */ + switch (spec->board_config) { + case CS421X_CDB4210: + snd_printd("CS4210 board: %s\n", cs421x_models[spec->board_config]); + +/* spec->gpio_mask = 3; + spec->gpio_dir = 3; + spec->gpio_data = 3; +*/ + spec->sense_b = 1; + + break; + } + + /* + Update the GPIO/DMIC/SENSE_B pinmux before the configuration + is auto-parsed. If GPIO or SENSE_B is forced, DMIC input + is disabled. + */ + cs421x_pinmux_init(codec); + + err = cs421x_parse_auto_config(codec); + if (err < 0) + goto error; + + codec->patch_ops = cs4210_patch_ops; + + return 0; + + error: + kfree(codec->spec); + codec->spec = NULL; + return err; +} + /* * patch entries */ -static const struct hda_codec_preset snd_hda_preset_cirrus[] = { +static struct hda_codec_preset snd_hda_preset_cirrus[] = { { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, + { .id = 0x10134210, .name = "CS4210", .patch = patch_cs421x }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:10134206"); MODULE_ALIAS("snd-hda-codec-id:10134207"); +MODULE_ALIAS("snd-hda-codec-id:10134210"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); -- 1.7.0.4

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[alsa-devel] [PATCH 0/6] ASoC/MFD: twl6040: PLL handling changes
by Peter Ujfalusi 06 Jul '11

06 Jul '11

Hello, The following series depends on the v6 of twl6040 MFD series. The aim is to simplify the PLL handling for twl6040: - Move the PLL selection control to the codec driver from machine driver - Sample rate constraint simplification for selected PLL - Avoid multiple, duplicated PLL configuration, which can slow down the audio startup. - Harmonize the PLL ID usage among the drivers, so we avoid redundancy. I have generated this series on top of l-o/devel-cleanup branch + my previous twl6040 MFD/Input/ASoC v6 series, since it depends on the changes introduced there. Is it OK, if Tony will be taking this as well to avoid cross tree merge issues? I can put the series on top of my previous series, and offer Tony a pull link. Regards, Peter --- Peter Ujfalusi (6): ASoC: twl6040: Use neutral name for power mode text/enum ASoC: twl6040: Move PLL selection to codec driver ASoC: twl6040: Simplify sample rate constraint handling ASoC: twl6040: Configure PLL only once MFD: twl6040: Remove enum for PLL tracking ASoC: twl6040: No need to convert the PLL ID drivers/mfd/twl6040-core.c | 26 +++--- include/linux/mfd/twl6040.h | 17 ++--- sound/soc/codecs/twl6040.c | 182 ++++++++++++++++++++++--------------------- sound/soc/codecs/twl6040.h | 1 + sound/soc/omap/sdp4430.c | 47 +---------- 5 files changed, 118 insertions(+), 155 deletions(-) -- 1.7.6

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[alsa-devel] [PATCH] ASoC: wm8731: set the ACTIVE bit if bias ON entered
by Nicolas Ferre 05 Jul '11

05 Jul '11

Resuming from a suspend/resume cycle, the ACTIVE bit was not set even if a stream had actually been processing. Setting this bit at entering SND_SOC_BIAS_ON level will ensure this. Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- Supersed the wrong patch: "[PATCH 1/5] ASoC: wm8731: rework power management" now based on broonie/sound-2.6 git tree branch: for-3.1. sound/soc/codecs/wm8731.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 2dc964b..ee4ec31 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -453,6 +453,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + snd_soc_write(codec, WM8731_ACTIVE, 0x0001); break; case SND_SOC_BIAS_PREPARE: break; -- 1.7.3

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