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Re: [alsa-devel] [Alsa-user] Audio-gd NFB-17.32 (VIA VT1731)
by Torstein Hegge 25 Mar '13

25 Mar '13

On Mon, Mar 25, 2013 at 08:38:16 +0100, Clemens Ladisch wrote: > Torstein Hegge wrote: > > I think we can conclude that the VIA VT1731 has the same issue as the > > C-Media CM6631. > > > > I'll add the VT1731 as one of the devices that needs a reset after > > sample rate change in the next iteration of the sample rate workaround > > patch. > > Why not just do it for all UAC 2 devices? If new devices that needs this just keeps popping up, that might indeed be the better solution. Torstein

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[alsa-devel] No dmix/dsnoop on Intel ICH4/5 by default?
by Rene Herman 24 Mar '13

24 Mar '13

Good day. I'm out of touch of with alsa -- but should dmix/dsnoop still be enabled by default on hardware that doesn't do hardware mixing? It seems it isn't on my Intel ICH4: $ cat /proc/asound/cards 0 [ICH5 ]: ICH4 - Intel ICH5 Intel ICH5 with AD1980 at irq 17 1 [SAA7134 ]: SAA7134 - SAA7134 saa7134[0] at 0xe8200000 irq 22 I believe that it should be from /usr/share/alsa/cards/ICH4.conf, but I don't get mixing on this setup when using the "default" pcm... (which I may not mind at that much, but I was looking at some other sound-related problem and this has me rather confused) This is arch linux, with alsa-lib, -plugins and -utils 1.0.26 installed, using kernel 3.6.11-1-ARCH (ie, standard). Regards, Rene

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[alsa-devel] [PATCH v4] ALSA: usb: Work around CM6631 sample rate change bug
by Torstein Hegge 24 Mar '13

24 Mar '13

The C-Media CM6631 USB-to-S/PDIF receiver doesn't respond to changes in sample rate while the interface is active. Reset the interface after setting the sampling frequency on sample rate changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous stream, causing distorted sound on sample rate changes. The reset is performed for all current C-Media UAC V2.0 devices, including CM6610, CM6620 and CM6631, but has only been tested with CM6631. Signed-off-by: Torstein Hegge <hegge(a)resisty.net> --- Changes since v3: - Use given target rate instead of the new rate read from device when checking if a reset is needed. Some devices (at least Schiit Bifrost) doesn't report a reliable sample rate back right after setting a new rate. This appears to fix the issues with v3 of this patch with the Schiit, as reported by Chris Hermansen in http://thread.gmane.org/gmane.linux.alsa.user/36935/focus=37284 Changes since v2: - More USB ids covered. - int crate -> int cur_rate - unsigned int previous_rate -> int prev_rate sound/usb/clock.c | 50 ++++++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 46 insertions(+), 4 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 5e634a2..60c2775 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -253,7 +253,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, { struct usb_device *dev = chip->dev; unsigned char data[4]; - int err, crate; + int err, cur_rate, prev_rate; int clock = snd_usb_clock_find_source(chip, fmt->clock); if (clock < 0) @@ -266,6 +266,18 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return -ENXIO; } + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data))) < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", + dev->devnum, iface, fmt->altsetting); + return err; + } + + prev_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; @@ -290,9 +302,39 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return err; } - crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); - if (crate != rate) - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + cur_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + if (cur_rate != rate) { + snd_printd(KERN_WARNING + "current rate %d is different from the runtime rate %d\n", + cur_rate, rate); + } + + /* Some devices doesn't respond to sample rate changes while the + * interface is active. */ + if (rate != prev_rate) { + switch (chip->usb_id) { + /* C-Media CM6610/CM6620/CM6631 */ + case USB_ID(0x054c, 0x06cf): /* Sony */ + case USB_ID(0x0b05, 0x17a8): /* Asus Xonar Essence One */ + case USB_ID(0x0d8c, 0x0301): + case USB_ID(0x0d8c, 0x0302): + case USB_ID(0x0d8c, 0x0304): /* CM6631 (Schiit) */ + case USB_ID(0x0d8c, 0x0305): + case USB_ID(0x0d8c, 0x0306): + case USB_ID(0x0d8c, 0x0309): /* CM6631 */ + case USB_ID(0x0d8c, 0x0310): + case USB_ID(0x0d8c, 0x0311): /* CM6610A */ + case USB_ID(0x0d8c, 0x0312): /* CM6620A */ + case USB_ID(0x0d8c, 0x0313): /* CM6630A */ + case USB_ID(0x0d8c, 0x0314): /* CM6631A */ + case USB_ID(0x0d8c, 0x0315): /* CM6632A */ + case USB_ID(0x0d8c, 0x0319): /* CM6631A */ + case USB_ID(0x200c, 0x1030): /* Reloop */ + usb_set_interface(dev, iface, 0); + usb_set_interface(dev, iface, fmt->altsetting); + break; + } + } return 0; } -- 1.7.10.4

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Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
by Jussi Laako 23 Mar '13

23 Mar '13

On 03/22/2013 06:05 PM, Andreas Koch wrote: > The 8-bit raw format is for 2.8MHz DSD and the 16-bit format is for > 5.6MHz DSD. In both cases the driver sets the sample rate to 352.8kHz on > the DAC, I see, this is different from how I've been dealing with it. Since I always use uint8 format, but instead vary the rate to 352.8/705.6/etc... Just because it's much more straightforward form the application's perspective -> output rate is always bit rate divided by eight (=packet rate). And switch between uint8 / uint32 sample format equals to switch between DSD <-> PCM. - Jussi

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[alsa-devel] [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
by Daniel Mack 23 Mar '13

23 Mar '13

Thanks to Andreas, I got my hands on a Playback Design MPD-3 and hacked up some small hanges necessary to make it fully work with snd-usb. The device reports 0x80000000 in the bmFormats field of two of three of its alternate settings, which wrongly made the driver believe it's a usual PCM endpoint (actually due to a fix-up fallback). However, bit 31 of this mask in fact denotes 'raw data'. The effect of this issue is that in addition to the first altsetting with a bit depth of 24, the driver exposed 8-bit and 16-bit native audio formats on the raw data endpoints, which do not work as expected. Also, those devices need a 50ms delay after switching the USB interface. Thanks, Daniel Daniel Mack (3): ALSA: snd-usb: handle the bmFormats field as unsigned int ALSA: snd-usb: handle raw data format of UAC2 devices ALSA: snd-usb: add delay quirk for "Playback Design" products include/linux/usb/audio-v2.h | 2 ++ sound/usb/format.c | 20 ++++++++++++-------- sound/usb/format.h | 2 +- sound/usb/pcm.c | 7 +++++++ sound/usb/stream.c | 2 +- 5 files changed, 23 insertions(+), 10 deletions(-) -- 1.7.10.4

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Re: [alsa-devel] Support for DSD streams
by Daniel Mack 23 Mar '13

23 Mar '13

Hi Andreas, On 23.03.2013 18:35, Andreas Koch wrote: >> Yes, understood. The thing is just that we shouldn't break exising >> solutions that already rely on this "bandaid". > > You wouldn't break any existing solution based on DoP, because that > solution is dealt with in the application on one side and the DAC on > the other side. The driver and USB transfer both think it is PCM data > that is being sent. Well, that was exactly the problem with your device. With the driver as it was, it exposed a PCM interface which is 16-bit wide, and that was picked by ALSA when playing 16-bit material (because ALSA takes whatever is offered by the hardware natively). > The applications have a preference setting to > select DoP and likewise the DAC's constantly monitor the PCM > formatted data stream and automatically switch to DSD if they find > the pattern. Hmm, but your device behaves differently here, right? At least, when normal PCM material is sent to the DSD interface, it doesn't play it or switch back or something. And I think other DACs will have similar behaviour. I need to think about all that more deeply next week :) Daniel

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Re: [alsa-devel] ASoC: tegra: Does the PCM_TRIGGER_{PAUSE, RESUME} emulation really work?
by Lars-Peter Clausen 23 Mar '13

23 Mar '13

On 03/23/2013 02:28 PM, Laxman Dewangan wrote: > On Saturday 23 March 2013 06:59 PM, Lars-Peter Clausen wrote: >> On 03/23/2013 02:16 PM, Laxman Dewangan wrote: >>> On Saturday 23 March 2013 05:05 AM, Stephen Warren wrote: >>>> On 03/22/2013 10:26 AM, Lars-Peter Clausen wrote: >>>>> Hi, >>>>> >>>>> The tegra pcm driver tries to emulate SNDRV_PCM_TRIGGER_PAUSE and >>>>> SNDRV_PCM_TRIGGER_RESUME. By passing SNDRV_PCM_TRIGGER_START and >>>>> SNDRV_PCM_TRIGGER_STOP to snd_dmaengine_pcm_trigger. But does this really >>>>> work? The old non-dmaengine based driver implemented this properly. I'd >>>>> reset the dma position pointer for SNDRV_PCM_TRIGGER_START, but keep it >>>>> where it was for SNDRV_PCM_TRIGGER_RESUME. But as far as I can see with >>>>> the >>>>> dmaengine based driver the pointer always gets reset to 0 for both >>>>> SNDRV_PCM_TRIGGER_START and SNDRV_PCM_TRIGGER_RESUME. >>>> It's quite possible this hasn't been tested, or tested recently. I don't >>>> recall how to test PAUSE/RESUME. Can you enlighten me? >>>> >>>> Laxman, can you comment here? >>> The tegra20/Tegra30 does not support the pause of the channel. It was >>> through global pause. hence the pause and resume is implemented like stop >>> transfer and again restart transfer. hence you are seeing this as the reset >>> pointer as 0. >>> >> Yea, but in that case you shouldn't implement pause/resume at all. You've >> changed the semantics on pause/resume on which applications rely. If >> pause/resume are not implemented applications usually workaround it. >> >> - Lars >> > > So should we return -EINVAL in case of following? > > case SNDRV_PCM_TRIGGER_RESUME: > case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > Don't set the SNDRV_PCM_INFO_PAUSE and SNDRV_PCM_INFO_RESUME bits in the pcm_hardware struct and use snd_dmaengine_pcm_trigger as your trigger callback. The ALSA core will take care of everything else. - Lars

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Re: [alsa-devel] ASoC: tegra: Does the PCM_TRIGGER_{PAUSE, RESUME} emulation really work?
by Lars-Peter Clausen 23 Mar '13

23 Mar '13

On 03/23/2013 02:16 PM, Laxman Dewangan wrote: > On Saturday 23 March 2013 05:05 AM, Stephen Warren wrote: >> On 03/22/2013 10:26 AM, Lars-Peter Clausen wrote: >>> Hi, >>> >>> The tegra pcm driver tries to emulate SNDRV_PCM_TRIGGER_PAUSE and >>> SNDRV_PCM_TRIGGER_RESUME. By passing SNDRV_PCM_TRIGGER_START and >>> SNDRV_PCM_TRIGGER_STOP to snd_dmaengine_pcm_trigger. But does this really >>> work? The old non-dmaengine based driver implemented this properly. I'd >>> reset the dma position pointer for SNDRV_PCM_TRIGGER_START, but keep it >>> where it was for SNDRV_PCM_TRIGGER_RESUME. But as far as I can see with the >>> dmaengine based driver the pointer always gets reset to 0 for both >>> SNDRV_PCM_TRIGGER_START and SNDRV_PCM_TRIGGER_RESUME. >> It's quite possible this hasn't been tested, or tested recently. I don't >> recall how to test PAUSE/RESUME. Can you enlighten me? >> >> Laxman, can you comment here? > The tegra20/Tegra30 does not support the pause of the channel. It was > through global pause. hence the pause and resume is implemented like stop > transfer and again restart transfer. hence you are seeing this as the reset > pointer as 0. > Yea, but in that case you shouldn't implement pause/resume at all. You've changed the semantics on pause/resume on which applications rely. If pause/resume are not implemented applications usually workaround it. - Lars > So pause is stop and resume is the start again. > > If it is require to implement the dmaengine_pause/resume() then it can not > be done truely, it can be done atelast in resolution of one buffer transfer. > We can not stop dma in mid of buffer transfer and start from there. We can > do cancel of current buffer transfer and then start from begining of current > buffer, not from the point it was paused. > > However, tegra114 support this pause and resume truely.

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[alsa-devel] Support for DSD streams (was: Re: [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work)
by Daniel Mack 23 Mar '13

23 Mar '13

Hi Andreas, On 22.03.2013 17:05, Andreas Koch wrote: > At 01:11 AM 3/22/2013, Daniel Mack wrote: >> Hi Jussi, >> Hi Andreas, >> >> On 21.03.2013 22:21, Jussi Laako wrote: >>> On 03/18/2013 03:49 PM, Daniel Mack wrote: >>>> Yes. If at all, we should add a SND_PCM_FORMAT_DSD, and a quirk for that >>>> device. But given that there is no application for DSD in userspace >>>> either, we probably don't need to care. >>> >>> Yes there is? At least my application. And I'd be happy to support any >>> custom format created for the purpose... :) >> >> Ah, interesting. So what *are* you using right now for your transport in >> userspace? Is your application abusing standard PCM interfaces that are >> 'known' to be in fact DSD? How do other DACs expose their interfaces in >> the Linux world, and which ones did you test your application with? >> >> If there's actual need for a new SND_PCM_FORMAT, it would be easy to >> patch it through all the layers down to userspace. >> >>> I would propose to create custom 8-bit sample formats for DSD purposes, >>> similar to the ASIO formats. Practically two main ones, byte with oldest >>> bit in MSB and byte with oldest bit in LSB. And maybe the "DSD-wide" one >>> which is 8-bit DSD sample. >> >> Andreas, which modes does the MPD-3 accept exactly on its two 'raw data' >> interfaces? The only hint I have is that one of it is 8-bit and the >> other is 16-bit, but given the lack of a proper test setup for this >> format, I can't currently test anything. > > The 8-bit raw format is for 2.8MHz DSD and the 16-bit format is for > 5.6MHz DSD. In both cases the driver sets the sample rate to 352.8kHz > on the DAC, but then sends either 8 bits or 16 bits depending on the > DSD rate. And prior to sending either 8 bits or 16 bits, currently the correct USB alternate interface has to be set, right? > Since 11.2MHz DSD is already on the horizon we may have to > change this to a single 8-bit raw interface and then vary the > underlying USB transfer sample rate between 1x, 2x, 4x 352.8kHz - Yes, especially given that if there's no common standard across different devices, either the drivers or the applications have to implement special handling code for this, and know which interface supports what. So for the ALSA layer, we should probably have a generic SND_PCM_FORMAT_DSD and then rely on the sample rate setting mechanisms to select the right DSD mode. Other solutions, like having a different SND_PCM_FORMAT_DSD_{2_8,5_6,11_2}_MHZ would be hairy to implement, as some devices need to get a sample rate update request, while others don't etc ... > that would be more logical and straight forward for the hardware > implementation. This is only supported through my own driver for the > Windows platform which uses the standard ASIO interface that supports > DSD at those rates. Since the Mac OS only supports PCM streams and > nothing else I brought together a bunch of DAC manufacturers and > playback software developers to standardize a method to "cannibalize" > the PCM stream for DSD data. The method is described in the attached > and is now supported by most manufacturers. Very interesting read. For ALSA though, it's no problem to support that in all layers and make the drivers tell applications about it. We just have to decide for a sane way :) > This also has been implemented on a few Linux platforms (Auraliti for > instance). Even though the DoP standard works really well so far, it > is a "bandaid" and should only be used if there is no other choice. Yes, understood. The thing is just that we shouldn't break exising solutions that already rely on this "bandaid". Daniel

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[alsa-devel] ASoC: tegra: Does the PCM_TRIGGER_{PAUSE, RESUME} emulation really work?
by Lars-Peter Clausen 23 Mar '13

23 Mar '13

Hi, The tegra pcm driver tries to emulate SNDRV_PCM_TRIGGER_PAUSE and SNDRV_PCM_TRIGGER_RESUME. By passing SNDRV_PCM_TRIGGER_START and SNDRV_PCM_TRIGGER_STOP to snd_dmaengine_pcm_trigger. But does this really work? The old non-dmaengine based driver implemented this properly. I'd reset the dma position pointer for SNDRV_PCM_TRIGGER_START, but keep it where it was for SNDRV_PCM_TRIGGER_RESUME. But as far as I can see with the dmaengine based driver the pointer always gets reset to 0 for both SNDRV_PCM_TRIGGER_START and SNDRV_PCM_TRIGGER_RESUME. - Lars

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Re: [alsa-devel] [PATCH V2] Sound: sgtl5000 Allow codec clock frequency to be set.
by Timur Tabi 22 Mar '13

22 Mar '13

On Fri, Mar 22, 2013 at 4:33 AM, Martin Fuzzey <mfuzzey(a)parkeon.com> wrote: > diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c > index 424347e..71ce1f6 100644 > --- a/sound/soc/fsl/imx-sgtl5000.c > +++ b/sound/soc/fsl/imx-sgtl5000.c > @@ -128,18 +128,30 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > goto fail; > } > > + ret = of_property_read_u32(codec_np, "clock-frequency", > + &data->clk_frequency); I just noticed that you assign 'ret' here, but you never check the value. So I think you should either: 1) Check the value of 'ret' 2) Remove the "ret = "

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[alsa-devel] [PATCH 0/2] ASoC: dmaengine_pcm: support generic DMA binding users
by Shawn Guo 22 Mar '13

22 Mar '13

The series adds a snd_dmaengine_generic_pcm_open() for users that adopt generic DMA binding and helplers. Vinod, AS per the request from Mark, I add the first patch to mark name parameter of dmaengine helpers as const, so that clients can mark const on their side. We need to have these two patches on a topic branch, as I've heard a few people converting their ASoC driver needing the patches. Can you ack the first patch, so that Mark can maintain them in a branch for anyone who needs it to pull? Shawn Guo (2): dmaengine: add const for name parameter ASoC: dmaengine_pcm: add snd_dmaengine_generic_pcm_open() drivers/dma/dmaengine.c | 2 +- drivers/dma/of-dma.c | 6 +++--- include/linux/dmaengine.h | 7 ++++--- include/linux/of_dma.h | 2 +- include/sound/dmaengine_pcm.h | 2 ++ sound/soc/soc-dmaengine-pcm.c | 39 +++++++++++++++++++++++++++++++++++++++ 6 files changed, 50 insertions(+), 8 deletions(-) -- 1.7.9.5

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Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
by Jussi Laako 22 Mar '13

22 Mar '13

On 03/22/2013 03:27 AM, Andreas Koch wrote: > I wouldn't worry about DSD wide, because that is a dead format that I > was involved in launching for professional recording applications more > than 10 years ago. It never took off. I just need it for my own purposes, but I can make custom stuff as necessary if there's no interest from others. Right now I'm just hoping to see a well defined common sample formats defined for ALSA that can be utilized by drivers. DoP could be implemented inside the driver or at ALSA layer instead of applications doing it. Not that I would mind, but I would find it cleaner than the current approach... :) Currently, ASIO is the only API to support DSD sample formats. I understand that Microsoft or Apple is a slow to turn giant, but I'd hope ALSA wouldn't be such. ;) There are already those IEC958 and DD/dts things being used for movie playback, so this wouldn't be such a big difference afterall. - Jussi

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[alsa-devel] [PATCH] SoC: dma-sh7760: Fix compile error
by Lars-Peter Clausen 22 Mar '13

22 Mar '13

The dma-sh7760 currently fails with the following compile error: sound/soc/sh/dma-sh7760.c:346:2: error: unknown field 'pcm_ops' specified in initializer sound/soc/sh/dma-sh7760.c:346:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c:347:2: error: unknown field 'pcm_new' specified in initializer sound/soc/sh/dma-sh7760.c:347:2: warning: initialization makes integer from pointer without a cast sound/soc/sh/dma-sh7760.c:348:2: error: unknown field 'pcm_free' specified in initializer sound/soc/sh/dma-sh7760.c:348:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c: In function 'sh7760_soc_platform_probe': sound/soc/sh/dma-sh7760.c:353:2: warning: passing argument 2 of 'snd_soc_register_platform' from incompatible pointer type include/sound/soc.h:368:5: note: expected 'struct snd_soc_platform_driver *' but argument is of type 'struct snd_soc_platform *' This is due the misnaming of the snd_soc_platform_driver type name and 'ops' field. The issue was introduced in commit f0fba2a("ASoC: multi-component - ASoC Multi-Component Support"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Given that nobody noticed this over the course of the last three years the code is probably pretty dead by now anyway. --- sound/soc/sh/dma-sh7760.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 19eff8f..1a8b03e 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -342,8 +342,8 @@ static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd) return 0; } -static struct snd_soc_platform sh7760_soc_platform = { - .pcm_ops = &camelot_pcm_ops, +static struct snd_soc_platform_driver sh7760_soc_platform = { + .ops = &camelot_pcm_ops, .pcm_new = camelot_pcm_new, .pcm_free = camelot_pcm_free, }; -- 1.8.0

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[alsa-devel] [PATCH 1/2] ASoC: compress - add support for metadata apis
by Vinod Koul 22 Mar '13

22 Mar '13

From: Jeeja KP <jeeja.kp(a)intel.com> Compress core added metadata apis in 9727b4, so add same in ASoC Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Mark, Takashi: How do you guys wnat to take this one, Ideally this should go to mark's tree but the dependent apis are in takashi's tree, either way is fine with me :) --- sound/soc/soc-compress.c | 27 +++++++++++++++++++++++++++ 1 files changed, 27 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3ea7956..4730789 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -302,11 +302,38 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, return 0; } +static int sst_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) + ret = platform->driver->compr_ops->set_metadata(cstream, metadata); + + return ret; +} + +static int sst_compr_get_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) + ret = platform->driver->compr_ops->get_metadata(cstream, metadata); + + return ret; +} /* ASoC Compress operations */ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, + .set_metadata = sst_compr_set_metadata, + .get_metadata = sst_compr_get_metadata, .get_params = soc_compr_get_params, .trigger = soc_compr_trigger, .pointer = soc_compr_pointer, -- 1.7.0.4

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Re: [alsa-devel] 答复: 答复: how to combine mutil-channel in playback or capture?
by Daniel Mack 22 Mar '13

22 Mar '13

Hi, please don't top-post. On 22.03.2013 11:11, Zhang wei wrote: > But what is the mechanism to synchronize the data from different > channel,and maybe different channel pairs use different DMA buffer. You can try and see if syncing the individual, non-combinded PCM streams helps: http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html#pcm_sync > So, I guess ,about the combine ,according to the alsa.conf,the alsa-lib > opened the 4 devices(in my case) and got every channel paris’s data in > turns in a monent. And then combine them together in order as one frame? It will feed the input to all pcm streams. I don't really know if it uses the sync mechanism on that, but I'd guess so. > Because all the CODEC have the same clock timing,so the different DMA > buffer stored the different channel data for the same moment. Alsa-lib > obtains the same size data from each buffer in turns in once.repeat it > again and again. > > Right? Yes, and so the streams won't drift. Why don't you just go ahead and try it? You should play back some wav file with a test pattern which you can recognize on a DSO or IIS analyzer on the bus. Then see if they are in sync. Daniel

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Re: [alsa-devel] 答复: how to combine mutil-channel in playback or capture?
by Daniel Mack 22 Mar '13

22 Mar '13

On 22.03.2013 10:23, Zhang wei wrote: > Thank you. Yes,you're right. I need 4 different stereo Codecs at > least more than one on my board and i think that will be more funy > than 1 Codec.In this case ,the struct snd_soc_card will be > initialized as following: Struct snd_soc_card test_card= { > .name=”Test Card”, .dai_link=&test_dai_codecs, .num_links=4; } As a > result ,there would be 4 devices named pcmC0Dnp(n=0,1,2,3) under the > directory /dev/snd. I wounder how and who combines the multi-channel > audio data togother? Is it the plugin ? Yes, you can use the alsa userspace for that. See http://www.alsa-project.org/main/index.php/Asoundrc Daniel

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[alsa-devel] how to combine mutil-channel in playback or capture?
by Zhang wei 22 Mar '13

22 Mar '13

I have already achieved the alsa-driver which can playback and capture function with two channel. My board based on ARM9 which have 3 ADC & 1 DAC in IIS bus and all of them shared the same BCLK and LRCK. The pins figure Like this: _________________ | BCLK |___________ | | | LRCK |___________ | | | ADCDAT0 |___________ | | | ADCDAT1 |___________ | | | ADCDAT2 |___________ | | | DACDATA |___________ | | |_______________| Every ADCDATn(n=0,1,2) has two channel so the question is how can I get multi-channel audio data such as 4 or 6 in capture funciton? BR

2 1

[alsa-devel] [PATCH] sound: max98090: Remove executable bit
by Joe Perches 21 Mar '13

21 Mar '13

Source files shouldn't have the executable bit set. Signed-off-by: Joe Perches <joe(a)perches.com> --- include/sound/max98090.h | 0 sound/soc/codecs/max98090.c | 0 sound/soc/codecs/max98090.h | 0 3 files changed, 0 insertions(+), 0 deletions(-) diff --git a/include/sound/max98090.h b/include/sound/max98090.h old mode 100755 new mode 100644 diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c old mode 100755 new mode 100644 diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h old mode 100755 new mode 100644

2 13

Re: [alsa-devel] [PATCH] Sound: sgtl5000 Allow codec clock frequency to be set.
by Timur Tabi 21 Mar '13

21 Mar '13

On 03/21/2013 12:46 PM, Martin Fuzzey wrote: > On 21/03/13 17:11, Timur Tabi wrote: >> On Thu, Mar 21, 2013 at 3:39 AM, Martin Fuzzey <mfuzzey(a)parkeon.com> wrote: >> >>> With this patch and that DT example the frequency of clock 162 will be _set_ >>> to 20MHz >> Does that mean that it ignores the ' clocks' parameter? > No, in this case the clocks parameter must contain the phandle of a clock > whose rate is configurable. > >> If clock-frequency is omitted the binding is still correct (hence the >> optional) but the frequency of clock 162 would not be modified. >> >> In the documentation I wrote "If both a clock and clock-frequency are >> provided the clock's rate will be set. " maybe this is not clear enough? >> It's not clear what it will be set *to*. > It will be set to the frequency specified by the clock-frquency attribute. > > Would > "If both a clock and clock-frequency are provided clock must support > the set_rate operation and its frequency will be set to the value specified > by clock-frequency" be better? Yes, that's much better. > > > The possible configurations and their use cases are: > > 1) only 'clocks' specified > clock points to a clock specified in the DT which already has an > appropriate frequency and is > configured by some other means external to the sgtl5000 driver > (bootloader, board setup code, or just a fixed rate clock) > > 2) only 'clock-frequency' specified > The chip is assumed to be clocked by a signal having the given > frequency, which may even be generated by a clock unknown to linux. > This could actually be represented as a special case of 1) by defining a > fixed-rate clock in the DT. > > 3) Both 'clocks' and 'clock-frequency' specified > The chip is assumed to be clocked by a rate programmable clock defined > in the clock tree. > clk_set_rate() will be called for this clock to set its rate to that > specified by clock-frequency This should all be in the binding document. > Cases 1 and 2 exist in the current code, this patch adds support for case 3. > > Prior to commit 81e8e4926167ab32593bbb915b45a42024ca1020 "ASoC: fsl: add > sgtl5000 clock support for imx-sgtl5000" > only case 2 was supported > This explains why 2 is not implemented as a special case of 1) > plain 'clock-frequency' was supported before the driver learnt how to > get a clock from the DT And this should be in the patch description. -- Timur Tabi

1 0

Re: [alsa-devel] [PATCH 50/54 v2] ASoC: switch over to use snd_soc_register_component() on tegra20 ac97
by Stephen Warren 21 Mar '13

21 Mar '13

On 03/21/2013 11:19 AM, Lucas Stach wrote: > Am Donnerstag, den 21.03.2013, 11:03 -0600 schrieb Stephen Warren: >> On 03/21/2013 04:37 AM, Kuninori Morimoto wrote: >> >> Lucas, can you please test this to make sure that the change from >> snd_soc_register_dais() to snd_soc_register_dai() (the latter inside >> snd_soc_register_component()) coupled with the change to the >> tegra_wm9712.c machine driver yields the correct results. Thanks. > > Can someone please FWD me the whole patch? I'm not subscribed to > ALSA-devel and wasn't on CC for this patch. I'll do that off-list.

1 0

[alsa-devel] [PATCH] ALSA: hda - Introduce "Headset Mic" name
by David Henningsson 21 Mar '13

21 Mar '13

Headset mic jacks, i e TRRS style jacks with Headphone Left, Headphone Right, Mic and GND signals, are becoming increasingly common and are now being shipped by several manufacturers. Unfortunately, the HDA specification does not give us any hint of whether a Mic pin belongs to such a jack or not, but it would still be helpful for the user to know (especially if there is one TRS Mic jack and one TRRS headset jack). This new fixup causes the first (non-dock, non-internal) mic to be a headset mic jack. The algorithm can be later refined if needed. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 24 ++++++++++++++++++++++-- sound/pci/hda/hda_auto_parser.h | 2 ++ sound/pci/hda/patch_sigmatel.c | 6 +++++- 3 files changed, 29 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index a3ea76a..6b173b3 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -260,6 +260,22 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, } } + /* Take first mic to be a headset mic pin */ + if (cond_flags & HDA_PINCFG_HEADSET_MIC) { + for (i = 0; i < cfg->num_inputs; i++) { + int attr; + unsigned int def_conf; + if (cfg->inputs[i].type != AUTO_PIN_MIC) + continue; + def_conf = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin); + attr = snd_hda_get_input_pin_attr(def_conf); + if (attr <= INPUT_PIN_ATTR_DOCK) + continue; + cfg->inputs[i].is_headset_mic = 1; + break; + } + } + /* FIX-UP: * If no line-out is defined but multiple HPs are found, * some of them might be the real line-outs. @@ -388,6 +404,7 @@ EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr); */ static const char *hda_get_input_pin_label(struct hda_codec *codec, + const struct auto_pin_cfg_item *item, hda_nid_t pin, bool check_location) { unsigned int def_conf; @@ -400,6 +417,8 @@ static const char *hda_get_input_pin_label(struct hda_codec *codec, switch (get_defcfg_device(def_conf)) { case AC_JACK_MIC_IN: + if (item && item->is_headset_mic) + return "Headset Mic"; if (!check_location) return "Mic"; attr = snd_hda_get_input_pin_attr(def_conf); @@ -480,7 +499,8 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec, has_multiple_pins = 1; if (has_multiple_pins && type == AUTO_PIN_MIC) has_multiple_pins &= check_mic_location_need(codec, cfg, input); - return hda_get_input_pin_label(codec, cfg->inputs[input].pin, + return hda_get_input_pin_label(codec, &cfg->inputs[input], + cfg->inputs[input].pin, has_multiple_pins); } EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); @@ -649,7 +669,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, } } if (!name) - name = hda_get_input_pin_label(codec, nid, true); + name = hda_get_input_pin_label(codec, NULL, nid, true); break; } if (!name) diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h index f748071..98a72d0 100644 --- a/sound/pci/hda/hda_auto_parser.h +++ b/sound/pci/hda/hda_auto_parser.h @@ -36,6 +36,7 @@ enum { struct auto_pin_cfg_item { hda_nid_t pin; int type; + int is_headset_mic:1; }; struct auto_pin_cfg; @@ -80,6 +81,7 @@ struct auto_pin_cfg { /* bit-flags for snd_hda_parse_pin_def_config() behavior */ #define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */ #define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */ +#define HDA_PINCFG_HEADSET_MIC (1 << 2) /* Take first mic as headset mic */ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, struct auto_pin_cfg *cfg, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3be877b..1d9d642 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3528,8 +3528,12 @@ static int stac_parse_auto_config(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; int err; + int flags = 0; - err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); + if (spec->headset_jack) + flags |= HDA_PINCFG_HEADSET_MIC; + + err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, flags); if (err < 0) return err; -- 1.7.9.5

2 7

Re: [alsa-devel] [PATCH] Sound: sgtl5000 Allow codec clock frequency to be set.
by Timur Tabi 21 Mar '13

21 Mar '13

On Thu, Mar 21, 2013 at 3:39 AM, Martin Fuzzey <mfuzzey(a)parkeon.com> wrote: > With this patch and that DT example the frequency of clock 162 will be _set_ > to 20MHz Does that mean that it ignores the ' clocks' parameter? > If clock-frequency is omitted the binding is still correct (hence the > optional) but the frequency of clock 162 would not be modified. > > In the documentation I wrote "If both a clock and clock-frequency are > provided the clock's rate will be set. " maybe this is not clear enough? It's not clear what it will be set *to*.

1 0

[alsa-devel] [PATCH v2 1/2] ALSA: hda - Introduce "Headset Mic" name
by David Henningsson 21 Mar '13

21 Mar '13

Headset mic jacks, i e TRRS style jacks with Headphone Left, Headphone Right, Mic and GND signals, are becoming increasingly common and are now being shipped by several manufacturers. Unfortunately, the HDA specification does not give us any hint of whether a Mic pin belongs to such a jack or not, but it would still be helpful for the user to know (especially if there is one TRS Mic jack and one TRRS headset jack). This new fixup causes the first (non-dock, non-internal) mic to be a headset mic jack. The algorithm can be later refined if needed. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 24 ++++++++++++++++++++++-- sound/pci/hda/hda_auto_parser.h | 2 ++ 2 files changed, 24 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index a3ea76a..6b173b3 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -260,6 +260,22 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, } } + /* Take first mic to be a headset mic pin */ + if (cond_flags & HDA_PINCFG_HEADSET_MIC) { + for (i = 0; i < cfg->num_inputs; i++) { + int attr; + unsigned int def_conf; + if (cfg->inputs[i].type != AUTO_PIN_MIC) + continue; + def_conf = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin); + attr = snd_hda_get_input_pin_attr(def_conf); + if (attr <= INPUT_PIN_ATTR_DOCK) + continue; + cfg->inputs[i].is_headset_mic = 1; + break; + } + } + /* FIX-UP: * If no line-out is defined but multiple HPs are found, * some of them might be the real line-outs. @@ -388,6 +404,7 @@ EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr); */ static const char *hda_get_input_pin_label(struct hda_codec *codec, + const struct auto_pin_cfg_item *item, hda_nid_t pin, bool check_location) { unsigned int def_conf; @@ -400,6 +417,8 @@ static const char *hda_get_input_pin_label(struct hda_codec *codec, switch (get_defcfg_device(def_conf)) { case AC_JACK_MIC_IN: + if (item && item->is_headset_mic) + return "Headset Mic"; if (!check_location) return "Mic"; attr = snd_hda_get_input_pin_attr(def_conf); @@ -480,7 +499,8 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec, has_multiple_pins = 1; if (has_multiple_pins && type == AUTO_PIN_MIC) has_multiple_pins &= check_mic_location_need(codec, cfg, input); - return hda_get_input_pin_label(codec, cfg->inputs[input].pin, + return hda_get_input_pin_label(codec, &cfg->inputs[input], + cfg->inputs[input].pin, has_multiple_pins); } EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); @@ -649,7 +669,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, } } if (!name) - name = hda_get_input_pin_label(codec, nid, true); + name = hda_get_input_pin_label(codec, NULL, nid, true); break; } if (!name) diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h index f748071..c7826ce 100644 --- a/sound/pci/hda/hda_auto_parser.h +++ b/sound/pci/hda/hda_auto_parser.h @@ -36,6 +36,7 @@ enum { struct auto_pin_cfg_item { hda_nid_t pin; int type; + unsigned int is_headset_mic:1; }; struct auto_pin_cfg; @@ -80,6 +81,7 @@ struct auto_pin_cfg { /* bit-flags for snd_hda_parse_pin_def_config() behavior */ #define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */ #define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */ +#define HDA_PINCFG_HEADSET_MIC (1 << 2) /* Take first mic as headset mic */ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, struct auto_pin_cfg *cfg, -- 1.7.9.5

1 1

Re: [alsa-devel] [PATCH] Sound: sgtl5000 Allow codec clock frequency to be set.
by Timur Tabi 21 Mar '13

21 Mar '13

Martin Fuzzey wrote: > +If a clock is provided, clock-frequency is optional > + > +If no clock is provided clock-frequency is required (this represents the codec > +being clocked by an external signal not present in the clock tree) > + > +If both a clock and clock-frequency are provided the clock's rate will be set. > + > + > Example: > > codec: sgtl5000@0a { > compatible = "fsl,sgtl5000"; > reg = <0x0a>; > + clock-frequency = <20000000>; > + clocks = <&clks 162>; /* cko1 */ The code looks okay, but I'm not sure this example is right. The binding says that clock-frequency is optional if 'clocks' is provided, but here you list both. You should probably have two examples. -- Timur Tabi

1 0

Re: [alsa-devel] [PATCH 1/2] ASoC: wm8903: Add the DAC boost control
by Mark Brown 20 Mar '13

20 Mar '13

On Wed, Mar 20, 2013 at 05:37:32PM +0100, Alban Bedel wrote: > Signed-off-by: Alban Bedel <alban.bedel(a)avionic-design.de> Applied, thanks.

1 0

[alsa-devel] [PATCH] salsa-lib: Rename _sndrv_pcm_format to _snd_pcm_format
by Mihail Zenkov 20 Mar '13

20 Mar '13

Rename _sndrv_pcm_format to _snd_pcm_format for compatible with alsa-lib. --- salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884/src/asound.h +++ salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884-1/src/asound.h @@ -133,7 +133,7 @@ SND_PCM_ACCESS_LAST = SND_PCM_ACCESS_RW_NONINTERLEAVED, } snd_pcm_access_t; -typedef enum _sndrv_pcm_format { +typedef enum _snd_pcm_format { SND_PCM_FORMAT_UNKNOWN = -1, SND_PCM_FORMAT_S8 = 0, /* signed */ SND_PCM_FORMAT_U8, /* unsigned */

2 1

[alsa-devel] [PATCH] salsa-lib: Set SNDRV_PCM_HW_PARAMS_NORESAMPLE by default
by Mihail Zenkov 20 Mar '13

20 Mar '13

As we don't allow resampling set SNDRV_PCM_HW_PARAMS_NORESAMPLE by default to respect snd_pcm_hw_rule_noresample in drivers. --- salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884/src/pcm_params.c +++ salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884-1/src/pcm_params.c @@ -436,6 +436,7 @@ params->rmask = ~0U; params->cmask = 0; params->info = ~0U; + params->flags = SNDRV_PCM_HW_PARAMS_NORESAMPLE; } int snd_pcm_hw_params_any(snd_pcm_t *pcm, snd_pcm_hw_params_t *params)

2 1

[alsa-devel] [PATCH] salsa-lib: Fix mistakes after refactoring pcm_params.c
by Mihail Zenkov 20 Mar '13

20 Mar '13

Fix some regressions after refactoring pcm_params.c These mistakes don't allow snd_pcm_hw_param_set_near to establish correct value. --- salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884/src/pcm_params.c +++ salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884-1/src/pcm_params.c @@ -590,11 +590,9 @@ save = *params; err = hw_param_set_max(params, var, *val, dir ? *dir : 0); - if (!err) { - err = hw_param_update_var(pcm, params, var, err); - if (!err) - return _snd_pcm_hw_param_get_max(params, var, val, dir); - } + err = hw_param_update_var(pcm, params, var, err); + if (!err) + return _snd_pcm_hw_param_get_max(params, var, val, dir); *params = save; return err; } @@ -683,7 +681,7 @@ snd_pcm_hw_params_t save = *params; int err = hw_param_set(params, var, val, dir); err = hw_param_update_var(pcm, params, var, err); - if (err) + if (err < 0) *params = save; return err; } @@ -712,7 +710,7 @@ save = *params; i->integer = 1; err = hw_param_update_var(pcm, params, var, 1); - if (err) + if (err < 0) *params = save; return err; } @@ -827,7 +825,7 @@ else if (valdir == 0) maxdir = -1; else { - valdir = 1; + maxdir = 1; max--; } save = *params;

2 1

Re: [alsa-devel] [PATCH] sound: soc: fix invalid free of devm_ allocated data
by Mark Brown 20 Mar '13

20 Mar '13

On Sat, Mar 16, 2013 at 01:45:34PM +0200, Silviu-Mihai Popescu wrote: > The objects allocated by devm_* APIs are managed by devres and are freed when > the device is detached. Hence there is no need to use kfree() explicitly. ...but I applied this anyway since it's clearly an improvement and a small leak is better than a crash.

1 0

Re: [alsa-devel] [PATCH] sound: soc: fix invalid free of devm_ allocated data
by Mark Brown 20 Mar '13

20 Mar '13

On Sat, Mar 16, 2013 at 01:45:34PM +0200, Silviu-Mihai Popescu wrote: > The objects allocated by devm_* APIs are managed by devres and are freed when > the device is detached. Hence there is no need to use kfree() explicitly. In this case since we're in error handling paths which might not be fatal (though I suspect they probably should be normally) it looks like the better fix is to use devm_kfree() to free immediately. Otherwise the unused memory will end up hanging around for as long as the device is bound.

1 0

[alsa-devel] [PATCH 1/2] ASoC: spear_pcm: Update to new pcm_new() API
by Lars-Peter Clausen 20 Mar '13

20 Mar '13

Commit 552d1ef6 ("ASoC: core - Optimise and refactor pcm_new() to pass only rtd") updated the pcm_new() callback to take the rtd as the only parameter. The spear PCM driver (which was merged much later) still uses the old API. This patch updates the driver to the new API. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Btw. where are the Kconfig and Makefile files for spear? --- sound/soc/spear/spear_pcm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 9b76cc5..5e7aebe 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -149,9 +149,9 @@ static void spear_pcm_free(struct snd_pcm *pcm) static u64 spear_pcm_dmamask = DMA_BIT_MASK(32); -static int spear_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; int ret; if (!card->dev->dma_mask) @@ -159,16 +159,16 @@ static int spear_pcm_new(struct snd_card *card, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->playback.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, spear_pcm_hardware.buffer_bytes_max); if (ret) return ret; } - if (dai->driver->capture.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->capture.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_CAPTURE, spear_pcm_hardware.buffer_bytes_max); if (ret) -- 1.8.0

3 3

[alsa-devel] [PATCH 0/3] ASoC: omap-mcpdm: Fix for full duplex audio
by Peter Ujfalusi 20 Mar '13

20 Mar '13

Hi, We had this issue for a long time but I had no time to think and fix it. With this series the full duplex audio will work correctly with McPDM. Under the hood we have HW limitation which prevents us to start up/downlink separately. Regards, Peter --- Peter Ujfalusi (3): ASoC: omap-mcpdm: Collect link direction configuration under a struct ASoC: omap-mcpdm: Fix for full duplex audio use case ASoC: omap-mcpdm: Remove leftower define for IO address sound/soc/omap/omap-mcpdm.c | 66 ++++++++++++++++++++++++++++++++------------- 1 file changed, 48 insertions(+), 18 deletions(-) -- 1.8.1.5

2 4

[alsa-devel] [PATCH] ASoC: fsl: imx-pcm-fiq: Use 'unsigned int' for period
by Fabio Estevam 20 Mar '13

20 Mar '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Fix the following warning when building with W=1 option: sound/soc/fsl/imx-pcm-fiq.c: In function 'snd_hrtimer_callback': sound/soc/fsl/imx-pcm-fiq.c:76:12: warning: comparison between signed and unsigned integer expressions [-Wsign-compare] Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-pcm-fiq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 920f945..47228c0 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -34,7 +34,7 @@ #include "imx-ssi.h" struct imx_pcm_runtime_data { - int period; + unsigned int period; int periods; unsigned long offset; unsigned long last_offset; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: snd-usb: mixer: propagate errors up the call chain
by Daniel Mack 20 Mar '13

20 Mar '13

In check_input_term() and parse_audio_feature_unit(), propagate the error value that has been returned by a failing function instead of -EINVAL. That helps cleaning up the error pathes in the mixer. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/mixer.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 638e7f7..db30d02 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -715,8 +715,9 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ case UAC2_CLOCK_SELECTOR: { struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ - if (check_input_term(state, d->baSourceID[0], term) < 0) - return -ENODEV; + err = check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; term->type = d->bDescriptorSubtype << 16; /* virtual type */ term->id = id; term->name = uac_selector_unit_iSelector(d); @@ -1356,8 +1357,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void return err; /* determine the input source type and name */ - if (check_input_term(state, hdr->bSourceID, &iterm) < 0) - return -EINVAL; + err = check_input_term(state, hdr->bSourceID, &iterm); + if (err < 0) + return err; master_bits = snd_usb_combine_bytes(bmaControls, csize); /* master configuration quirks */ -- 1.8.1.4

2 2

[alsa-devel] [PATCH] ALSA: usb: Parse UAC2 extension unit like for UAC1
by Torstein Hegge 20 Mar '13

20 Mar '13

UAC2_EXTENSION_UNIT_V2 differs from UAC1_EXTENSION_UNIT, but can be handled in the same way when parsing the unit. Otherwise parse_audio_unit() fails when it sees an extension unit on a UAC2 device. UAC2_EXTENSION_UNIT_V2 is outside the range allocated by UAC1. Signed-off-by: Torstein Hegge <hegge(a)resisty.net> --- sound/usb/mixer.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 638e7f7..8eb84c0 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -725,7 +725,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ case UAC1_PROCESSING_UNIT: case UAC1_EXTENSION_UNIT: /* UAC2_PROCESSING_UNIT_V2 */ - /* UAC2_EFFECT_UNIT */ { + /* UAC2_EFFECT_UNIT */ + case UAC2_EXTENSION_UNIT_V2: { struct uac_processing_unit_descriptor *d = p1; if (state->mixer->protocol == UAC_VERSION_2 && @@ -2052,6 +2053,8 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_extension_unit(state, unitid, p1); else /* UAC_VERSION_2 */ return parse_audio_processing_unit(state, unitid, p1); + case UAC2_EXTENSION_UNIT_V2: + return parse_audio_extension_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); return -EINVAL; -- 1.7.10.4

3 6

[alsa-devel] [TINYCOMPRESS] Remove local copies of kernel headers?
by Richard Fitzgerald 20 Mar '13

20 Mar '13

I'm assuming that the local copies of compress_offload.h and compress_params.h were there as a convenience during early development Can we remove these local copies now?

3 2

[alsa-devel] ASoC: atmel-pcm-dma: Leaking memory in atmel_pcm_hw_params
by Lars-Peter Clausen 19 Mar '13

19 Mar '13

Hi, The dmaengine based pcm driver for atmel calls snd_dmaengine_pcm_open from it's hw_params callback. There is nothing preventing an application calling hw_params more than once. snd_dmaengine_pcm_open allocates a new prtd struct each time it gets called. So in case hw_params is called multiple times we leak memory here. Is there any specific reason why snd_dmaengine_pcm_open() needs to be called from the hw_params callback and why it can't be called from the open callback? The filter function depends on the data returned by snd_soc_dai_get_dma_data(), but as far as I can see the DAI driver calls snd_soc_dai_set_dma_data() from its startup() callback, so the data is available in the PCM drivers open callback. - Lars

2 3

[alsa-devel] [PATCH 1/2] snd-usb-audio: Playback and MIDI support for Novation Twitch DJ controller
by Mark Hills 18 Mar '13

18 Mar '13

The hardware also has a PCM capture device which is not implemented in this patch. It may be possible to generalise this to Saffire 6 USB support and some of the other Focusrite interfaces, but as I don't have access to these devices we should wait until capture support is working first. Capture support is not implemented because the code assumes the endpoint to have its own interface (instead, it shares the interface with playback) and some thought will be needed to lift this limitation. --- sound/usb/quirks-table.h | 40 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 15 +++++++++++++++ 2 files changed, 55 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 820580a..1c8db85 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2677,6 +2677,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE(0x1235, 0x0018), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Novation", + .product_name = "Twitch", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 1, + .type = QUIRK_MIDI_RAW_BYTES + }, + { + .ifnum = -1 + } + } + } +}, +{ USB_DEVICE_VENDOR_SPEC(0x1235, 0x4661), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Novation", diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0115289..b0f7140 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -446,6 +446,17 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) } /* + * Novation Twitch DJ controller + */ +static int snd_usb_twitch_boot_quirk(struct usb_device *dev) +{ + /* preemptively set up the device because otherwise the + * raw MIDI endpoints are not active */ + usb_set_interface(dev, 0, 1); + return 0; +} + +/* * This call will put the synth in "USB send" mode, i.e it will send MIDI * messages through USB (this is disabled at startup). The synth will * acknowledge by sending a sysex on endpoint 0x85 and by displaying a USB @@ -746,6 +757,10 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); + case USB_ID(0x1235, 0x0018): + /* Focusrite Novation Twitch */ + return snd_usb_twitch_boot_quirk(dev); + case USB_ID(0x133e, 0x0815): /* Access Music VirusTI Desktop */ return snd_usb_accessmusic_boot_quirk(dev); -- 1.7.12.1

2 4

Re: [alsa-devel] [PATCH] documentation: alsa: Fix typo in Documentation/sound
by Takashi Iwai 17 Mar '13

17 Mar '13

At Sun, 17 Mar 2013 02:57:28 +0900, Masanari Iida wrote: > > Correct spelling typos in Documentation/sound/alsa > > Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> Thanks, applied. Takashi > --- > Documentation/sound/alsa/ALSA-Configuration.txt | 2 +- > Documentation/sound/alsa/seq_oss.html | 2 +- > 2 files changed, 2 insertions(+), 2 deletions(-) > > diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt > index ce6581c..4499bd9 100644 > --- a/Documentation/sound/alsa/ALSA-Configuration.txt > +++ b/Documentation/sound/alsa/ALSA-Configuration.txt > @@ -912,7 +912,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. > models depending on the codec chip. The list of available models > is found in HD-Audio-Models.txt > > - The model name "genric" is treated as a special case. When this > + The model name "generic" is treated as a special case. When this > model is given, the driver uses the generic codec parser without > "codec-patch". It's sometimes good for testing and debugging. > > diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html > index d9776cf..9663b45 100644 > --- a/Documentation/sound/alsa/seq_oss.html > +++ b/Documentation/sound/alsa/seq_oss.html > @@ -285,7 +285,7 @@ sample data. > <H4> > 7.2.4 Close Callback</H4> > The <TT>close</TT> callback is called when this device is closed by the > -applicaion. If any private data was allocated in open callback, it must > +application. If any private data was allocated in open callback, it must > be released in the close callback. The deletion of ALSA port should be > done here, too. This callback must not be NULL. > <H4> > -- > 1.8.2 >

1 0

[alsa-devel] [PATCH] ASoC: core: Add support for platform and CODEC drivers on same device
by Charles Keepax 15 Mar '13

15 Mar '13

Currently DAI playback and capture widgets are created during soc_probe_codec and soc_probe_platform, using the device associated with the DAI to check which widgets should be created. If a device registers both a CODEC and platform driver this leads the CODEC playback and capture widgets being overwritten by the widgets created by the platform probe. It is more sensible to retain the CODEC widgets as the most common use case for registering both a CODEC and platform driver on the same chip is a CODEC which contains a DSP for compressed playback. In this situation it is more sensible to attach the routing information to the CODEC and add a thin platform driver interface to link into the compressed API. So this patch will check for existing widgets during soc_probe_platform and only create new widgets if no existing ones exist. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 04af2a6..53efe1d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1159,7 +1159,8 @@ static int soc_probe_platform(struct snd_soc_card *card, /* Create DAPM widgets for each DAI stream */ list_for_each_entry(dai, &dai_list, list) { - if (dai->dev != platform->dev) + if (dai->dev != platform->dev || + dai->playback_widget || dai->capture_widget) continue; snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); -- 1.7.2.5

2 6

[alsa-devel] [PATCH V2 0/2] ASoC: OMAP2+: Update Audio IP with sDMA binding for DT boot
by Sebastien Guiriec 15 Mar '13

15 Mar '13

This patch serie is converting OMAP Audio IPs in order to use OMAP sDMA DT binding in case of DT boot on OMAP2+ platforms. It is depending on the next patches: [1] http://www.spinics.net/lists/arm-kernel/msg227836.html [2] http://www.mail-archive.com/linux-omap@vger.kernel.org/msg84668.html It has been tested on: - OMAP5 (sEVM/panda)in DT boot (McPDM/DMIC). - OMAP4 (SDP4430/Panda) in both DT/non DT boot (McPDM/DMIC) - OMAP3 (Beagle XM) in both DT/non DT boot (McBSP) Update since v1: - Put DTS patch in first. - Squash all OMAP IPs patch together. Sebastien Guiriec (2): ARM: dts: OMAP2+: Add SDMA Audio IPs bindings ASoC: OMAP2+: Move McBSP, DMIC and McPDM to generic DMA DT binding arch/arm/boot/dts/omap2420.dtsi | 6 ++++++ arch/arm/boot/dts/omap2430.dtsi | 15 +++++++++++++++ arch/arm/boot/dts/omap3.dtsi | 15 +++++++++++++++ arch/arm/boot/dts/omap4.dtsi | 17 +++++++++++++++++ arch/arm/boot/dts/omap5.dtsi | 14 ++++++++++++++ sound/soc/omap/mcbsp.c | 26 ++++++++++++++++---------- sound/soc/omap/omap-dmic.c | 15 +++++++++------ sound/soc/omap/omap-mcpdm.c | 22 ++++++++++++++-------- sound/soc/omap/omap-pcm.c | 12 ++++++++++-- sound/soc/omap/omap-pcm.h | 1 + 10 files changed, 117 insertions(+), 26 deletions(-) -- 1.7.10.4

6 16

[alsa-devel] [PATCH] ALSA: at73c213: Use dev_pm_ops
by Lars-Peter Clausen 15 Mar '13

15 Mar '13

Use dev_pm_ops instead of the deprecated legacy suspend/resume callbacks. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/spi/at73c213.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 4dd60d8..a1a24b9 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -1075,10 +1075,11 @@ out: return 0; } -#ifdef CONFIG_PM -static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) +#ifdef CONFIG_PM_SLEEP + +static int snd_at73c213_suspend(struct device *dev) { - struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); @@ -1087,9 +1088,9 @@ static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) return 0; } -static int snd_at73c213_resume(struct spi_device *spi) +static int snd_at73c213_resume(struct device *dev) { - struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; clk_enable(chip->board->dac_clk); @@ -1097,18 +1098,21 @@ static int snd_at73c213_resume(struct spi_device *spi) return 0; } + +static SIMPLE_DEV_PM_OPS(at73c213_pm_ops, snd_at73c213_suspend, + snd_at73c213_resume); +#define AT73C213_PM_OPS (&at73c213_pm_ops) + #else -#define snd_at73c213_suspend NULL -#define snd_at73c213_resume NULL +#define AT73C213_PM_OPS NULL #endif static struct spi_driver at73c213_driver = { .driver = { .name = "at73c213", + .pm = AT73C213_PM_OPS, }, .probe = snd_at73c213_probe, - .suspend = snd_at73c213_suspend, - .resume = snd_at73c213_resume, .remove = snd_at73c213_remove, }; -- 1.8.0

1 0

Re: [alsa-devel] [PATCH 08/10] ASoC: dmaengine_pcm: Add open function for DT DMA request
by Shawn Guo 15 Mar '13

15 Mar '13

On Thu, Mar 14, 2013 at 02:08:32PM +0100, Markus Pargmann wrote: > On Wed, Mar 13, 2013 at 10:18:29AM +0800, Shawn Guo wrote: > > On Tue, Mar 12, 2013 at 07:02:07PM +0000, Mark Brown wrote: > > > On Sun, Mar 10, 2013 at 07:33:09PM +0100, Markus Pargmann wrote: > > > > Add a function to open a DMA PCM substream using devicetree data > > > > provided via the client device node. The patch introduces a public > > > > function and a private subfunction that is called by both open > > > > functions. > > > > > > Someone (I think it was Shawn) sent a very similar patch just recently > > > which appears to have fallen out of my inbox unfortunately - can you > > > please check what's going on there and coordinate with them? Let me > > > know if you can't find the patch and I'll dig it out. > > > > It's here, Markus. > > > > http://thread.gmane.org/gmane.linux.alsa.devel/106027/focus=106031 > > > > Thanks. Do you already have a patch for the non-generic function which I > could use instead? > I do not follow. With my patch in place, there will be two dmaengine_pcm APIs. int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) int snd_dmaengine_generic_pcm_open(struct snd_pcm_substream *substream, struct device *dev, const char *name) The first one is for users that their dmaengine driver hasn't converted to generic device tree bindings, and the second one is for users that their dmaengine driver is converted to generic device tree bindings. Shawn

2 1

[alsa-devel] [PATCH 0/3] ALSA: hda/ca0132 - Better handle a failed DSP firmware load.
by Dylan Reid 15 Mar '13

15 Mar '13

Enabling CONFIG_SND_HDA_CODEC_CA0132_DSP on the Chromebook Pixel leads to 10+ a second delay between running aplay and hearing audio. This was caused by the driver thinking the DSP was loaded when it had actually failed. The first two patches avoid that situation, the DSP still fails to load every time, but at least the system continues without a painful delay. The third patch is just a cleanup that touches the same state variable. Dylan Reid (3): ALSA: hda/ca0132 - Check if dspload_image succeeded. ALSA: hda/ca0132 - Check download state of DSP. ALSA: hda/ca0132 - Remove extra setting of dsp_state. sound/pci/hda/patch_ca0132.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) -- 1.8.1.3.605.g02339dd

2 4

[alsa-devel] [patch] sound: sequencer: cap array index in seq_chn_common_event()
by Dan Carpenter 15 Mar '13

15 Mar '13

"chn" here is a number between 0 and 255, but ->chn_info[] only has 16 elements so there is a potential write beyond the end of the array. If the seq_mode isn't SEQ_2 then we let the individual drivers (either opl3.c or midi_synth.c) handle it. Those functions all do a bounds check on "chn" so I haven't changed anything here. The opl3.c driver has up to 18 channels and not 16. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Static checker stuff. diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index 30bcfe4..4ff60a6 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -545,6 +545,9 @@ static void seq_chn_common_event(unsigned char *event_rec) case MIDI_PGM_CHANGE: if (seq_mode == SEQ_2) { + if (chn > 15) + break; + synth_devs[dev]->chn_info[chn].pgm_num = p1; if ((int) dev >= num_synths) synth_devs[dev]->set_instr(dev, chn, p1); @@ -596,6 +599,9 @@ static void seq_chn_common_event(unsigned char *event_rec) case MIDI_PITCH_BEND: if (seq_mode == SEQ_2) { + if (chn > 15) + break; + synth_devs[dev]->chn_info[chn].bender_value = w14; if ((int) dev < num_synths)

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs4271: switch to mute_stream
by Daniel Mack 15 Mar '13

15 Mar '13

Use the newly introduced mute_stream DAI operation, and don't mute the codec if it's called for the _CAPTURE stream. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/cs4271.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ac0d3b4..03036b3 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -388,7 +388,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, return cs4271_set_deemph(codec); } -static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) +static int cs4271_mute_stream(struct snd_soc_dai *dai, int mute, int stream) { struct snd_soc_codec *codec = dai->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); @@ -396,6 +396,9 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) int val_a = 0; int val_b = 0; + if (stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + if (mute) { val_a = CS4271_VOLA_MUTE; val_b = CS4271_VOLB_MUTE; @@ -442,7 +445,7 @@ static const struct snd_soc_dai_ops cs4271_dai_ops = { .hw_params = cs4271_hw_params, .set_sysclk = cs4271_set_dai_sysclk, .set_fmt = cs4271_set_dai_fmt, - .digital_mute = cs4271_digital_mute, + .mute_stream = cs4271_mute_stream, }; static struct snd_soc_dai_driver cs4271_dai = { -- 1.8.1.4

1 1

[alsa-devel] [PATCH] ASoC: dapm: Fix pointer dereference in is_connected_output_ep()
by Peter Ujfalusi 15 Mar '13

15 Mar '13

*path is not yet initialized when we check if the widget is connected. The compiler also warns about this: sound/soc/soc-dapm.c: In function 'is_connected_output_ep': sound/soc/soc-dapm.c:824:18: warning: 'path' may be used uninitialized in this function Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Liam, This patch is for the for-next branch. I got NULL pointer dereference because of this depending on the used toolchain. Regards, Peter sound/soc/soc-dapm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index eaf9b74..33acd8b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -821,7 +821,6 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources))) { widget->outputs = snd_soc_dapm_suspend_check(widget); - path->walking = 0; return widget->outputs; } } -- 1.8.1.5

2 1

[alsa-devel] [PATCH 0/2] ASoC: add new struct snd_soc_chip
by Kuninori Morimoto 15 Mar '13

15 Mar '13

Hi Mark, Liam, Stephen, Lars In my understanding on ASoC, - We will clean-up ASoC code. - CODEC/CPU will be common struct (platform too ??) As 1st trigger for this common struct, I created very basic struct snd_soc_chip. I hope it is accepted, and updated to be more useful. Kuninori Morimoto (2): ASoC: add snd_soc_register_chip() ASoC: fsi: use snd_soc_register_chip() instead of snd_soc_register_dais() include/sound/soc.h | 19 +++++++++++++ sound/soc/sh/fsi.c | 11 +++++--- sound/soc/soc-core.c | 77 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 103 insertions(+), 4 deletions(-)

4 18

[alsa-devel] [PATCH v3] ALSA: ASoC: cs4271: convert to direct regmap API usage
by Daniel Mack 15 Mar '13

15 Mar '13

By using the regmap API directly, we can make use of the .write_flag_mask for SPI, which allows us to drop the strange register hacks that were necessary so far. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- v3: dropped one more (now misleading) comment v2: added a .write_flag_mask sound/soc/codecs/cs4271.c | 159 ++++++++++++++++++++++++++-------------------- 1 file changed, 91 insertions(+), 68 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 2415a41..ac0d3b4 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -39,17 +39,15 @@ /* * CS4271 registers - * High byte represents SPI chip address (0x10) + write command (0) - * Low byte - codec register address */ -#define CS4271_MODE1 0x2001 /* Mode Control 1 */ -#define CS4271_DACCTL 0x2002 /* DAC Control */ -#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */ -#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */ -#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */ -#define CS4271_ADCCTL 0x2006 /* ADC Control */ -#define CS4271_MODE2 0x2007 /* Mode Control 2 */ -#define CS4271_CHIPID 0x2008 /* Chip ID */ +#define CS4271_MODE1 0x01 /* Mode Control 1 */ +#define CS4271_DACCTL 0x02 /* DAC Control */ +#define CS4271_DACVOL 0x03 /* DAC Volume & Mixing Control */ +#define CS4271_VOLA 0x04 /* DAC Channel A Volume Control */ +#define CS4271_VOLB 0x05 /* DAC Channel B Volume Control */ +#define CS4271_ADCCTL 0x06 /* ADC Control */ +#define CS4271_MODE2 0x07 /* Mode Control 2 */ +#define CS4271_CHIPID 0x08 /* Chip ID */ #define CS4271_FIRSTREG CS4271_MODE1 #define CS4271_LASTREG CS4271_MODE2 @@ -144,23 +142,27 @@ * Array do not include Chip ID, as codec driver does not use * registers read operations at all */ -static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { - 0, - 0, - CS4271_DACCTL_AMUTE, - CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, - 0, - 0, - 0, - 0, +static const struct reg_default cs4271_reg_defaults[] = { + { CS4271_MODE1, 0, }, + { CS4271_DACCTL, CS4271_DACCTL_AMUTE, }, + { CS4271_DACVOL, CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, }, + { CS4271_VOLA, 0, }, + { CS4271_VOLB, 0, }, + { CS4271_ADCCTL, 0, }, + { CS4271_MODE2, 0, }, }; +static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == CS4271_CHIPID; +} + struct cs4271_private { /* SND_SOC_I2C or SND_SOC_SPI */ - enum snd_soc_control_type bus_type; unsigned int mclk; bool master; bool deemph; + struct regmap *regmap; /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ @@ -210,14 +212,14 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_LEFT_J: val |= CS4271_MODE1_DAC_DIF_LJ; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); if (ret < 0) return ret; break; case SND_SOC_DAIFMT_I2S: val |= CS4271_MODE1_DAC_DIF_I2S; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); if (ret < 0) return ret; @@ -227,7 +229,7 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); if (ret < 0) return ret; @@ -252,7 +254,7 @@ static int cs4271_set_deemph(struct snd_soc_codec *codec) val <<= 4; } - ret = snd_soc_update_bits(codec, CS4271_DACCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_DACCTL, CS4271_DACCTL_DEM_MASK, val); if (ret < 0) return ret; @@ -341,14 +343,14 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, !dai->capture_active) || (substream->stream == SNDRV_PCM_STREAM_CAPTURE && !dai->playback_active)) { - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, + CS4271_MODE2_PDN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; } @@ -378,7 +380,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, val |= cs4271_clk_tab[i].ratio_mask; - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); if (ret < 0) return ret; @@ -389,6 +391,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); int ret; int val_a = 0; int val_b = 0; @@ -398,10 +401,13 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) val_b = CS4271_VOLB_MUTE; } - ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLA, + CS4271_VOLA_MUTE, val_a); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLB, + CS4271_VOLB_MUTE, val_b); if (ret < 0) return ret; @@ -463,25 +469,33 @@ static struct snd_soc_dai_driver cs4271_dai = { static int cs4271_soc_suspend(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Set power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, CS4271_MODE2_PDN); if (ret < 0) return ret; + return 0; } static int cs4271_soc_resume(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Restore codec state */ - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(cs4271->regmap); if (ret < 0) return ret; + /* then disable the power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; + return 0; } #else @@ -542,40 +556,22 @@ static int cs4271_probe(struct snd_soc_codec *codec) cs4271->gpio_nreset = gpio_nreset; - /* - * In case of I2C, chip address specified in board data. - * So cache IO operations use 8 bit codec register address. - * In case of SPI, chip address and register address - * passed together as 16 bit value. - * Anyway, register address is masked with 0xFF inside - * soc-cache code. - */ - if (cs4271->bus_type == SND_SOC_SPI) - ret = snd_soc_codec_set_cache_io(codec, 16, 8, - cs4271->bus_type); - else - ret = snd_soc_codec_set_cache_io(codec, 8, 8, - cs4271->bus_type); - if (ret) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; /* Power-up sequence requires 85 uS */ udelay(85); if (amutec_eq_bmutec) - snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_MUTECAEQUB, - CS4271_MODE2_MUTECAEQUB); + regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_MUTECAEQUB, + CS4271_MODE2_MUTECAEQUB); return snd_soc_add_codec_controls(codec, cs4271_snd_controls, ARRAY_SIZE(cs4271_snd_controls)); @@ -597,13 +593,24 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .remove = cs4271_remove, .suspend = cs4271_soc_suspend, .resume = cs4271_soc_resume, - .reg_cache_default = cs4271_dflt_reg, - .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg), - .reg_word_size = sizeof(cs4271_dflt_reg[0]), - .compress_type = SND_SOC_FLAT_COMPRESSION, }; #if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config cs4271_spi_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = CS4271_LASTREG, + .read_flag_mask = 0x21, + .write_flag_mask = 0x20, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_spi_probe(struct spi_device *spi) { struct cs4271_private *cs4271; @@ -613,7 +620,9 @@ static int cs4271_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, cs4271); - cs4271->bus_type = SND_SOC_SPI; + cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); @@ -643,6 +652,18 @@ static const struct i2c_device_id cs4271_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); +static const struct regmap_config cs4271_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = CS4271_LASTREG, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -653,7 +674,9 @@ static int cs4271_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, cs4271); - cs4271->bus_type = SND_SOC_I2C; + cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); -- 1.8.1.4

2 3

Re: [alsa-devel] [PATCH 08/10] ASoC: dmaengine_pcm: Add open function for DT DMA request
by Mark Brown 15 Mar '13

15 Mar '13

On Sun, Mar 10, 2013 at 07:33:09PM +0100, Markus Pargmann wrote: > Add a function to open a DMA PCM substream using devicetree data > provided via the client device node. The patch introduces a public > function and a private subfunction that is called by both open > functions. Someone (I think it was Shawn) sent a very similar patch just recently which appears to have fallen out of my inbox unfortunately - can you please check what's going on there and coordinate with them? Let me know if you can't find the patch and I'll dig it out.

2 4

Re: [alsa-devel] [Alsa-user] Master volume control
by Paul D. DeRocco 14 Mar '13

14 Mar '13

> From: Clemens Ladisch [mailto:cladisch@googlemail.com] > > #include <stdio.h> > #include <stdlib.h> > #include <alsa/asoundlib.h> > > static void check(int err, const char *f) > { > if (err < 0) { > fprintf(stderr, "%s failed: %s\n", f, > snd_strerror(err)); > exit(EXIT_FAILURE); > } > } > #define CHECK(f) check(f, #f) > > int main() > { > snd_ctl_t *ctl; > snd_ctl_elem_value_t *value; > > CHECK(snd_ctl_open(&ctl, "default", 0)); > > snd_ctl_elem_value_alloca(&value); > snd_ctl_elem_value_set_interface(value, > SND_CTL_ELEM_IFACE_MIXER); > snd_ctl_elem_value_set_name(value, "Master Playback Volume"); > snd_ctl_elem_value_set_integer(value, 0, 42); > CHECK(snd_ctl_elem_write(ctl, value)); > > snd_ctl_close(ctl); > return 0; > } Well, I finally got a chance to try this. It "works", but the snd_ctl_elem_write call takes a couple of milliseconds (that's around a million instructions on my Atom), and it apparently spends this time in kernel state, because it causes my sound thread to underrun, so it's not usable. If I use alsamixer, I don't have this problem; I can manipulate the master volume smoothly while I'm playing my synth app. So I went through the source, and found that it's using snd_mixer_selem_set_playback_volume. This is apparently part of something called the "Simple Mixer Interface", yet another section of ALSA which is completely undocumented, other than the Doxygen stuff (which is only meaningful as a reference for someone who already knows it inside out, but can't remember e.g. whether playback_pcm comes before or after capture_pcm in the snd_mixer_selem_regopt structure). So what's the minimum I need to do in order to get at the Master Playback Volume, via the Simple Mixer Interface? -- Ciao, Paul D. DeRocco Paul mailto:pderocco@ix.netcom.com

3 6

[alsa-devel] [PATCH v2] ALSA: hda - Disable IDT eapd_switch if there are no internal speakers
by David Henningsson 14 Mar '13

14 Mar '13

If there are no internal speakers, we should not turn the eapd switch off, because it might be necessary to keep high for Headphone. BugLink: https://bugs.launchpad.net/bugs/1155016 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) I'm okay with your suggestion. I only changed from "false" to "0" to be consistent with other usage of eapd_switch. diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d57c81e..3566731 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -815,6 +815,29 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) return 0; } +/* check whether a built-in speaker is included in parsed pins */ +static bool has_builtin_speaker(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + hda_nid_t *nid_pin; + int nids, i; + + if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) { + nid_pin = spec->gen.autocfg.line_out_pins; + nids = spec->gen.autocfg.line_outs; + } else { + nid_pin = spec->gen.autocfg.speaker_pins; + nids = spec->gen.autocfg.speaker_outs; + } + + for (i = 0; i < nids; i++) { + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid_pin[i]); + if (snd_hda_get_input_pin_attr(def_conf) == INPUT_PIN_ATTR_INT) + return true; + } + return false; +} + /* * PC beep controls */ @@ -3891,6 +3914,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; } + /* Don't GPIO-mute speakers if there are no internal speakers, because + * the GPIO might be necessary for Headphone + */ + if (spec->eapd_switch && !has_builtin_speaker(codec)) + spec->eapd_switch = 0; + codec->proc_widget_hook = stac92hd7x_proc_hook; snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Disable IDT eapd_switch if there are no internal speakers
by David Henningsson 14 Mar '13

14 Mar '13

If there are no internal speakers, we should not turn the eapd switch off, because it might be necessary to keep high for Headphone. BugLink: https://bugs.launchpad.net/bugs/1155016 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) Hi Takashi, I encountered the bug when working with a pre-release machine (so no alsa-info, unfortunately). Feel free to commit if you think this is a good idea. Or can you think of a use case where this would cause a regression? diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d57c81e..66200ea8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3524,6 +3524,31 @@ static int stac_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + /* Don't GPIO-mute speakers if there are no internal speakers, because + the GPIO might be necessary for Headphone */ + if (spec->eapd_switch) { + hda_nid_t *nid_pin; + int nids, i; + bool found = false; + if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) { + nid_pin = spec->gen.autocfg.line_out_pins; + nids = spec->gen.autocfg.line_outs; + } else { + nid_pin = spec->gen.autocfg.speaker_pins; + nids = spec->gen.autocfg.speaker_outs; + } + for (i = 0; i < nids; i++) { + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid_pin[i]); + unsigned int attr = snd_hda_get_input_pin_attr(def_conf); + if (attr == INPUT_PIN_ATTR_INT) { + found = true; + break; + } + } + if (!found) + spec->eapd_switch = 0; + } + /* minimum value is actually mute */ spec->gen.vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ALSA: info: Avoid leaking kernel memory
by Takashi Iwai 14 Mar '13

14 Mar '13

Make sure that the allocated buffer for reading the proc file won't expose the uncleared kernel memory. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/info.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/info.c b/sound/core/info.c index db308db..58e97b3 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -89,7 +89,7 @@ static int resize_info_buffer(struct snd_info_buffer *buffer, char *nbuf; nsize = PAGE_ALIGN(nsize); - nbuf = krealloc(buffer->buffer, nsize, GFP_KERNEL); + nbuf = krealloc(buffer->buffer, nsize, GFP_KERNEL | __GFP_ZERO); if (! nbuf) return -ENOMEM; @@ -353,7 +353,7 @@ static int snd_info_entry_open(struct inode *inode, struct file *file) goto __nomem; data->rbuffer = buffer; buffer->len = PAGE_SIZE; - buffer->buffer = kmalloc(buffer->len, GFP_KERNEL); + buffer->buffer = kzalloc(buffer->len, GFP_KERNEL); if (buffer->buffer == NULL) goto __nomem; } -- 1.8.1.4

2 3

[alsa-devel] [PATCH] ASoC: arizona: Ensure we clock two channels for I2S mode
by Mark Brown 13 Mar '13

13 Mar '13

I2S requires stereo clocking even for mono data. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0c70d503..2b0803e 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -818,7 +818,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, struct arizona *arizona = priv->arizona; int base = dai->driver->base; const int *rates; - int i, ret; + int i, ret, val; int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; @@ -834,6 +834,13 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, bclk_target *= chan_limit; } + /* Force stereo for I2S mode */ + val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); + if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) { + arizona_aif_dbg(dai, "Forcing stereo mode\n"); + bclk_target *= 2; + } + for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { if (rates[i] >= bclk_target && rates[i] % params_rate(params) == 0) { -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/3] ALSA: hda - Drop explicit memset() by reallocation with __GFP_ZERO
by Takashi Iwai 13 Mar '13

13 Mar '13

Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 04b5738..a1c5053 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5539,10 +5539,9 @@ void *snd_array_new(struct snd_array *array) void *nlist; if (snd_BUG_ON(num >= 4096)) return NULL; - nlist = krealloc(array->list, size, GFP_KERNEL); + nlist = krealloc(array->list, size, GFP_KERNEL | __GFP_ZERO); if (!nlist) return NULL; - memset(nlist + oldsize, 0, size - oldsize); array->list = nlist; array->alloced = num; } -- 1.8.1.4

1 2

[alsa-devel] How to send non-SysEx system messages via ALSA?
by Joerg-Cyril.Hoehle＠t-systems.com 13 Mar '13

13 Mar '13

Hi, seqmid.h contains function that declare the major event types like snd_seq_ev_set_noteon and snd_seq_ev_set_sysex. However, seq_event.h knows many more types, e.g. SND_SEQ_EVENT_START and SND_SEQ_EVENT_CLOCK/TICK. Now, consider an app playing a MIDI file or a debug tool that allows to send arbitrary messages. How should such an app send e.g. the 2-byte MTC Quarter Frame sequence via ALSA? A. Use a 2 byte buffer F1 xx and snd_seq_ev_set_sysex(&event, 2, buf); B. Use SND_SEQ_EVENT_TICK and figure out some copy&paste code from seqmidi.h, perhaps snd_seq_ev_set_queue_control(ev,SND_SEQ_EVENT_TICK,SND_SEQ_QUEUE_DIRECT,xx) But why does that alter my destination port via snd_seq_ev_set_dest(ev, SND_SEQ_CLIENT_SYSTEM, SND_SEQ_PORT_SYSTEM_TIMER)? C. It's SND_SEQ_EVENT_CLOCK actually, TICK being the one byte F8 real-time message? Background: Wine on Linux needs to generate such messages on behalf of applications and users with real MIDI HW. So far Wine assumes that it can encapsulate all system messages in a SysEx, IOW Wine does A. Is that usage of the set_sysex API correct? More generally, can Wine transparently stuff system messages into a byte buffer and send them off via snd_seq_ev_set_sysex? Thank you, Jörg Höhle

5 8

[alsa-devel] [patch] ALSA: ASoC: tas5086: signedness bug in tas5086_hw_params()
by Dan Carpenter 13 Mar '13

13 Mar '13

"val" has to be signed for the error handling to work. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 008bea4..41d03ae 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -251,7 +251,7 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - unsigned int val; + int val; int ret; priv->rate = params_rate(params);

3 2

[alsa-devel] [PATCH] ASoC: fsi: remove unused irq
by Kuninori Morimoto 13 Mar '13

13 Mar '13

FSI is using devm_request_irq() from 1ddd82868cc888e008ed520465c172a6cdddd689 (ASoC: fsi: use devm_request_irq()) master->irq is no longer needed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index a606d0f..7a874a3 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -300,7 +300,6 @@ struct fsi_core { struct fsi_master { void __iomem *base; - int irq; struct fsi_priv fsia; struct fsi_priv fsib; struct fsi_core *core; @@ -2086,7 +2085,6 @@ static int fsi_probe(struct platform_device *pdev) } /* master setting */ - master->irq = irq; master->core = (struct fsi_core *)id_entry->driver_data; spin_lock_init(&master->lock); -- 1.7.9.5

2 3

[alsa-devel] [PATCH 2/2] ASoC: fsl: imx-audmux: Use devm_clk_get()
by Fabio Estevam 13 Mar '13

13 Mar '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> By using devm_clk_get() we can save a call to clk_put(). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 3f333e5..47f046a 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -262,7 +262,7 @@ static int imx_audmux_probe(struct platform_device *pdev) return PTR_ERR(pinctrl); } - audmux_clk = clk_get(&pdev->dev, "audmux"); + audmux_clk = devm_clk_get(&pdev->dev, "audmux"); if (IS_ERR(audmux_clk)) { dev_dbg(&pdev->dev, "cannot get clock: %ld\n", PTR_ERR(audmux_clk)); @@ -282,7 +282,6 @@ static int imx_audmux_remove(struct platform_device *pdev) { if (audmux_type == IMX31_AUDMUX) audmux_debugfs_remove(); - clk_put(audmux_clk); return 0; } -- 1.7.9.5

2 3

Re: [alsa-devel] [PATCH] ASoC: samsung: remove last traces of neo1973-gta01
by Mark Brown 13 Mar '13

13 Mar '13

On Tue, Mar 12, 2013 at 11:09:35PM +0100, Paul Bolle wrote: > The (rudimentary) support for the Openmoko Neo1973 GTA01 got removed in > commit 1ae5cbc52e7c6619a3f44b87809fd25370df31bb ("ASoC: neo1973_wm8753: > remove references to the neo1973-gta01 machine"). Remove its last traces > in the Kconfig file too. The support wasn't rudimentary, it was a pretty much complete audio driver. Anyway I've applied this with a fixed up commit message.

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Re: [alsa-devel] [PATCH] ASoC: samsung: remove last traces of neo1973-gta01
by Kukjin Kim 13 Mar '13

13 Mar '13

Paul Bolle wrote: > > The (rudimentary) support for the Openmoko Neo1973 GTA01 got removed in > commit 1ae5cbc52e7c6619a3f44b87809fd25370df31bb ("ASoC: neo1973_wm8753: > remove references to the neo1973-gta01 machine"). Remove its last traces > in the Kconfig file too. > > Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> Acked-by: Kukjin Kim <kgene.kim(a)samsung.com> - Kukjin > --- > Untested. > > sound/soc/samsung/Kconfig | 5 ++--- > 1 file changed, 2 insertions(+), 3 deletions(-) > > diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig > index 90e7e66..475fb0d 100644 > --- a/sound/soc/samsung/Kconfig > +++ b/sound/soc/samsung/Kconfig > @@ -35,11 +35,10 @@ config SND_SAMSUNG_I2S > tristate > > config SND_SOC_SAMSUNG_NEO1973_WM8753 > - tristate "Audio support for Openmoko Neo1973 Smartphones (GTA01/GTA02)" > - depends on SND_SOC_SAMSUNG && (MACH_NEO1973_GTA01 || MACH_NEO1973_GTA02) > + tristate "Audio support for Openmoko Neo1973 Smartphones (GTA02)" > + depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02 > select SND_S3C24XX_I2S > select SND_SOC_WM8753 > - select SND_SOC_LM4857 if MACH_NEO1973_GTA01 > select SND_SOC_DFBMCS320 > help > Say Y here to enable audio support for the Openmoko Neo1973 > -- > 1.7.11.7

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Re: [alsa-devel] HDA regression on Fujitsu S7020 laptop (ALC260 codec)
by Jonathan Woithe 12 Mar '13

12 Mar '13

Hi Takashi > > On Wed, Feb 20, 2013 at 02:14:20PM +0100, Takashi Iwai wrote: > > > > Sounds good - I will test in the next day or so and report back. What's the > > > > most efficient way to get hold of this tree / branch given that I have the > > > > for-next branch of sound.git? Can I somehow leverage this, or do I need to > > > > do a completely separate checkout? Apologies if this is a basic question - > > > > I'm still coming to terms with git. > > > > > > sound-unstable git tree always contains the sound git tree master > > > branch, so everything is there basically. ... > > > > What's the next step in the process? I've reported the outcome of the tests > > of sound-unstable as per that report progress is being made. Is it a case > > of waiting for things to stabilise, or do you need me to do something to > > move things along? If so, what? > > I merged the patches to the main sound git tree for-next branch now. > > Also a few updates are merged there, too: now the hp mic jack mode > should be able to change the pin control no matter whether which > capture source is. Also, the capture source won't change the hp/mic > pinctl when the jack mode enum is created. This reduces the > automation but also reduces the confusion, too. After all, when jack > mode is present, user wants to control the pin by himself. > > Let me know if anything else is missing or broken. I've given this new revision a quick test; sorry for the delay. With this revision, when using "model=fujitsu-jwse" things seem to be working pretty well. In particular, the updates you mention greatly improve usability; the lack of automatic interaction between mixer controls makes it much easier to use and understand. I think everything is present and accounted for in this latest revision. Thanks for your work on this. The only slightly odd thing I noticed was that whenever the driver is loaded the "headphone mic jack mode" defaults to "mic bias 80pc", rather than "line out" or "headphone" which I would have expected. Obviously this can be manually or automatically overridden after module load, but in many ways it would be good if it defaulted to one of the output modes (given that the jack's primary purpose is for output). Where does this leave us now? Is the "fujitsu-jwse" a permanent thing, or will the respective changes be rolled into something more generic? Regards jonathan

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Re: [alsa-devel] [PATCH linux-next v2] ASoC: adau1373: adau1373_hw_params: Silence overflow warning
by Mark Brown 12 Mar '13

12 Mar '13

On Tue, Mar 12, 2013 at 01:17:26PM -0600, Tim Gardner wrote: > > threads (I nearly deleted this unread) and avoid including > > enormous CC lists like this in the body of the mail. > I'm just going by the default output of scripts/get_maintainer.pl. Are > there better options to filter the roles ? In this case I received a > review from a supporter (Lars), and the patch is being committed by > yet another supporter (yourself). Would you have noticed this patch > had I restricted the Cc list to just maintainers and lists ? I specifically mean don't include the lists in the body of the mail - it means they either litter the git log or need to be manually deleted. If you're just using unfiltered get_maintainer output then I'd suggest that you generate the CCs in your scripts when sending rather than putting them in the changelog.

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[alsa-devel] [PATCH] ALSA: ASoC: davinci-mcasp: clean ip davinci_hw_common_param()
by Daniel Mack 12 Mar '13

12 Mar '13

As pointed of by Vaibhav, commit 2952b27e2 ("ASoC: davinci-mcasp: Add support for multichannel playback") duplicated the logic of counting the active serializers. That can be avoided by shifting the code around a bit. Also, drop two unused defines introduced by the same commit. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Michal Bachraty <michal.bachraty(a)streamunlimited.com> Cc: Vaibhav Bedia <vaibhav.bedia(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 34 +++++++++++----------------------- 1 file changed, 11 insertions(+), 23 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index b386cad..8a9c0ee 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -237,8 +237,6 @@ #define RXSTATE BIT(5) #define SRMOD_MASK 3 #define SRMOD_INACTIVE 0 -#define SRMOD_TX 1 -#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -691,27 +689,6 @@ static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, } for (i = 0; i < dev->num_serializer; i++) { - if (dev->serial_dir[i] == TX_MODE) - tx_ser++; - if (dev->serial_dir[i] == RX_MODE) - rx_ser++; - } - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - ser = tx_ser; - else - ser = rx_ser; - - if (ser < max_active_serializers) { - dev_warn(dev->dev, "stream has more channels (%d) than are " - "enabled in mcasp (%d)\n", channels, ser * slots); - return -EINVAL; - } - - tx_ser = 0; - rx_ser = 0; - - for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); if (dev->serial_dir[i] == TX_MODE && @@ -730,6 +707,17 @@ static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, } } + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ser = tx_ser; + else + ser = rx_ser; + + if (ser < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + return -EINVAL; + } + if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { if (dev->txnumevt * tx_ser > 64) dev->txnumevt = 1; -- 1.8.1.4

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Re: [alsa-devel] [PATCH 07/10] ASoC: phycore-ac97: Add DT support
by Mark Brown 12 Mar '13

12 Mar '13

On Sun, Mar 10, 2013 at 07:33:08PM +0100, Markus Pargmann wrote: > Add devicetree support for this audio soc fabric driver. > > Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> > --- > sound/soc/fsl/phycore-ac97.c | 148 ++++++++++++++++++++++++++++++++++++------- > 1 file changed, 125 insertions(+), 23 deletions(-) A binding document is mandatory for new bindings. > +#ifdef CONFIG_MACH_IMX27_DT > + .cpu_dai_name = "10010000.ssi", > + .platform_name = "imx-fiq-pcm-audio", > +#else This looks wrong, at least the CPU DAI name should be being looked up via the DT. > +static const struct of_device_id imx_phycore_ac97_of_dev_id[] = { > + { > + .compatible = "phytec,imx27-pca100-ac97", > + .data = (void *)MX27_PCA100, > + }, { > + .compatible = "phytec,imx27-pcm043-ac97", > + .data = (void *)MX27_PCM043 > + }, { > + /* sentinel */ > + } > +}; > +MODULE_DEVICE_TABLE(of, imx_phycore_ac97_of_dev_id); It seems like we ought to be able to describe the properties of the boards in this class rather than just enumerating the boards. What are the differences? > MODULE_AUTHOR("Sascha Hauer <s.hauer(a)pengutronix.de>"); > -MODULE_DESCRIPTION("PhyCORE ALSA SoC driver"); > +MODULE_DESCRIPTION(DRV_NAME ": PhyCORE ALSA SoC fabric driver"); Please don't make up terms like "fabric driver".

1 0

Re: [alsa-devel] [PATCH 05/10] ASoC: pcm030 audio fabric: remove __init from probe
by Mark Brown 12 Mar '13

12 Mar '13

On Sun, Mar 10, 2013 at 07:33:06PM +0100, Markus Pargmann wrote: > Remove probe function from the init section. > > Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 04/10] ASoC: Kconfig: imx-ssi needs imx-fiq and imx-dma
by Mark Brown 12 Mar '13

12 Mar '13

On Sun, Mar 10, 2013 at 07:33:05PM +0100, Markus Pargmann wrote: > config SND_SOC_IMX_SSI > + select SND_SOC_IMX_PCM_DMA > + select SND_SOC_IMX_PCM_FIQ > tristate Really? I'd expect this to be an or, if we've got DMA I can't see any reason why we'd want to use the FIQ.

1 0

Re: [alsa-devel] [PATCH 03/10] mx-ssi: flush fifos
by Mark Brown 12 Mar '13

12 Mar '13

On Sun, Mar 10, 2013 at 07:33:04PM +0100, Markus Pargmann wrote: > From: Sascha Hauer <s.hauer(a)pengutronix.de> > > Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> > Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> Why? Is this a bug fix? I'm guessing it is but...

1 0

Re: [alsa-devel] [PATCH 02/10] imx-ssi: Fix occasional AC97 reset failure
by Mark Brown 12 Mar '13

12 Mar '13

On Sun, Mar 10, 2013 at 07:33:03PM +0100, Markus Pargmann wrote: > From: Sascha Hauer <s.hauer(a)pengutronix.de> > > Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> > Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> Applied but *always* use subject lines appropriate to the subsystem and bugfixes like this should be pulled to the front of the series as they should be sent to mainline earlier.

1 0

Re: [alsa-devel] [PATCH 01/10] imx-ssi: Fix AC97 rates
by Mark Brown 12 Mar '13

12 Mar '13

On Sun, Mar 10, 2013 at 07:33:02PM +0100, Markus Pargmann wrote: > From: Sascha Hauer <s.hauer(a)pengutronix.de> > > This device supports multiple rates as described in later AC97 > standards. This patch allows playback of different sample frequencies > without conversion. Applied, but some notes about why this is already supported would be useful and *always* use subject lines appropriate for the subsystem.

1 0

Re: [alsa-devel] [PATCH linux-next] ASoC: wm_hubs: Silence reg_r and reg_l 'may be used uninitialized' warnings
by Mark Brown 12 Mar '13

12 Mar '13

On Sun, Mar 10, 2013 at 10:58:21AM -0600, Tim Gardner wrote: > Return an error from wm_hubs_read_dc_servo() if hubs->dcs_readback_mode is not > correctly initialized. You might as well bail out since nothing is likely to > work correctly afterwards. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH linux-next v2] ASoC: adau1373: adau1373_hw_params: Silence overflow warning
by Mark Brown 12 Mar '13

12 Mar '13

On Mon, Mar 11, 2013 at 01:18:23PM -0600, Tim Gardner wrote: > ADAU1373_BCLKDIV_SOURCE is defined as BIT(5) which uses UL constants. On > amd64 the result of the ones complement operator is then truncated to Applied, but please don't bury patches in the middle of e-mail threads (I nearly deleted this unread) and avoid including > Cc: Lars-Peter Clausen <lars(a)metafoo.de> > Cc: Liam Girdwood <lgirdwood(a)gmail.com> > Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: device-drivers-devel(a)blackfin.uclinux.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> enormous CC lists like this in the body of the mail.

1 0

Re: [alsa-devel] [PATCH] omap: convert to devm_ioremap_resource()
by Mark Brown 12 Mar '13

12 Mar '13

On Mon, Mar 11, 2013 at 05:58:57PM +0200, Silviu-Mihai Popescu wrote: > Convert all uses of devm_request_and_ioremap() to the newly introduced > devm_ioremap_resource() which provides more consistent error handling. > > devm_ioremap_resource() provides its own error messages so all explicit > error messages can be removed from the failure code paths. Applied, thanks. Please use subject lines appropriate to the subsystem.

1 0

Re: [alsa-devel] [PATCH] ASoC: core: fix possible memory leak in snd_soc_bytes_put()
by Mark Brown 12 Mar '13

12 Mar '13

On Tue, Mar 12, 2013 at 12:35:14AM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > 'data' is malloced in snd_soc_bytes_put() and should be freed > before leaving from the error handling cases, otherwise it will cause > memory leak. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: wm_adsp: fix possible memory leak in wm_adsp_load_coeff()
by Mark Brown 12 Mar '13

12 Mar '13

On Tue, Mar 12, 2013 at 12:23:15AM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > 'file' is malloced in wm_adsp_load_coeff() and should be freed > before leaving from the error handling cases, otherwise it will cause > memory leak. Applied, thanks.

1 0

[alsa-devel] [RFC][PATCH] ASoC: remove codec from list if registration failed
by Kuninori Morimoto 12 Mar '13

12 Mar '13

Current snd_soc_register_codec() adds codec to list, and calls snd_soc_register_dais(). But, this listed codec should be removed if dais registration was failed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- If my understanding is correct... sound/soc/soc-core.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1ac96df..38d6182 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4097,6 +4097,10 @@ int snd_soc_register_codec(struct device *dev, return 0; fail_codec_name: + mutex_lock(&client_mutex); + list_del(&codec->list); + mutex_unlock(&client_mutex); + kfree(codec->name); fail_codec: kfree(codec); -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH -next] ASoC: tas5086: use module_i2c_driver to simplify the code
by Mark Brown 12 Mar '13

12 Mar '13

On Tue, Mar 12, 2013 at 09:36:24PM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Use the module_i2c_driver() macro to make the code smaller > and a bit simpler. Applied, thanks.

1 0

[alsa-devel] [PATCH] hda-emu: stream tag should always be non-zero
by David Henningsson 12 Mar '13

12 Mar '13

Sending zero as stream tag is discouraged, because it means no data, and the real code doesn't do that. Avoid it here too. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-emu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/hda-emu.c b/hda-emu.c index c98a3ea..13756bb 100644 --- a/hda-emu.c +++ b/hda-emu.c @@ -760,7 +760,7 @@ void hda_test_pcm(int id, int op, int subid, } hda_log(HDA_LOG_INFO, "PCM format_val = 0x%x\n", format_val); #ifdef HAVE_COMMON_PREPARE - err = snd_hda_codec_prepare(_codec, hinfo, subid, + err = snd_hda_codec_prepare(_codec, hinfo, subid + 1, format_val, substream); #else err = hinfo->ops.prepare(hinfo, _codec, 1, format_val, substream); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] hda-emu: Fix typo in HAVE_COMMON_PREPARE
by David Henningsson 12 Mar '13

12 Mar '13

This typo caused snd_hda_codec_prepare never to be executed by hda-emu. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 4b20f50..01a3a16 100644 --- a/configure.ac +++ b/configure.ac @@ -501,7 +501,7 @@ if grep -q snd_hda_codec_prepare hda/hda_codec.h; then else AC_MSG_RESULT(no) fi -if test "$HAVE_COMMON_PREAPRE" = "yes"; then +if test "$HAVE_COMMON_PREPARE" = "yes"; then AC_DEFINE(HAVE_COMMON_PREPARE) fi -- 1.7.9.5

2 1

[alsa-devel] [PATCH] Revert "ALSA: hda - Remove limit of widget connections"
by David Henningsson 12 Mar '13

12 Mar '13

This reverts commit 4eea30914facd2c99061cd70e5b05d3c76c743a2. The commit causes an error with node ranges: In my case, connection list length is 3, with a connection list read of 0x1e9915: a total of 6 nids (0x15, 0x16, 0x17, 0x18, 0x19, 0x1e). Since the new code makes us allocate only 3 nids to the array sent to get_raw_connections, the function fails. --- sound/pci/hda/hda_codec.c | 62 +++++++++++++++----------------------------- sound/pci/hda/hda_codec.h | 4 ++- sound/pci/hda/hda_proc.c | 20 +++----------- sound/pci/hda/patch_hdmi.c | 3 --- 4 files changed, 28 insertions(+), 61 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 97c68dd..e5e8549 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -382,23 +382,13 @@ static void remove_conn_list(struct hda_codec *codec) /* read the connection and add to the cache */ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) { - hda_nid_t list[32]; - hda_nid_t *result = list; + hda_nid_t list[HDA_MAX_CONNECTIONS]; int len; len = snd_hda_get_raw_connections(codec, nid, list, ARRAY_SIZE(list)); - if (len == -ENOSPC) { - len = snd_hda_get_num_raw_conns(codec, nid); - result = kmalloc(sizeof(hda_nid_t) * len, GFP_KERNEL); - if (!result) - return -ENOMEM; - len = snd_hda_get_raw_connections(codec, nid, result, len); - } - if (len >= 0) - len = snd_hda_override_conn_list(codec, nid, len, result); - if (result != list) - kfree(result); - return len; + if (len < 0) + return len; + return snd_hda_override_conn_list(codec, nid, len, list); } /** @@ -476,27 +466,6 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_get_connections); -/* return CONNLIST_LEN parameter of the given widget */ -static unsigned int get_num_conns(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int wcaps = get_wcaps(codec, nid); - unsigned int parm; - - if (!(wcaps & AC_WCAP_CONN_LIST) && - get_wcaps_type(wcaps) != AC_WID_VOL_KNB) - return 0; - - parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); - if (parm == -1) - parm = 0; - return parm; -} - -int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid) -{ - return get_num_conns(codec, nid) & AC_CLIST_LENGTH; -} - /** * snd_hda_get_raw_connections - copy connection list without cache * @codec: the HDA codec @@ -514,16 +483,19 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, unsigned int parm; int i, conn_len, conns; unsigned int shift, num_elems, mask; + unsigned int wcaps; hda_nid_t prev_nid; int null_count = 0; if (snd_BUG_ON(!conn_list || max_conns <= 0)) return -EINVAL; - parm = get_num_conns(codec, nid); - if (!parm) + wcaps = get_wcaps(codec, nid); + if (!(wcaps & AC_WCAP_CONN_LIST) && + get_wcaps_type(wcaps) != AC_WID_VOL_KNB) return 0; + parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); if (parm & AC_CLIST_LONG) { /* long form */ shift = 16; @@ -580,13 +552,21 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, continue; } for (n = prev_nid + 1; n <= val; n++) { - if (conns >= max_conns) - return -ENOSPC; + if (conns >= max_conns) { + snd_printk(KERN_ERR "hda_codec: " + "Too many connections %d for NID 0x%x\n", + conns, nid); + return -EINVAL; + } conn_list[conns++] = n; } } else { - if (conns >= max_conns) - return -ENOSPC; + if (conns >= max_conns) { + snd_printk(KERN_ERR "hda_codec: " + "Too many connections %d for NID 0x%x\n", + conns, nid); + return -EINVAL; + } conn_list[conns++] = val; } prev_nid = val; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 23ca172..246ed53 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -551,6 +551,9 @@ enum { AC_JACK_PORT_BOTH, }; +/* max. connections to a widget */ +#define HDA_MAX_CONNECTIONS 32 + /* max. codec address */ #define HDA_MAX_CODEC_ADDRESS 0x0f @@ -957,7 +960,6 @@ snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) { return snd_hda_get_connections(codec, nid, NULL, 0); } -int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid); int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 0fee8fa..5e02f26 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -22,7 +22,6 @@ */ #include <linux/init.h> -#include <linux/slab.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" @@ -624,7 +623,7 @@ static void print_codec_info(struct snd_info_entry *entry, snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); unsigned int wid_type = get_wcaps_type(wid_caps); - hda_nid_t *conn = NULL; + hda_nid_t conn[HDA_MAX_CONNECTIONS]; int conn_len = 0; snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid, @@ -661,18 +660,9 @@ static void print_codec_info(struct snd_info_entry *entry, if (wid_type == AC_WID_VOL_KNB) wid_caps |= AC_WCAP_CONN_LIST; - if (wid_caps & AC_WCAP_CONN_LIST) { - conn_len = snd_hda_get_num_raw_conns(codec, nid); - if (conn_len > 0) { - conn = kmalloc(sizeof(hda_nid_t) * conn_len, - GFP_KERNEL); - if (!conn) - return; - if (snd_hda_get_raw_connections(codec, nid, conn, - conn_len) < 0) - conn_len = 0; - } - } + if (wid_caps & AC_WCAP_CONN_LIST) + conn_len = snd_hda_get_raw_connections(codec, nid, conn, + HDA_MAX_CONNECTIONS); if (wid_caps & AC_WCAP_IN_AMP) { snd_iprintf(buffer, " Amp-In caps: "); @@ -745,8 +735,6 @@ static void print_codec_info(struct snd_info_entry *entry, if (codec->proc_widget_hook) codec->proc_widget_hook(buffer, codec, nid); - - kfree(conn); } snd_hda_power_down(codec); } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78e1827..d24af4a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -64,9 +64,6 @@ struct hdmi_spec_per_cvt { unsigned int maxbps; }; -/* max. connections to a widget */ -#define HDA_MAX_CONNECTIONS 32 - struct hdmi_spec_per_pin { hda_nid_t pin_nid; int num_mux_nids; -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH -next] ASoC: tas5086: use module_i2c_driver to simplify the code
by Daniel Mack 12 Mar '13

12 Mar '13

On 12.03.2013 14:36, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Use the module_i2c_driver() macro to make the code smaller > and a bit simpler. Thanks for catching this! > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Acked-by: Daniel Mack <zonque(a)gmail.com> > --- > sound/soc/codecs/tas5086.c | 12 +----------- > 1 file changed, 1 insertion(+), 11 deletions(-) > > diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c > index 008bea4..40cee84 100644 > --- a/sound/soc/codecs/tas5086.c > +++ b/sound/soc/codecs/tas5086.c > @@ -584,17 +584,7 @@ static struct i2c_driver tas5086_i2c_driver = { > .remove = tas5086_i2c_remove, > }; > > -static int __init tas5086_modinit(void) > -{ > - return i2c_add_driver(&tas5086_i2c_driver); > -} > -module_init(tas5086_modinit); > - > -static void __exit tas5086_modexit(void) > -{ > - i2c_del_driver(&tas5086_i2c_driver); > -} > -module_exit(tas5086_modexit); > +module_i2c_driver(tas5086_i2c_driver); > > MODULE_AUTHOR("Daniel Mack <zonque(a)gmail.com>"); > MODULE_DESCRIPTION("Texas Instruments TAS5086 ALSA SoC Codec Driver"); > >

1 0

[alsa-devel] [PATCH] salsa-lib: define __SALSA_EXPORT_FUNC in version.h
by Mihail Zenkov 12 Mar '13

12 Mar '13

We have to include "global.h" to define __SALSA_EXPORT_FUNC --- salsa-lib-0.1.3/src/version.h.in +++ salsa-lib-0.1.3-1/src/version.h.in @@ -1,6 +1,8 @@ #ifndef __ALSA_VERSION_H #define __ALSA_VERSION_H +#include "global.h" + #define SND_LIB_MAJOR @SND_LIB_MAJOR@ #define SND_LIB_MINOR @SND_LIB_MINOR@ #define SND_LIB_SUBMINOR @SND_LIB_SUBMINOR@

2 2

Re: [alsa-devel] [PATCH] omap: convert to devm_ioremap_resource()
by Peter Ujfalusi 12 Mar '13

12 Mar '13

On 03/11/2013 04:58 PM, Silviu-Mihai Popescu wrote: > Convert all uses of devm_request_and_ioremap() to the newly introduced > devm_ioremap_resource() which provides more consistent error handling. > > devm_ioremap_resource() provides its own error messages so all explicit > error messages can be removed from the failure code paths. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > > Signed-off-by: Silviu-Mihai Popescu <silviupopescu1990(a)gmail.com> > --- > sound/soc/omap/omap-dmic.c | 9 +++------ > sound/soc/omap/omap-mcpdm.c | 8 +++----- > 2 files changed, 6 insertions(+), 11 deletions(-) > > diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c > index 77e9e7e..8ebaf11 100644 > --- a/sound/soc/omap/omap-dmic.c > +++ b/sound/soc/omap/omap-dmic.c > @@ -493,12 +493,9 @@ static int asoc_dmic_probe(struct platform_device *pdev) > goto err_put_clk; > } > > - dmic->io_base = devm_request_and_ioremap(&pdev->dev, res); > - if (!dmic->io_base) { > - dev_err(&pdev->dev, "cannot remap\n"); > - ret = -ENOMEM; > - goto err_put_clk; > - } > + dmic->io_base = devm_ioremap_resource(&pdev->dev, res); > + if (IS_ERR(dmic->io_base)) > + return PTR_ERR(dmic->io_base); > > ret = snd_soc_register_dai(&pdev->dev, &omap_dmic_dai); > if (ret) > diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c > index 079f277..ddfcc18 100644 > --- a/sound/soc/omap/omap-mcpdm.c > +++ b/sound/soc/omap/omap-mcpdm.c > @@ -464,11 +464,9 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) > if (res == NULL) > return -ENOMEM; > > - mcpdm->io_base = devm_request_and_ioremap(&pdev->dev, res); > - if (!mcpdm->io_base) { > - dev_err(&pdev->dev, "cannot remap\n"); > - return -ENOMEM; > - } > + mcpdm->io_base = devm_ioremap_resource(&pdev->dev, res); > + if (IS_ERR(mcpdm->io_base)) > + return PTR_ERR(mcpdm->io_base); > > mcpdm->irq = platform_get_irq(pdev, 0); > if (mcpdm->irq < 0) > -- Péter

1 0

Re: [alsa-devel] [PATCH] omap: convert to devm_ioremap_resource()
by Peter Ujfalusi 12 Mar '13

12 Mar '13

On 03/12/2013 09:24 AM, Silviu Popescu wrote: > As far as I can tell from the mails that you have provided, those > patches replace devm_request_mem_region(), followed by devm_ioremap() > with devm_request_and_ioremap(). > What this patch attempts to do is replace devm_request_and_ioremap() > with the newly introduced devm_ioremap_resource(), for the reasons > expressed in the patch body. Ah, true. -- Péter

1 0

[alsa-devel] [PATCH v3] ALSA: usb: Work around CM6631 sample rate change bug
by Torstein Hegge 12 Mar '13

12 Mar '13

The C-Media CM6631 USB-to-S/PDIF receiver doesn't respond to changes in sample rate while the interface is active. Reset the interface after setting the sampling frequency on sample rate changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous stream, causing distorted sound on sample rate changes. The reset is performed for all current C-Media UAC V2.0 devices, including CM6610, CM6620 and CM6631, but has only been tested with CM6631. Signed-off-by: Torstein Hegge <hegge(a)resisty.net> --- Changes since v2: - More USB ids covered. - int crate -> int cur_rate - unsigned int previous_rate -> int prev_rate sound/usb/clock.c | 50 ++++++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 46 insertions(+), 4 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 5e634a2..98c4e26 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -253,7 +253,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, { struct usb_device *dev = chip->dev; unsigned char data[4]; - int err, crate; + int err, cur_rate, prev_rate; int clock = snd_usb_clock_find_source(chip, fmt->clock); if (clock < 0) @@ -266,6 +266,18 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return -ENXIO; } + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data))) < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", + dev->devnum, iface, fmt->altsetting); + return err; + } + + prev_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; @@ -290,9 +302,39 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return err; } - crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); - if (crate != rate) - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + cur_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + if (cur_rate != rate) { + snd_printd(KERN_WARNING + "current rate %d is different from the runtime rate %d\n", + cur_rate, rate); + } + + /* Some devices doesn't respond to sample rate changes while the + * interface is active. */ + if (cur_rate != prev_rate) { + switch (chip->usb_id) { + /* C-Media CM6610/CM6620/CM6631 */ + case USB_ID(0x054c, 0x06cf): /* Sony */ + case USB_ID(0x0b05, 0x17a8): /* Asus Xonar Essence One */ + case USB_ID(0x0d8c, 0x0301): + case USB_ID(0x0d8c, 0x0302): + case USB_ID(0x0d8c, 0x0304): /* CM6631 (Schiit) */ + case USB_ID(0x0d8c, 0x0305): + case USB_ID(0x0d8c, 0x0306): + case USB_ID(0x0d8c, 0x0309): /* CM6631 */ + case USB_ID(0x0d8c, 0x0310): + case USB_ID(0x0d8c, 0x0311): /* CM6610A */ + case USB_ID(0x0d8c, 0x0312): /* CM6620A */ + case USB_ID(0x0d8c, 0x0313): /* CM6630A */ + case USB_ID(0x0d8c, 0x0314): /* CM6631A */ + case USB_ID(0x0d8c, 0x0315): /* CM6632A */ + case USB_ID(0x0d8c, 0x0319): /* CM6631A */ + case USB_ID(0x200c, 0x1030): /* Reloop */ + usb_set_interface(dev, iface, 0); + usb_set_interface(dev, iface, fmt->altsetting); + break; + } + } return 0; } -- 1.7.10.4

2 3

Re: [alsa-devel] [PATCH] omap: convert to devm_ioremap_resource()
by Peter Ujfalusi 12 Mar '13

12 Mar '13

Hi, On 03/11/2013 04:58 PM, Silviu-Mihai Popescu wrote: > Convert all uses of devm_request_and_ioremap() to the newly introduced > devm_ioremap_resource() which provides more consistent error handling. > > devm_ioremap_resource() provides its own error messages so all explicit > error messages can be removed from the failure code paths. I believe both of the drivers has been patched for this: omap-mcbpdm: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-February/059484.h… http://mailman.alsa-project.org/pipermail/alsa-devel/2013-February/059427.h… omap-dmic: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-February/059485.h… and they should be already queued for next. -- Péter > > Signed-off-by: Silviu-Mihai Popescu <silviupopescu1990(a)gmail.com> > --- > sound/soc/omap/omap-dmic.c | 9 +++------ > sound/soc/omap/omap-mcpdm.c | 8 +++----- > 2 files changed, 6 insertions(+), 11 deletions(-) > > diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c > index 77e9e7e..8ebaf11 100644 > --- a/sound/soc/omap/omap-dmic.c > +++ b/sound/soc/omap/omap-dmic.c > @@ -493,12 +493,9 @@ static int asoc_dmic_probe(struct platform_device *pdev) > goto err_put_clk; > } > > - dmic->io_base = devm_request_and_ioremap(&pdev->dev, res); > - if (!dmic->io_base) { > - dev_err(&pdev->dev, "cannot remap\n"); > - ret = -ENOMEM; > - goto err_put_clk; > - } > + dmic->io_base = devm_ioremap_resource(&pdev->dev, res); > + if (IS_ERR(dmic->io_base)) > + return PTR_ERR(dmic->io_base); > > ret = snd_soc_register_dai(&pdev->dev, &omap_dmic_dai); > if (ret) > diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c > index 079f277..ddfcc18 100644 > --- a/sound/soc/omap/omap-mcpdm.c > +++ b/sound/soc/omap/omap-mcpdm.c > @@ -464,11 +464,9 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) > if (res == NULL) > return -ENOMEM; > > - mcpdm->io_base = devm_request_and_ioremap(&pdev->dev, res); > - if (!mcpdm->io_base) { > - dev_err(&pdev->dev, "cannot remap\n"); > - return -ENOMEM; > - } > + mcpdm->io_base = devm_ioremap_resource(&pdev->dev, res); > + if (IS_ERR(mcpdm->io_base)) > + return PTR_ERR(mcpdm->io_base); > > mcpdm->irq = platform_get_irq(pdev, 0); > if (mcpdm->irq < 0) >

1 0

Re: [alsa-devel] [PATCH][RESEND] Add a USB audio quirk for the NuForce UDH-100 device.
by Clemens Ladisch 12 Mar '13

12 Mar '13

>> David Helstroom wrote: >>> Interface 1 does not exist Please try the patch below. Regards, Clemens --8<---------------------------------------------------------------->8-- ALSA: usb-audio: add a workaround for the NuForce UDH-100 The NuForce UDH-100 numbers its interfaces incorrectly, which makes the interface associations come out wrong, which results in the driver erroring out with the message "Audio class v2 interfaces need an interface association". Work around this by searching for the interface association descriptor also in some other place where it might have ended up. Reported-by: Dave Helstroom <helstroom(a)google.com> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/card.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index df2f6d0..34dc3e8 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -245,6 +245,21 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) usb_ifnum_to_if(dev, ctrlif)->intf_assoc; if (!assoc) { + /* + * Firmware writers cannot count to three. So to find + * the IAD on the NuForce UDH-100, also check the next + * interface. + */ + struct usb_interface *iface = + usb_ifnum_to_if(dev, ctrlif + 1); + if (iface && + iface->intf_assoc && + iface->intf_assoc->bFunctionClass == USB_CLASS_AUDIO && + iface->intf_assoc->bFunctionProtocol == UAC_VERSION_2) + assoc = iface->intf_assoc; + } + + if (!assoc) { snd_printk(KERN_ERR "Audio class v2 interfaces need an interface association\n"); return -EINVAL; }

2 1

Re: [alsa-devel] [PATCH] ALSA: asihpi - fix potential NULL pointer dereference
by Takashi Iwai 12 Mar '13

12 Mar '13

At Tue, 12 Mar 2013 00:16:40 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > The dereference should be moved below the NULL test. > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Applied now. Thanks. Takashi > --- > sound/pci/asihpi/asihpi.c | 3 ++- > 1 file changed, 2 insertions(+), 1 deletion(-) > > diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c > index 3536b07..0aabfed 100644 > --- a/sound/pci/asihpi/asihpi.c > +++ b/sound/pci/asihpi/asihpi.c > @@ -2549,7 +2549,7 @@ static int snd_asihpi_sampleclock_add(struct snd_card_asihpi *asihpi, > > static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi) > { > - struct snd_card *card = asihpi->card; > + struct snd_card *card; > unsigned int idx = 0; > unsigned int subindex = 0; > int err; > @@ -2557,6 +2557,7 @@ static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi) > > if (snd_BUG_ON(!asihpi)) > return -EINVAL; > + card = asihpi->card; > strcpy(card->mixername, "Asihpi Mixer"); > > err = > >

1 0

[alsa-devel] In 3.8 kernel alsa only talks HDMI, hardware also has S/PDIF optical
by Tom Horsley 12 Mar '13

12 Mar '13

See https://bugzilla.redhat.com/show_bug.cgi?id=919630 for details. Comment 4 in that bug is from someone who says it looks like the driver only created the HDMI and not the separate optical S/PDIF output. On my Asus P8H67-V there really are two physically separate audio outputs. I have the optical port connected to my receiver for when I want to pass through the 5.1 audio tracks in movie files and DVDs and the HDMI connected to the TV for more ordinary audio (error beeps from the computer, etc). In kernel 3.7.9 I could output to both devices (one at a time - never tried simultaneous). I've seen a bunch of patches in the archives that seem to be related to this, but if fedora 18 has those patches, they don't seem to be working.

3 4

Re: [alsa-devel] [PATCH linux-next v2] ASoC: adau1373: adau1373_hw_params: Silence overflow warning
by Lars-Peter Clausen 11 Mar '13

11 Mar '13

On 03/11/2013 08:18 PM, Tim Gardner wrote: > ADAU1373_BCLKDIV_SOURCE is defined as BIT(5) which uses UL constants. On > amd64 the result of the ones complement operator is then truncated to > unsigned int according to the prototype of snd_soc_update_bits(). I think > gcc is correctly warning that the upper 32 bits are lost. > > sound/soc/codecs/adau1373.c: In function 'adau1373_hw_params': > sound/soc/codecs/adau1373.c:940:3: warning: large integer implicitly truncated to unsigned type [-Woverflow] > > gcc version 4.6.3 > > Add 2 more BCLKDIV mask macros as explained by Lars: > > The BCLKDIV has three fields. The bitclock divider (bit 0-1), the samplerate > (bit 2-4) and the source select (bit 5). Here we want to update the bitclock > divider field and the samplerate field. When I wrote the code I was lazy and > used ~ADAU1373_BCLKDIV_SOURCE as the mask, which for this register is > functionally equivalent to ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK. > > Cc: Lars-Peter Clausen <lars(a)metafoo.de> > Cc: Liam Girdwood <lgirdwood(a)gmail.com> > Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: device-drivers-devel(a)blackfin.uclinux.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Thanks. > --- > v1 - simple cast solution > v2 - Implement a full set of field masks as suggested by Lars. > > sound/soc/codecs/adau1373.c | 5 ++++- > 1 file changed, 4 insertions(+), 1 deletion(-) > > diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c > index 068b3ae..1aa10dd 100644 > --- a/sound/soc/codecs/adau1373.c > +++ b/sound/soc/codecs/adau1373.c > @@ -133,6 +133,8 @@ struct adau1373 { > #define ADAU1373_DAI_FORMAT_DSP 0x3 > > #define ADAU1373_BCLKDIV_SOURCE BIT(5) > +#define ADAU1373_BCLKDIV_SR_MASK (0x07 << 2) > +#define ADAU1373_BCLKDIV_BCLK_MASK 0x03 > #define ADAU1373_BCLKDIV_32 0x03 > #define ADAU1373_BCLKDIV_64 0x02 > #define ADAU1373_BCLKDIV_128 0x01 > @@ -937,7 +939,8 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, > adau1373_dai->enable_src = (div != 0); > > snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), > - ~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64); > + ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK, > + (div << 2) | ADAU1373_BCLKDIV_64); > > switch (params_format(params)) { > case SNDRV_PCM_FORMAT_S16_LE:

1 0

Re: [alsa-devel] [PATCH] ASoC: adau1373: adau1373_hw_params: Silence overflow warning
by Lars-Peter Clausen 11 Mar '13

11 Mar '13

On 03/11/2013 04:37 PM, Tim Gardner wrote: > On 03/11/2013 04:01 AM, Lars-Peter Clausen wrote: >> On 03/10/2013 06:34 PM, Tim Gardner wrote: >>> ADAU1373_BCLKDIV_SOURCE is defined as BIT(5) which uses UL constants. On >>> amd64 the result of the ones complement operator is then truncated to >>> unsigned int according to the prototype of snd_soc_update_bits(). I think >>> gcc is correctly warning that the upper 32 bits are lost. >>> >>> sound/soc/codecs/adau1373.c: In function 'adau1373_hw_params': >>> sound/soc/codecs/adau1373.c:940:3: warning: large integer implicitly truncated to unsigned type [-Woverflow] >>> >>> gcc version 4.6.3 >>> >>> Cc: Lars-Peter Clausen <lars(a)metafoo.de> >>> Cc: Liam Girdwood <lgirdwood(a)gmail.com> >>> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> >>> Cc: Jaroslav Kysela <perex(a)perex.cz> >>> Cc: Takashi Iwai <tiwai(a)suse.de> >>> Cc: device-drivers-devel(a)blackfin.uclinux.org >>> Cc: alsa-devel(a)alsa-project.org >>> Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> >>> --- >>> sound/soc/codecs/adau1373.c | 2 +- >>> 1 file changed, 1 insertion(+), 1 deletion(-) >>> >>> diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c >>> index 068b3ae..56ed788 100644 >>> --- a/sound/soc/codecs/adau1373.c >>> +++ b/sound/soc/codecs/adau1373.c >>> @@ -132,7 +132,7 @@ struct adau1373 { >>> #define ADAU1373_DAI_FORMAT_I2S 0x2 >>> #define ADAU1373_DAI_FORMAT_DSP 0x3 >>> >>> -#define ADAU1373_BCLKDIV_SOURCE BIT(5) >>> +#define ADAU1373_BCLKDIV_SOURCE (unsigned int)BIT(5) >> >> Hm, that's a bit ugly. How about the following instead: >> >> diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c >> index 068b3ae..1aa10dd 100644 >> --- a/sound/soc/codecs/adau1373.c >> +++ b/sound/soc/codecs/adau1373.c >> @@ -133,6 +133,8 @@ struct adau1373 { >> #define ADAU1373_DAI_FORMAT_DSP 0x3 >> >> #define ADAU1373_BCLKDIV_SOURCE BIT(5) >> +#define ADAU1373_BCLKDIV_SR_MASK (0x07 << 2) >> +#define ADAU1373_BCLKDIV_BCLK_MASK 0x03 >> #define ADAU1373_BCLKDIV_32 0x03 >> #define ADAU1373_BCLKDIV_64 0x02 >> #define ADAU1373_BCLKDIV_128 0x01 >> @@ -937,7 +939,8 @@ static int adau1373_hw_params(struct snd_pcm_substream >> *substream, >> adau1373_dai->enable_src = (div != 0); >> >> snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), >> - ~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64); >> + ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK, >> + (div << 2) | ADAU1373_BCLKDIV_64); >> >> switch (params_format(params)) { >> case SNDRV_PCM_FORMAT_S16_LE: >> >> > > That seems way more complicated then need be. It also uses 2 bits in the > mask whereas the original code only used 1, e.g., bit 5. Is that correct ? The BCLKDIV has three fields. The bitclock divider (bit 0-1), the samplerate (bit 2-4) and the source select (bit 5). Here we want to update the bitclock divider field and the samplerate field. When I wrote the code I was lazy and used ~ADAU1373_BCLKDIV_SOURCE as the mask, which for this register is functional equivalent to ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK. > > How about just open coding the macro thusly: > > #define ADAU1373_BCLKDIV_SOURCE (1<<5) /*BIT(5)*/ But now there is a inconsistency where we sometimes use (1<<x) and sometimes BIT(x). I prefer my solution. - Lars

1 0

Re: [alsa-devel] [PATCH] omap: convert to devm_ioremap_resource()
by Jarkko Nikula 11 Mar '13

11 Mar '13

Hi On 03/11/2013 05:58 PM, Silviu-Mihai Popescu wrote: > Convert all uses of devm_request_and_ioremap() to the newly introduced > devm_ioremap_resource() which provides more consistent error handling. > > devm_ioremap_resource() provides its own error messages so all explicit > error messages can be removed from the failure code paths. > > Signed-off-by: Silviu-Mihai Popescu <silviupopescu1990(a)gmail.com> > --- > sound/soc/omap/omap-dmic.c | 9 +++------ > sound/soc/omap/omap-mcpdm.c | 8 +++----- > 2 files changed, 6 insertions(+), 11 deletions(-) > Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com>

1 0

[alsa-devel] twl4030 and Puredata/jack
by Anders Alex 11 Mar '13

11 Mar '13

Ok, Let me start off that I know very little about audio and drivers. Recently I started looking in to running Puredata and possibly jack on the Pandora handheld. However I ran in to a number of problems getting these to interface with alsa. I should note that on the pandora we don't have too up to date versions of alsa and kernel so perhaps these are already resolved. In SL4P (slackware for pandora) I have: ALSA: 1.0.24 Kernel: 3.2.30 I will paste a couple of use-cases and their corresponding dmesg output: When I run ~$ pd -alsa I get: snd_pcm_hw_params (input): Invalid argument snd_pcm_hw_params (output): Invalid argument and: [201429.385345] omap_mcbsp_dai_hw_params: not enough bandwidth for desired rate and channels [201429.385375] asoc: interface omap-mcbsp-dai.1 hw params failed With 'jackd -d alsa -S' (need s16_le): loading driver .. creating alsa driver ... hw:0|hw:0|1024|2|48000|0|0|nomon|swmeter|-|16bit control device hw:0 configuring for 48000Hz, period = 1024 frames (21.3 ms), buffer = 2 periods ALSA: final selected sample format for capture: 16bit little-endian ALSA: use 2 periods for capture ALSA: cannot set hardware parameters for capture ALSA: cannot configure capture channel cannot load driver module alsa and: [201746.759857] asoc: can't set codec twl4030-hifi hw params The pandora kernel-dev explained that the hardware can only handle stereo streams. However someone in #jack explains it as the following: "This typically means that the ALSA driver is incomplete or buggy when used the way JACK does (poll+mmap) vs. the way aplay does (blocking read+write). So things that do stuff the "aplay way" will work; things that do stuff the JACK way will not work." I hope someone can shed some light on this and perhaps give some ideas to how to get this working. cheers, alexander

4 7

[alsa-devel] [PATCH v2] ALSA: ASoC: davinci-mcasp: clean up davinci_hw_common_param()
by Daniel Mack 11 Mar '13

11 Mar '13

As pointed of by Vaibhav, commit 2952b27e2 ("ASoC: davinci-mcasp: Add support for multichannel playback") duplicated the logic of counting the active serializers. That can be avoided by shifting the code around a bit. Also, drop two unused defines introduced by the same commit. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Michal Bachraty <michal.bachraty(a)streamunlimited.com> Cc: Vaibhav Bedia <vaibhav.bedia(a)ti.com> --- v2: fix typo in subject sound/soc/davinci/davinci-mcasp.c | 34 +++++++++++----------------------- 1 file changed, 11 insertions(+), 23 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index b386cad..8a9c0ee 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -237,8 +237,6 @@ #define RXSTATE BIT(5) #define SRMOD_MASK 3 #define SRMOD_INACTIVE 0 -#define SRMOD_TX 1 -#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -691,27 +689,6 @@ static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, } for (i = 0; i < dev->num_serializer; i++) { - if (dev->serial_dir[i] == TX_MODE) - tx_ser++; - if (dev->serial_dir[i] == RX_MODE) - rx_ser++; - } - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - ser = tx_ser; - else - ser = rx_ser; - - if (ser < max_active_serializers) { - dev_warn(dev->dev, "stream has more channels (%d) than are " - "enabled in mcasp (%d)\n", channels, ser * slots); - return -EINVAL; - } - - tx_ser = 0; - rx_ser = 0; - - for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); if (dev->serial_dir[i] == TX_MODE && @@ -730,6 +707,17 @@ static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, } } + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ser = tx_ser; + else + ser = rx_ser; + + if (ser < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + return -EINVAL; + } + if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { if (dev->txnumevt * tx_ser > 64) dev->txnumevt = 1; -- 1.8.1.4

1 0

Re: [alsa-devel] [PATCH 06/10] ARM: soc-audio phycore-ac97: fix driver init order
by Mark Brown 11 Mar '13

11 Mar '13

On Sun, Mar 10, 2013 at 07:33:07PM +0100, Markus Pargmann wrote: > Adding a soc-audio sound driver internally uses snd_soc_register_device, > which tries to bind it with the codec. The adding of the driver is > deferred because the codec driver is not loaded. > This patch changes the order of registration. The codec driver is there > before the ac97 driver, so the snd_soc_register_device call is > successfull. You shouldn't be using soc-audio in the first place...

1 0

[alsa-devel] [PATCH 0/7] TCO fixes for RME MADI/Raydat/AIO/AES cards
by Adrian Knoth 11 Mar '13

11 Mar '13

Hi! I've spent another day at IOSONO to work with them on QA patches for the RME MADI series, especially the optional TCO module (LTC synchronisation with video cameras). Here are our results, all tested for an entire week in their lab. Cheers Adrian Knoth (7): ALSA: hdspm - Refactor sample rate acquisition ALSA: hdspm - Allow the TCO and SYNC-IN to be used in slave mode ALSA: hdspm - Provide ALSA control to disable 96K frames ALSA: hdspm - Remove duplicate code from ALSA controls ALSA: hdspm - Also check for TCO sync states ALSA: hdspm - Add ALSA controls to read the TCO LTC state ALSA: hdspm - Enable new TCO ALSA controls sound/pci/rme9652/hdspm.c | 379 ++++++++++++++++++++++++++++----------------- 1 file changed, 237 insertions(+), 142 deletions(-) -- 1.7.10.4

2 10

[alsa-devel] Send to stable for "ALSA: hda - Move one-time init codes from generic_hdmi_init()"?
by David Henningsson 11 Mar '13

11 Mar '13

Hi Takashi, Should this patch be sent to stable? commit 8b8d654b55648561287bd8baca0f75f964a17038 Author: Takashi Iwai <tiwai(a)suse.de> Date: Wed Jun 20 16:32:22 2012 +0200 ALSA: hda - Move one-time init codes from generic_hdmi_init() I'm investigating a kernel panic under kernel 3.5, and I'm not 100% sure (still waiting for person with hw to report back), but the only reasonable explanation I can find is that there is some power management activity going while the repoll timer (for hdmi_present_sense) is running, and that generic_hdmi_init reinitializes the delayed_work struct, thus causing the panic when the timer has elapsed. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

2 1

Re: [alsa-devel] running 32 chan AD/DA on Linux
by Daniel Mack 11 Mar '13

11 Mar '13

Hi Igor, On 11.03.2013 09:56, Igor Levin wrote: > Thank you so much for a quick answer, are you in Europe by any chance? Yes, in Germany. > Could you give us a link where we can browse the source code on the Net, > or tell us where to get those files > mentioned as there seem to be so many Linux source files floating in the > web. Please forgive my ignorance > if this question seems so naive, I am really a hardware engineer :) There's Linus' repository which ends up in the official tarball releases on a regular basis. You can browse that tree online in its cgit: https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/… > Do you believe the code base as it stands will work with 32 channels > In/Out pipes or do we need to > get the unit to one of you guys to test it and make corrections as needed? I'm not aware of any sound specific limitations in that area. Just try it, and let us know how it goes. Sending the output of 'lsusb -v' for a descriptor dump is mandatory in order to understand how your device is designed. Thanks, Daniel

1 0

Re: [alsa-devel] [PATCH] ASoC: adau1373: adau1373_hw_params: Silence overflow warning
by Lars-Peter Clausen 11 Mar '13

11 Mar '13

On 03/10/2013 06:34 PM, Tim Gardner wrote: > ADAU1373_BCLKDIV_SOURCE is defined as BIT(5) which uses UL constants. On > amd64 the result of the ones complement operator is then truncated to > unsigned int according to the prototype of snd_soc_update_bits(). I think > gcc is correctly warning that the upper 32 bits are lost. > > sound/soc/codecs/adau1373.c: In function 'adau1373_hw_params': > sound/soc/codecs/adau1373.c:940:3: warning: large integer implicitly truncated to unsigned type [-Woverflow] > > gcc version 4.6.3 > > Cc: Lars-Peter Clausen <lars(a)metafoo.de> > Cc: Liam Girdwood <lgirdwood(a)gmail.com> > Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: device-drivers-devel(a)blackfin.uclinux.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> > --- > sound/soc/codecs/adau1373.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c > index 068b3ae..56ed788 100644 > --- a/sound/soc/codecs/adau1373.c > +++ b/sound/soc/codecs/adau1373.c > @@ -132,7 +132,7 @@ struct adau1373 { > #define ADAU1373_DAI_FORMAT_I2S 0x2 > #define ADAU1373_DAI_FORMAT_DSP 0x3 > > -#define ADAU1373_BCLKDIV_SOURCE BIT(5) > +#define ADAU1373_BCLKDIV_SOURCE (unsigned int)BIT(5) Hm, that's a bit ugly. How about the following instead: diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 068b3ae..1aa10dd 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -133,6 +133,8 @@ struct adau1373 { #define ADAU1373_DAI_FORMAT_DSP 0x3 #define ADAU1373_BCLKDIV_SOURCE BIT(5) +#define ADAU1373_BCLKDIV_SR_MASK (0x07 << 2) +#define ADAU1373_BCLKDIV_BCLK_MASK 0x03 #define ADAU1373_BCLKDIV_32 0x03 #define ADAU1373_BCLKDIV_64 0x02 #define ADAU1373_BCLKDIV_128 0x01 @@ -937,7 +939,8 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, adau1373_dai->enable_src = (div != 0); snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), - ~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64); + ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK, + (div << 2) | ADAU1373_BCLKDIV_64); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE:

1 0

[alsa-devel] snd_seq_timer_open() NULL pointer dereference
by Tommi Rantala 11 Mar '13

11 Mar '13

Hello, I'm hitting this while fuzzing the kernel with Trinity: [ 239.006244] BUG: unable to handle kernel NULL pointer dereference at 00000000000002ae [ 239.007011] IP: [<ffffffff819b3477>] snd_seq_timer_open+0xe7/0x130 [ 239.007011] PGD 785cd067 PUD 76964067 PMD 0 [ 239.007011] Oops: 0002 [#4] SMP [ 239.007011] CPU 0 [ 239.007011] Pid: 4288, comm: trinity-child7 Tainted: G D W 3.9.0-rc1+ #100 Bochs Bochs [ 239.007011] RIP: 0010:[<ffffffff819b3477>] [<ffffffff819b3477>] snd_seq_timer_open+0xe7/0x130 [ 239.007011] RSP: 0018:ffff88006ece7d38 EFLAGS: 00010246 [ 239.007011] RAX: 0000000000000286 RBX: ffff88007851b400 RCX: 0000000000000000 [ 239.007011] RDX: 000000000000ffff RSI: ffff88006ece7d58 RDI: ffff88006ece7d38 [ 239.007011] RBP: ffff88006ece7d98 R08: 000000000000000a R09: 000000000000fffe [ 239.007011] R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 [ 239.007011] R13: ffff8800792c5400 R14: 0000000000e8f000 R15: 0000000000000007 [ 239.007011] FS: 00007f7aaa650700(0000) GS:ffff88007f800000(0000) knlGS:0000000000000000 [ 239.007011] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 239.007011] CR2: 00000000000002ae CR3: 000000006efec000 CR4: 00000000000006f0 [ 239.007011] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 239.007011] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 [ 239.007011] Process trinity-child7 (pid: 4288, threadinfo ffff88006ece6000, task ffff880076a8a290) [ 239.007011] Stack: [ 239.007011] 0000000000000286 ffffffff828f2be0 ffff88006ece7d58 ffffffff810f354d [ 239.007011] 65636e6575716573 2065756575712072 ffff8800792c0030 0000000000000000 [ 239.007011] ffff88006ece7d98 ffff8800792c5400 ffff88007851b400 ffff8800792c5520 [ 239.007011] Call Trace: [ 239.007011] [<ffffffff810f354d>] ? trace_hardirqs_on+0xd/0x10 [ 239.007011] [<ffffffff819b17e9>] snd_seq_queue_timer_open+0x29/0x70 [ 239.007011] [<ffffffff819ae01a>] snd_seq_ioctl_set_queue_timer+0xda/0x120 [ 239.007011] [<ffffffff819acb9b>] snd_seq_do_ioctl+0x9b/0xd0 [ 239.007011] [<ffffffff819acbe0>] snd_seq_ioctl+0x10/0x20 [ 239.007011] [<ffffffff811b9542>] do_vfs_ioctl+0x522/0x570 [ 239.007011] [<ffffffff8130a4b3>] ? file_has_perm+0x83/0xa0 [ 239.007011] [<ffffffff810f354d>] ? trace_hardirqs_on+0xd/0x10 [ 239.007011] [<ffffffff811b95ed>] sys_ioctl+0x5d/0xa0 [ 239.007011] [<ffffffff813663fe>] ? trace_hardirqs_on_thunk+0x3a/0x3f [ 239.007011] [<ffffffff81faed69>] system_call_fastpath+0x16/0x1b [ 239.007011] Code: e8 ef b4 fd ff 85 c0 41 89 c4 79 18 44 89 e6 48 c7 c7 e0 f0 5f 82 31 c0 e8 e7 34 5e 00 eb 35 0f 1f 44 00 00 48 8b 45 a0 45 31 e4 <48> c7 40 28 e0 2e 9b 81 4c 89 68 38 83 48 10 08 48 89 43 50 eb [ 239.007011] RIP [<ffffffff819b3477>] snd_seq_timer_open+0xe7/0x130 [ 239.007011] RSP <ffff88006ece7d38> [ 239.007011] CR2: 00000000000002ae [ 239.055294] ---[ end trace bfb5b965ff1d73a3 ]--- Tommi

2 3

Re: [alsa-devel] running 32 chan AD/DA on Linux
by Daniel Mack 11 Mar '13

11 Mar '13

Hi Igor, please don't remove the alsa ML from Cc:. On 11.03.2013 09:37, Igor Levin wrote: [...] > But now we got a bit of a breather and our developers are finally able > to give Linux the attention it deserves > to get it going. I have couple fundamental question which you guys can > help us with: > > 1) Does Linux audio support the Implicit feedback mode where the Out > pipe will source the same amount of > samples as are coming into related IN pipe. Yes. > This is the sync mode > we use in Orion 32. If this is so, could > you kindly provide the file/line number in the source code where we > can see how it makes the packetising > decision. Please have a look at sound/usb/endpoint.c - this is where the packet logic lives in. Please read the comments in this file - the packet sending for this implicit feedback mode is implemented in queue_pending_output_urbs(). > 2) We are upgrading our Zodiac DAC to use asynchronous feedback mode. > What Async feedback mode formats > does Linux support? Where can we see it in the source? Linux supports feedback over dedicated isochronous endpoints as well. The format is auto-detected in snd_usb_handle_sync_urb(), also in sound/usb/endpoint.c, after each stream start. Internally, the driver uses Q16.16. Hope this helps. In case of more specific questions, please just let me know. Thanks, Daniel

1 0

Re: [alsa-devel] [PATCH][RESEND] Add a USB audio quirk for the NuForce UDH-100 device.
by Clemens Ladisch 11 Mar '13

11 Mar '13

David Helstroom wrote: > Interface 1 does not exist Then it doesn't need a quirk, does it? > and Interface 0 should be ignored. Why? If the driver doesn't like something in interface 0, that bug should be fixed. What is the output of "lsusb -v" for this device? Regards, Clemens

1 0

[alsa-devel] davici-mcasp: "tx-num-evt" confusion with number of serializers
by Michal Bachraty 11 Mar '13

11 Mar '13

Hi, I'm working on multichannel version of davinci-mcasp and also davinci-pcm. I have first version running and now I want to refine code. I found one confusion in davinci-mcasp with using of DT property "tx-num-evt". In DT binding documentation "tx-num-evt" is defined as "FIFO levels", but in Mcasp src, there is code, which mixes tx-num-evt with number of serializers (i2s data lines) that are enabled for data playback and receive (dev->txnumevt * tx_ser) mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, (((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); >From dacinci pcm, DMA data tranfer use txnumevt as number of serializers and also for data prefetching. I undestand definition "FIFO levels" as how many prefetched data are in FIFO. Prefetched data are for me frame data / 2 ( all left or right channels for one sampling time). This results me from AM335x FIFO documantation. What is original purpose for "tx-num-evt" parameter? Thanks, Michal

5 12

[alsa-devel] Add ALSA driver to Buildroot image or add / load it after booting the image
by Marc Schultz 11 Mar '13

11 Mar '13

Hey ALSA-Devel community, I'm successfully booting a Buildroot made Linux image on my board. This board has an Cirrus Logic CS4270 sound chip and now I want to activate the specific sound card driver for it. Is there a good way to put the CS4270 driver files into the Linux Kernel folders of Buildroot to get them compressed into the Linux image or do I have to go one of the following ways after booting the image on my target hardware? http://www.alsa-project.org/main/index.php/Matrix:Module-cs4270 http://www.alsa-project.org/main/index.php/Driver_Compilation) Do you have a good workflow for me to get this thing working? Best regards, Marc Schultz

2 1

Re: [alsa-devel] [PATCH] emu10k1: Fix coccicheck warning
by Clemens Ladisch 10 Mar '13

10 Mar '13

Vicentiu Ciorbaru wrote: > On Sun, Mar 10, 2013 at 6:32 PM, Clemens Ladisch <clemens(a)ladisch.de> wrote: >> Vicentiu Ciorbaru wrote: >>> Removed redundant cast of kmalloc return pointer. >> >>> - (icode->gpr_map = (u_int32_t __user *) >>> - kcalloc(512 + 256 + 256 + 2 * 1024, sizeof(u_int32_t), >>> - GFP_KERNEL)) == NULL || >>> + (icode->gpr_map = kcalloc(512 + 256 + 256 + 2 * 1024, >>> + sizeof(u_int32_t), GFP_KERNEL)) == NULL || >> >> Why would casting to __user be redundant? > > My reasoning behind the patch is that the C standard does not require a > specific cast of a void pointer in order to set it to the type of the > member gpr_map of the struct snd_emu10k1_fx8010_code. I named it redundant > because the compiler would cast to it anyway. That's correct as far as the C standard is concerned. However, __user is a non-standard extension. > The line also appears as a warning when running make coccicheck. And when you remove the __user, you get a warning when running make C=x. Both coccinelle and sparse are merely tools for finding likely bugs. When it isn't possible to make them both happy with or without the cast, that just shows that the code itself is badly designed. The driver should be redesigned to not access user and kernel space memory through the same pointer. Regards, Clemens

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Re: [alsa-devel] [PATCH 01/10] imx-ssi: Fix AC97 rates
by Fabio Estevam 10 Mar '13

10 Mar '13

Hi Markus, On Sun, Mar 10, 2013 at 3:33 PM, Markus Pargmann <mpa(a)pengutronix.de> wrote: > From: Sascha Hauer <s.hauer(a)pengutronix.de> > > This device supports multiple rates as described in later AC97 > standards. This patch allows playback of different sample frequencies > without conversion. > > Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> > Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> > --- > sound/soc/fsl/imx-ssi.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) Currently we have the following ssi drivers for imx: sound/soc/fsl/imx-ssi.c: for i.mx SoCs running on non-dt platform sound/soc/fsl/fsl_ssi.c: for PowerPC and i.mx SoCs running on dt platforms I see that in this series you convert sound/soc/fsl/imx-ssi.c into dt. Shouldn't we use sound/soc/fsl/fsl_ssi.c instead and maybe even get rid of sound/soc/fsl/imx-ssi.c when all imx become converted to dt?

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Re: [alsa-devel] [PATCH] emu10k1: Fix coccicheck warning
by Clemens Ladisch 10 Mar '13

10 Mar '13

Vicentiu Ciorbaru wrote: > Removed redundant cast of kmalloc return pointer. > - (icode->gpr_map = (u_int32_t __user *) > - kcalloc(512 + 256 + 256 + 2 * 1024, sizeof(u_int32_t), > - GFP_KERNEL)) == NULL || > + (icode->gpr_map = kcalloc(512 + 256 + 256 + 2 * 1024, > + sizeof(u_int32_t), GFP_KERNEL)) == NULL || Why would casting to __user be redundant? Regards, Clemens

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

10 Mar '13

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

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

09 Mar '13

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

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

08 Mar '13

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

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

08 Mar '13

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

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

08 Mar '13

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

2 1

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

08 Mar '13

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

2 2

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

08 Mar '13

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

2 1

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

08 Mar '13

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

4 10

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

07 Mar '13

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

4 44

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

07 Mar '13

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

1 0

[alsa-devel] ASoC updates for v3.9
by Mark Brown 07 Mar '13

07 Mar '13

The following changes since commit 6dbe51c251a327e012439c4772097a13df43c5b8: Linux 3.9-rc1 (2013-03-03 15:11:05 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.9-rc1 for you to fetch changes up to f73c06dc2cbf1ddf1c92f05f9843f923c8c69529: Merge remote-tracking branch 'asoc/fix/wm8960' into tmp (2013-03-07 14:29:43 +0800) ---------------------------------------------------------------- ASoC: Updates for v3.9 A few driver fixes, none of them terribly dramatic. ---------------------------------------------------------------- Axel Lin (1): ASoC: wm8350: Use jiffies rather than msecs in schedule_delayed_work() Charles Keepax (2): ASoC: wm5102: Correct OUT2 volume and switch names ASoC: wm5110: Correct OUT2/3 volume and switch names Mark Brown (7): ASoC: wm8960: Correct register 0 and 1 defaults ASoC: wm8960: Fix ADC power bits ASoC: wm5102: Apply a SYSCLK patch for later revs Merge remote-tracking branch 'asoc/fix/arizona' into tmp Merge remote-tracking branch 'asoc/fix/tegra' into tmp Merge remote-tracking branch 'asoc/fix/wm8350' into tmp Merge remote-tracking branch 'asoc/fix/wm8960' into tmp Stephen Warren (1): ASoC: tegra: fix I2S bit count mask sound/soc/codecs/wm5102.c | 15 +++++++++++++-- sound/soc/codecs/wm5110.c | 16 ++++++++-------- sound/soc/codecs/wm8350.c | 4 ++-- sound/soc/codecs/wm8960.c | 8 ++++---- sound/soc/tegra/tegra20_i2s.h | 2 +- sound/soc/tegra/tegra30_i2s.h | 2 +- 6 files changed, 29 insertions(+), 18 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Avoid division by zero in dspxfr_one_seg()
by Xi Wang 07 Mar '13

07 Mar '13

Move the zero check `hda_frame_size_words == 0' before the modulus `buffer_size_words % hda_frame_size_words'. Also remove the redundant null check `buffer_addx == NULL'. Signed-off-by: Xi Wang <xi.wang(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index db02c1e..eefc456 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2298,6 +2298,11 @@ static int dspxfr_one_seg(struct hda_codec *codec, hda_frame_size_words = ((sample_rate_div == 0) ? 0 : (num_chans * sample_rate_mul / sample_rate_div)); + if (hda_frame_size_words == 0) { + snd_printdd(KERN_ERR "frmsz zero\n"); + return -EINVAL; + } + buffer_size_words = min(buffer_size_words, (unsigned int)(UC_RANGE(chip_addx, 1) ? 65536 : 32768)); @@ -2308,8 +2313,7 @@ static int dspxfr_one_seg(struct hda_codec *codec, chip_addx, hda_frame_size_words, num_chans, sample_rate_mul, sample_rate_div, buffer_size_words); - if ((buffer_addx == NULL) || (hda_frame_size_words == 0) || - (buffer_size_words < hda_frame_size_words)) { + if (buffer_size_words < hda_frame_size_words) { snd_printdd(KERN_ERR "dspxfr_one_seg:failed\n"); return -EINVAL; } -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ALSA: hda - check NULL pointer when creating SPDIF PCM switch
by mengdong.lin＠intel.com 07 Mar '13

07 Mar '13

From: Mengdong Lin <mengdong.lin(a)intel.com> If the new control cannot be created, this function will return to avoid snd_hda_ctl_add dereferencing a NULL control pointer. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3dc6566..97c68dd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3433,11 +3433,16 @@ static struct snd_kcontrol_new spdif_share_sw = { int snd_hda_create_spdif_share_sw(struct hda_codec *codec, struct hda_multi_out *mout) { + struct snd_kcontrol *kctl; + if (!mout->dig_out_nid) return 0; + + kctl = snd_ctl_new1(&spdif_share_sw, mout); + if (!kctl) + return -ENOMEM; /* ATTENTION: here mout is passed as private_data, instead of codec */ - return snd_hda_ctl_add(codec, mout->dig_out_nid, - snd_ctl_new1(&spdif_share_sw, mout)); + return snd_hda_ctl_add(codec, mout->dig_out_nid, kctl); } EXPORT_SYMBOL_HDA(snd_hda_create_spdif_share_sw); -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: samsung: let device core setup the default pin configuration
by Thomas Abraham 07 Mar '13

07 Mar '13

With device core now able to setup the default pin configuration, the pin configuration code based on the deprecated Samsung specific gpio bindings in i2s driver is removed. Signed-off-by: Thomas Abraham <thomas.abraham(a)linaro.org> --- .../devicetree/bindings/sound/samsung-i2s.txt | 30 +-------- sound/soc/samsung/i2s.c | 67 +------------------ 2 files changed, 8 insertions(+), 89 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt index 3070046..1db65e5 100644 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt @@ -20,23 +20,11 @@ Optional SoC Specific Properties: then this flag is enabled. - samsung,idma-addr: Internal DMA register base address of the audio sub system(used in secondary sound source). - -Required Board Specific Properties: - -- gpios: The gpio specifier for data out,data in, LRCLK, CDCLK and SCLK - interface lines. The format of the gpio specifier depends on the gpio - controller. - The syntax of samsung gpio specifier is - <[phandle of the gpio controller node] - [pin number within the gpio controller] - [mux function] - [flags and pull up/down] - [drive strength]> +- pinctrl-0: Should specify pin control groups used for this controller. +- pinctrl-names: Should contain only one value - "default". Example: -- SoC Specific Portion: - i2s@03830000 { compatible = "samsung,i2s-v5"; reg = <0x03830000 0x100>; @@ -48,16 +36,6 @@ i2s@03830000 { samsung,supports-rstclr; samsung,supports-secdai; samsung,idma-addr = <0x03000000>; -}; - -- Board Specific Portion: - -i2s@03830000 { - gpios = <&gpz 0 2 0 0>, /* I2S_0_SCLK */ - <&gpz 1 2 0 0>, /* I2S_0_CDCLK */ - <&gpz 2 2 0 0>, /* I2S_0_LRCK */ - <&gpz 3 2 0 0>, /* I2S_0_SDI */ - <&gpz 4 2 0 0>, /* I2S_0_SDO[1] */ - <&gpz 5 2 0 0>, /* I2S_0_SDO[2] */ - <&gpz 6 2 0 0>; /* I2S_0_SDO[3] */ + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; }; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d7231e3..f982a83 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -16,7 +16,6 @@ #include <linux/io.h> #include <linux/module.h> #include <linux/of.h> -#include <linux/of_gpio.h> #include <linux/pm_runtime.h> #include <sound/soc.h> @@ -79,7 +78,6 @@ struct i2s_dai { u32 suspend_i2smod; u32 suspend_i2scon; u32 suspend_i2spsr; - unsigned long gpios[7]; /* i2s gpio line numbers */ }; /* Lock for cross i/f checks */ @@ -1009,52 +1007,6 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) return i2s; } -#ifdef CONFIG_OF -static int samsung_i2s_parse_dt_gpio(struct i2s_dai *i2s) -{ - struct device *dev = &i2s->pdev->dev; - int index, gpio, ret; - - for (index = 0; index < 7; index++) { - gpio = of_get_gpio(dev->of_node, index); - if (!gpio_is_valid(gpio)) { - dev_err(dev, "invalid gpio[%d]: %d\n", index, gpio); - goto free_gpio; - } - - ret = gpio_request(gpio, dev_name(dev)); - if (ret) { - dev_err(dev, "gpio [%d] request failed\n", gpio); - goto free_gpio; - } - i2s->gpios[index] = gpio; - } - return 0; - -free_gpio: - while (--index >= 0) - gpio_free(i2s->gpios[index]); - return -EINVAL; -} - -static void samsung_i2s_dt_gpio_free(struct i2s_dai *i2s) -{ - unsigned int index; - for (index = 0; index < 7; index++) - gpio_free(i2s->gpios[index]); -} -#else -static int samsung_i2s_parse_dt_gpio(struct i2s_dai *dai) -{ - return -EINVAL; -} - -static void samsung_i2s_dt_gpio_free(struct i2s_dai *dai) -{ -} - -#endif - static const struct of_device_id exynos_i2s_match[]; static inline int samsung_i2s_get_driver_data(struct platform_device *pdev) @@ -1223,18 +1175,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->sec_dai = sec_dai; } - if (np) { - if (samsung_i2s_parse_dt_gpio(pri_dai)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; - } - } else { - if (i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; - } + if (i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err; } snd_soc_register_dai(&pri_dai->pdev->dev, &pri_dai->i2s_dai_drv); @@ -1259,9 +1203,6 @@ static int samsung_i2s_remove(struct platform_device *pdev) i2s = dev_get_drvdata(&pdev->dev); other = i2s->pri_dai ? : i2s->sec_dai; - if (!i2s_pdata->cfg_gpio && pdev->dev.of_node) - samsung_i2s_dt_gpio_free(i2s->pri_dai); - if (other) { other->pri_dai = NULL; other->sec_dai = NULL; -- 1.6.6.rc2

3 2

[alsa-devel] [PATCH] ALSA: hda - check NULL pointer when creating SPDIF controls
by mengdong.lin＠intel.com 07 Mar '13

07 Mar '13

From: Mengdong Lin <mengdong.lin(a)intel.com> If the SPDIF control array cannot be reallocated, the function will return to avoid dereferencing a NULL pointer. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 04b5738..3dc6566 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3334,6 +3334,8 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec, return -EBUSY; } spdif = snd_array_new(&codec->spdif_out); + if (!spdif) + return -ENOMEM; for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); if (!kctl) -- 1.7.10.4

2 1

[alsa-devel] [PATCH v2] ALSA: ASoC: cs4271: convert to direct regmap API usage
by Daniel Mack 07 Mar '13

07 Mar '13

By using the regmap API directly, we can make use of the .write_flag_mask for SPI, which allows us to drop the strange register hacks that were necessary so far. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- v2: add a read_flag_mask to SPI regmap config sound/soc/codecs/cs4271.c | 157 +++++++++++++++++++++++++++------------------- 1 file changed, 91 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 2415a41..6d77a1e 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -42,14 +42,14 @@ * High byte represents SPI chip address (0x10) + write command (0) * Low byte - codec register address */ -#define CS4271_MODE1 0x2001 /* Mode Control 1 */ -#define CS4271_DACCTL 0x2002 /* DAC Control */ -#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */ -#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */ -#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */ -#define CS4271_ADCCTL 0x2006 /* ADC Control */ -#define CS4271_MODE2 0x2007 /* Mode Control 2 */ -#define CS4271_CHIPID 0x2008 /* Chip ID */ +#define CS4271_MODE1 0x01 /* Mode Control 1 */ +#define CS4271_DACCTL 0x02 /* DAC Control */ +#define CS4271_DACVOL 0x03 /* DAC Volume & Mixing Control */ +#define CS4271_VOLA 0x04 /* DAC Channel A Volume Control */ +#define CS4271_VOLB 0x05 /* DAC Channel B Volume Control */ +#define CS4271_ADCCTL 0x06 /* ADC Control */ +#define CS4271_MODE2 0x07 /* Mode Control 2 */ +#define CS4271_CHIPID 0x08 /* Chip ID */ #define CS4271_FIRSTREG CS4271_MODE1 #define CS4271_LASTREG CS4271_MODE2 @@ -144,23 +144,27 @@ * Array do not include Chip ID, as codec driver does not use * registers read operations at all */ -static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { - 0, - 0, - CS4271_DACCTL_AMUTE, - CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, - 0, - 0, - 0, - 0, +static const struct reg_default cs4271_reg_defaults[] = { + { CS4271_MODE1, 0, }, + { CS4271_DACCTL, CS4271_DACCTL_AMUTE, }, + { CS4271_DACVOL, CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, }, + { CS4271_VOLA, 0, }, + { CS4271_VOLB, 0, }, + { CS4271_ADCCTL, 0, }, + { CS4271_MODE2, 0, }, }; +static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == CS4271_CHIPID; +} + struct cs4271_private { /* SND_SOC_I2C or SND_SOC_SPI */ - enum snd_soc_control_type bus_type; unsigned int mclk; bool master; bool deemph; + struct regmap *regmap; /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ @@ -210,14 +214,14 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_LEFT_J: val |= CS4271_MODE1_DAC_DIF_LJ; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); if (ret < 0) return ret; break; case SND_SOC_DAIFMT_I2S: val |= CS4271_MODE1_DAC_DIF_I2S; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); if (ret < 0) return ret; @@ -227,7 +231,7 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); if (ret < 0) return ret; @@ -252,7 +256,7 @@ static int cs4271_set_deemph(struct snd_soc_codec *codec) val <<= 4; } - ret = snd_soc_update_bits(codec, CS4271_DACCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_DACCTL, CS4271_DACCTL_DEM_MASK, val); if (ret < 0) return ret; @@ -341,14 +345,14 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, !dai->capture_active) || (substream->stream == SNDRV_PCM_STREAM_CAPTURE && !dai->playback_active)) { - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, + CS4271_MODE2_PDN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; } @@ -378,7 +382,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, val |= cs4271_clk_tab[i].ratio_mask; - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); if (ret < 0) return ret; @@ -389,6 +393,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); int ret; int val_a = 0; int val_b = 0; @@ -398,10 +403,13 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) val_b = CS4271_VOLB_MUTE; } - ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLA, + CS4271_VOLA_MUTE, val_a); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLB, + CS4271_VOLB_MUTE, val_b); if (ret < 0) return ret; @@ -463,25 +471,33 @@ static struct snd_soc_dai_driver cs4271_dai = { static int cs4271_soc_suspend(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Set power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, CS4271_MODE2_PDN); if (ret < 0) return ret; + return 0; } static int cs4271_soc_resume(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Restore codec state */ - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(cs4271->regmap); if (ret < 0) return ret; + /* then disable the power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; + return 0; } #else @@ -542,40 +558,22 @@ static int cs4271_probe(struct snd_soc_codec *codec) cs4271->gpio_nreset = gpio_nreset; - /* - * In case of I2C, chip address specified in board data. - * So cache IO operations use 8 bit codec register address. - * In case of SPI, chip address and register address - * passed together as 16 bit value. - * Anyway, register address is masked with 0xFF inside - * soc-cache code. - */ - if (cs4271->bus_type == SND_SOC_SPI) - ret = snd_soc_codec_set_cache_io(codec, 16, 8, - cs4271->bus_type); - else - ret = snd_soc_codec_set_cache_io(codec, 8, 8, - cs4271->bus_type); - if (ret) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; /* Power-up sequence requires 85 uS */ udelay(85); if (amutec_eq_bmutec) - snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_MUTECAEQUB, - CS4271_MODE2_MUTECAEQUB); + regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_MUTECAEQUB, + CS4271_MODE2_MUTECAEQUB); return snd_soc_add_codec_controls(codec, cs4271_snd_controls, ARRAY_SIZE(cs4271_snd_controls)); @@ -597,13 +595,24 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .remove = cs4271_remove, .suspend = cs4271_soc_suspend, .resume = cs4271_soc_resume, - .reg_cache_default = cs4271_dflt_reg, - .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg), - .reg_word_size = sizeof(cs4271_dflt_reg[0]), - .compress_type = SND_SOC_FLAT_COMPRESSION, }; #if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config cs4271_spi_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = CS4271_LASTREG, + .read_flag_mask = 0x21, + .write_flag_mask = 0x20, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_spi_probe(struct spi_device *spi) { struct cs4271_private *cs4271; @@ -613,7 +622,9 @@ static int cs4271_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, cs4271); - cs4271->bus_type = SND_SOC_SPI; + cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); @@ -643,6 +654,18 @@ static const struct i2c_device_id cs4271_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); +static const struct regmap_config cs4271_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = CS4271_LASTREG, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -653,7 +676,9 @@ static int cs4271_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, cs4271); - cs4271->bus_type = SND_SOC_I2C; + cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); -- 1.8.1.4

1 0

[alsa-devel] [PATCH 1/2] ALSA: ASoC: ak4104: convert to direct regmap API usage
by Daniel Mack 07 Mar '13

07 Mar '13

Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/ak4104.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 6f6c335..58f390d 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -55,6 +55,7 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { struct snd_soc_codec *codec = codec_dai->codec; + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); int val = 0; int ret; @@ -77,9 +78,9 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) return -EINVAL; - ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1, - val); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1, + val); if (ret < 0) return ret; @@ -91,11 +92,12 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); int val = 0; /* set the IEC958 bits: consumer mode, no copyright bit */ val |= IEC958_AES0_CON_NOT_COPYRIGHT; - snd_soc_write(codec, AK4104_REG_CHN_STATUS(0), val); + regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(0), val); val = 0; @@ -132,7 +134,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return snd_soc_write(codec, AK4104_REG_CHN_STATUS(3), val); + return regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(3), val); } static const struct snd_soc_dai_ops ak4101_dai_ops = { @@ -160,20 +162,17 @@ static int ak4104_probe(struct snd_soc_codec *codec) int ret; codec->control_data = ak4104->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) - return ret; /* set power-up and non-reset bits */ - ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); if (ret < 0) return ret; /* enable transmitter */ - ret = snd_soc_update_bits(codec, AK4104_REG_TX, - AK4104_TX_TXE, AK4104_TX_TXE); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX, + AK4104_TX_TXE, AK4104_TX_TXE); if (ret < 0) return ret; @@ -182,8 +181,10 @@ static int ak4104_probe(struct snd_soc_codec *codec) static int ak4104_remove(struct snd_soc_codec *codec) { - snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0); + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0); return 0; } -- 1.8.1.4

2 2

[alsa-devel] Creative CA0132 1102:0012 (rev 01)
by Stefan de Konink 07 Mar '13

07 Mar '13

I've been the proud owner of a Creative PCIe soundcard, sadly getting anything else but static noise out of it while playing a file seems impossible. After the device is closed the noise stops again. The mixer does function in all aspects, muting, volume etc. HDAAnalyzer shows that I should get output from CA0132 Analog, I have validated this by manipulating the PIN Widget Controls. Is there any documentation available on this specific device? It seemed that some DSP work was done recently by Creative. I wouldn't mind to do some hacking myself, but it is hard for me to believe that a driver was posted in the 1.0.25 release that didn't have any normal functioning analogue output. Stefan 03:00.0 0403: 1102:0012 (rev 01) Subsystem: 1102:0013 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 18 Region 0: Memory at f7104000 (64-bit, non-prefetchable) [size=16K] Region 2: Memory at f7100000 (64-bit, non-prefetchable) [size=16K] Capabilities: [40] Power Management version 3 Flags: PMEClk- DSI- D1- D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst+ PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] MSI: Enable- Count=1/1 Maskable+ 64bit+ Address: 0000000000000000 Data: 0000 Masking: 00000000 Pending: 00000000 Capabilities: [70] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 512 bytes, PhantFunc 0, Latency L0s <1us, L1 <8us ExtTag- AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 512 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Latency L0 <512ns, L1 <16us ClockPM- Surprise- LLActRep- BwNot- LnkCtl: ASPM L0s Enabled; RCB 64 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- DevCap2: Completion Timeout: Not Supported, TimeoutDis+, LTR-, OBFF Not Supported DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis-, LTR-, OBFF Disabled LnkCtl2: Target Link Speed: 2.5GT/s, EnterCompliance- SpeedDis- Transmit Margin: Normal Operating Range, EnterModifiedCompliance- ComplianceSOS- Compliance De-emphasis: -6dB LnkSta2: Current De-emphasis Level: -6dB, EqualizationComplete-, EqualizationPhase1- EqualizationPhase2-, EqualizationPhase3-, LinkEqualizationRequest- Capabilities: [100 v1] Advanced Error Reporting UESta: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UEMsk: DLP- SDES- TLP- FCP- CmpltTO- CmpltAbrt- UnxCmplt- RxOF- MalfTLP- ECRC- UnsupReq- ACSViol- UESvrt: DLP+ SDES+ TLP- FCP+ CmpltTO- CmpltAbrt- UnxCmplt- RxOF+ MalfTLP+ ECRC- UnsupReq- ACSViol- CESta: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr- CEMsk: RxErr- BadTLP- BadDLLP- Rollover- Timeout- NonFatalErr+ AERCap: First Error Pointer: 00, GenCap+ CGenEn- ChkCap+ ChkEn- Capabilities: [140 v1] Virtual Channel Caps: LPEVC=0 RefClk=100ns PATEntryBits=1 Arb: Fixed- WRR32- WRR64- WRR128- Ctrl: ArbSelect=Fixed Status: InProgress- VC0: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable+ ID=0 ArbSelect=Fixed TC/VC=01 Status: NegoPending- InProgress- VC1: Caps: PATOffset=00 MaxTimeSlots=1 RejSnoopTrans- Arb: Fixed- WRR32- WRR64- WRR128- TWRR128- WRR256- Ctrl: Enable- ID=1 ArbSelect=Fixed TC/VC=00 Status: NegoPending- InProgress- Capabilities: [170 v1] Device Serial Number 00-00-00-00-00-00-00-00 Capabilities: [180 v1] Power Budgeting <?> Kernel driver in use: snd_hda_intel snd 49214 11 snd_hwdep,snd_timer,snd_pcm,snd_hda_codec,snd_hda_intel,snd_hda_codec_ca0132

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[alsa-devel] [PATCH] ALSA: ASoC: cs4271: convert to direct regmap API usage
by Daniel Mack 06 Mar '13

06 Mar '13

By using the regmap API directly, we can make use of the .write_flag_mask for SPI, which allows us to drop the strange register hacks that were necessary so far. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- I could only test this on I2C, so if anyone with appropriate hardware would give it a try on SPI, that would be good. Thanks, Daniel sound/soc/codecs/cs4271.c | 156 ++++++++++++++++++++++++++-------------------- 1 file changed, 90 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 2415a41..b6d670f 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -42,14 +42,14 @@ * High byte represents SPI chip address (0x10) + write command (0) * Low byte - codec register address */ -#define CS4271_MODE1 0x2001 /* Mode Control 1 */ -#define CS4271_DACCTL 0x2002 /* DAC Control */ -#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */ -#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */ -#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */ -#define CS4271_ADCCTL 0x2006 /* ADC Control */ -#define CS4271_MODE2 0x2007 /* Mode Control 2 */ -#define CS4271_CHIPID 0x2008 /* Chip ID */ +#define CS4271_MODE1 0x01 /* Mode Control 1 */ +#define CS4271_DACCTL 0x02 /* DAC Control */ +#define CS4271_DACVOL 0x03 /* DAC Volume & Mixing Control */ +#define CS4271_VOLA 0x04 /* DAC Channel A Volume Control */ +#define CS4271_VOLB 0x05 /* DAC Channel B Volume Control */ +#define CS4271_ADCCTL 0x06 /* ADC Control */ +#define CS4271_MODE2 0x07 /* Mode Control 2 */ +#define CS4271_CHIPID 0x08 /* Chip ID */ #define CS4271_FIRSTREG CS4271_MODE1 #define CS4271_LASTREG CS4271_MODE2 @@ -144,23 +144,27 @@ * Array do not include Chip ID, as codec driver does not use * registers read operations at all */ -static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { - 0, - 0, - CS4271_DACCTL_AMUTE, - CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, - 0, - 0, - 0, - 0, +static const struct reg_default cs4271_reg_defaults[] = { + { CS4271_MODE1, 0, }, + { CS4271_DACCTL, CS4271_DACCTL_AMUTE, }, + { CS4271_DACVOL, CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, }, + { CS4271_VOLA, 0, }, + { CS4271_VOLB, 0, }, + { CS4271_ADCCTL, 0, }, + { CS4271_MODE2, 0, }, }; +static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == CS4271_CHIPID; +} + struct cs4271_private { /* SND_SOC_I2C or SND_SOC_SPI */ - enum snd_soc_control_type bus_type; unsigned int mclk; bool master; bool deemph; + struct regmap *regmap; /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ @@ -210,14 +214,14 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_LEFT_J: val |= CS4271_MODE1_DAC_DIF_LJ; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); if (ret < 0) return ret; break; case SND_SOC_DAIFMT_I2S: val |= CS4271_MODE1_DAC_DIF_I2S; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); if (ret < 0) return ret; @@ -227,7 +231,7 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); if (ret < 0) return ret; @@ -252,7 +256,7 @@ static int cs4271_set_deemph(struct snd_soc_codec *codec) val <<= 4; } - ret = snd_soc_update_bits(codec, CS4271_DACCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_DACCTL, CS4271_DACCTL_DEM_MASK, val); if (ret < 0) return ret; @@ -341,14 +345,14 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, !dai->capture_active) || (substream->stream == SNDRV_PCM_STREAM_CAPTURE && !dai->playback_active)) { - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, + CS4271_MODE2_PDN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; } @@ -378,7 +382,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, val |= cs4271_clk_tab[i].ratio_mask; - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); if (ret < 0) return ret; @@ -389,6 +393,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); int ret; int val_a = 0; int val_b = 0; @@ -398,10 +403,13 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) val_b = CS4271_VOLB_MUTE; } - ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLA, + CS4271_VOLA_MUTE, val_a); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLB, + CS4271_VOLB_MUTE, val_b); if (ret < 0) return ret; @@ -463,25 +471,33 @@ static struct snd_soc_dai_driver cs4271_dai = { static int cs4271_soc_suspend(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Set power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, CS4271_MODE2_PDN); if (ret < 0) return ret; + return 0; } static int cs4271_soc_resume(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Restore codec state */ - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(cs4271->regmap); if (ret < 0) return ret; + /* then disable the power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; + return 0; } #else @@ -542,40 +558,22 @@ static int cs4271_probe(struct snd_soc_codec *codec) cs4271->gpio_nreset = gpio_nreset; - /* - * In case of I2C, chip address specified in board data. - * So cache IO operations use 8 bit codec register address. - * In case of SPI, chip address and register address - * passed together as 16 bit value. - * Anyway, register address is masked with 0xFF inside - * soc-cache code. - */ - if (cs4271->bus_type == SND_SOC_SPI) - ret = snd_soc_codec_set_cache_io(codec, 16, 8, - cs4271->bus_type); - else - ret = snd_soc_codec_set_cache_io(codec, 8, 8, - cs4271->bus_type); - if (ret) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; /* Power-up sequence requires 85 uS */ udelay(85); if (amutec_eq_bmutec) - snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_MUTECAEQUB, - CS4271_MODE2_MUTECAEQUB); + regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_MUTECAEQUB, + CS4271_MODE2_MUTECAEQUB); return snd_soc_add_codec_controls(codec, cs4271_snd_controls, ARRAY_SIZE(cs4271_snd_controls)); @@ -597,13 +595,23 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .remove = cs4271_remove, .suspend = cs4271_soc_suspend, .resume = cs4271_soc_resume, - .reg_cache_default = cs4271_dflt_reg, - .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg), - .reg_word_size = sizeof(cs4271_dflt_reg[0]), - .compress_type = SND_SOC_FLAT_COMPRESSION, }; #if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config cs4271_spi_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = CS4271_LASTREG, + .write_flag_mask = 0x20, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_spi_probe(struct spi_device *spi) { struct cs4271_private *cs4271; @@ -613,7 +621,9 @@ static int cs4271_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, cs4271); - cs4271->bus_type = SND_SOC_SPI; + cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); @@ -643,6 +653,18 @@ static const struct i2c_device_id cs4271_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); +static const struct regmap_config cs4271_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = CS4271_LASTREG, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -653,7 +675,9 @@ static int cs4271_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, cs4271); - cs4271->bus_type = SND_SOC_I2C; + cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); -- 1.8.1.4

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[alsa-devel] [PATCH] Make snd_BUG_ON() always evaluate and return the conditional expression.
by Christine Spang 06 Mar '13

06 Mar '13

Having snd_BUG_ON() only evaluate its conditional when CONFIG_SND_DEBUG is set leads to frequent bugs, since other similar macros in the kernel have different behavior. Let's make snd_BUG_ON() act like those macros so it will stop being accidentally misused. Signed-off-by: Christine Spang <christine.spang(a)oracle.com> --- Documentation/DocBook/writing-an-alsa-driver.tmpl | 12 +++++------- include/sound/core.h | 24 ++++++++--------------- 2 files changed, 13 insertions(+), 23 deletions(-) diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl index bd6fee2..06741e9 100644 --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -6164,14 +6164,12 @@ struct _snd_pcm_runtime { <para> The macro takes an conditional expression to evaluate. - When <constant>CONFIG_SND_DEBUG</constant>, is set, the - expression is actually evaluated. If it's non-zero, it shows - the warning message such as + When <constant>CONFIG_SND_DEBUG</constant>, is set, if the + expression is non-zero, it shows the warning message such as <computeroutput>BUG? (xxx)</computeroutput> - normally followed by stack trace. It returns the evaluated - value. - When no <constant>CONFIG_SND_DEBUG</constant> is set, this - macro always returns zero. + normally followed by stack trace. + + In both cases it returns the evaluated value. </para> </section> diff --git a/include/sound/core.h b/include/sound/core.h index 7cede2d..a63680b 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -379,18 +379,10 @@ void __snd_printk(unsigned int level, const char *file, int line, * snd_BUG_ON - debugging check macro * @cond: condition to evaluate * - * When CONFIG_SND_DEBUG is set, this macro evaluates the given condition, - * and call WARN() and returns the value if it's non-zero. - * - * When CONFIG_SND_DEBUG is not set, this just returns zero, and the given - * condition is ignored. - * - * NOTE: the argument won't be evaluated at all when CONFIG_SND_DEBUG=n. - * Thus, don't put any statement that influences on the code behavior, - * such as pre/post increment, to the argument of this macro. - * If you want to evaluate and give a warning, use standard WARN_ON(). + * Has the same behavior as WARN_ON when CONFIG_SND_DEBUG is set, + * otherwise just evaluates the conditional and returns the value. */ -#define snd_BUG_ON(cond) WARN((cond), "BUG? (%s)\n", __stringify(cond)) +#define snd_BUG_ON(cond) WARN_ON((cond)) #else /* !CONFIG_SND_DEBUG */ @@ -400,11 +392,11 @@ __printf(2, 3) static inline void _snd_printd(int level, const char *format, ...) {} #define snd_BUG() do { } while (0) -static inline int __snd_bug_on(int cond) -{ - return 0; -} -#define snd_BUG_ON(cond) __snd_bug_on(0 && (cond)) /* always false */ + +#define snd_BUG_ON(condition) ({ \ + int __ret_warn_on = !!(condition); \ + unlikely(__ret_warn_on); \ +}) #endif /* CONFIG_SND_DEBUG */ -- 1.8.2.rc0

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Re: [alsa-devel] [PATCH] ASoC: samsung: let device core setup the default pin configuration
by Mark Brown 06 Mar '13

06 Mar '13

On Wed, Mar 06, 2013 at 12:42:09PM +0100, Heiko Stübner wrote: > What is going away is the deprecated non-pinctrl way to setup the pins on dt > platforms. OK, can we have a changelog improvement please? I didn't read the patch properly because the changelog said it would break my primary development system.

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Re: [alsa-devel] ALSA: usb: Work around CM6631 sample
by Demian Martin 06 Mar '13

06 Mar '13

Date: Tue, 5 Mar 2013 23:03:03 +0100 From: Torstein Hegge <hegge(a)resisty.net> To: alsa-devel(a)alsa-project.org Subject: [alsa-devel] [PATCH v3] ALSA: usb: Work around CM6631 sample rate change bug Message-ID: <20130305220303.GD20417(a)pvv.ntnu.no> Content-Type: text/plain; charset=utf-8 The C-Media CM6631 USB-to-S/PDIF receiver doesn't respond to changes in sample rate while the interface is active. Reset the interface after setting the sampling frequency on sample rate changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous stream, causing distorted sound on sample rate changes. The reset is performed for all current C-Media UAC V2.0 devices, including CM6610, CM6620 and CM6631, but has only been tested with CM6631. Signed-off-by: Torstein Hegge <hegge(a)resisty.net> I met with the CMedia team late last year. They told me that the chips are different and they were showing me a CM6631a that has some key differences from the CM6631. The two are pin compatible. I got the impression that the driver requirements are different. I asked about Linux support and what the problem was. First, on UAC2 Their software wizard dug deep into the problem with their CM6631 and ALSA. He concluded that there is a sequence problem between setting sample rate and sending the "play' command. Something about UAC1 method not being the same as UAC2. He indicated that the ALSA driver handles this differently from the OSX and their Windows drivers. I may well have this description confused. In any case he has a revised firmware for the new CM6631A chip that addresses this in ALSA. I asked them to contribute the change but they seemed to get distracted with other projects. Needless to say for them the volume on these chips has been small. I just requested that they contribute to this effort again. I don't know if it will happen. Demian

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[alsa-devel] [PATCH 4/4] ASoC: arizona: Optimise FLL loop gains
by Mark Brown 06 Mar '13

06 Mar '13

For optimal performance the FLL loop gain should be adjusted depending on the frequency of the input clock for the loop. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 43 +++++++++++++++++++++++++++++++++++++++---- 1 file changed, 39 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 41c116f..29dd641 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1175,6 +1175,16 @@ static struct { { 1000000, 13500000, 0, 1 }, }; +static struct { + unsigned int min; + unsigned int max; + u16 gain; +} fll_gains[] = { + { 0, 256000, 0 }, + { 256000, 1000000, 2 }, + { 1000000, 13500000, 4 }, +}; + struct arizona_fll_cfg { int n; int theta; @@ -1182,6 +1192,7 @@ struct arizona_fll_cfg { int refdiv; int outdiv; int fratio; + int gain; }; static int arizona_calc_fll(struct arizona_fll *fll, @@ -1241,6 +1252,18 @@ static int arizona_calc_fll(struct arizona_fll *fll, return -EINVAL; } + for (i = 0; i < ARRAY_SIZE(fll_gains); i++) { + if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) { + cfg->gain = fll_gains[i].gain; + break; + } + } + if (i == ARRAY_SIZE(fll_gains)) { + arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n", + Fref); + return -EINVAL; + } + cfg->n = target / (ratio * Fref); if (target % (ratio * Fref)) { @@ -1268,13 +1291,15 @@ static int arizona_calc_fll(struct arizona_fll *fll, cfg->n, cfg->theta, cfg->lambda); arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv); + arizona_fll_dbg(fll, "GAIN=%d\n", cfg->gain); return 0; } static void arizona_apply_fll(struct arizona *arizona, unsigned int base, - struct arizona_fll_cfg *cfg, int source) + struct arizona_fll_cfg *cfg, int source, + bool sync) { regmap_update_bits(arizona->regmap, base + 3, ARIZONA_FLL1_THETA_MASK, cfg->theta); @@ -1289,6 +1314,15 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base, cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); + if (sync) + regmap_update_bits(arizona->regmap, base + 0x7, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + else + regmap_update_bits(arizona->regmap, base + 0x9, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + regmap_update_bits(arizona->regmap, base + 2, ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, ARIZONA_FLL1_CTRL_UPD | cfg->n); @@ -1326,17 +1360,18 @@ static void arizona_enable_fll(struct arizona_fll *fll, ARIZONA_FLL1_OUTDIV_MASK, ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); - arizona_apply_fll(arizona, fll->base, ref, fll->ref_src); + arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, + false); if (fll->sync_src >= 0) arizona_apply_fll(arizona, fll->base + 0x10, sync, - fll->sync_src); + fll->sync_src, true); } else if (fll->sync_src >= 0) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, sync, - fll->sync_src); + fll->sync_src, false); } else { arizona_fll_err(fll, "No clocks provided\n"); return; -- 1.7.10.4

1 0

[alsa-devel] [PATCH 3/4] ASoC: arizona: Increase FLL synchroniser bandwidth for high frequencies
by Mark Brown 06 Mar '13

06 Mar '13

If we are using a high freqency SYNCCLK then increasing the bandwidth of the synchroniser improves performance. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 81da6ef..41c116f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1342,6 +1342,17 @@ static void arizona_enable_fll(struct arizona_fll *fll, return; } + /* + * Increase the bandwidth if we're not using a low frequency + * sync source. + */ + if (fll->sync_src >= 0 && fll->sync_freq > 100000) + regmap_update_bits(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, 0); + else + regmap_update_bits(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, ARIZONA_FLL1_SYNC_BW); + if (!arizona_is_enabled_fll(fll)) pm_runtime_get(arizona->dev); -- 1.7.10.4

1 0

[alsa-devel] [PATCH 2/4] mfd: wm5102: Map in additional FLL control registers
by Mark Brown 06 Mar '13

06 Mar '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5102-tables.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/drivers/mfd/wm5102-tables.c b/drivers/mfd/wm5102-tables.c index f3c2e9a..768fd42 100644 --- a/drivers/mfd/wm5102-tables.c +++ b/drivers/mfd/wm5102-tables.c @@ -291,12 +291,14 @@ static const struct reg_default wm5102_reg_default[] = { { 0x00000176, 0x0000 }, /* R374 - FLL1 Control 6 */ { 0x00000177, 0x0181 }, /* R375 - FLL1 Loop Filter Test 1 */ { 0x00000178, 0x0000 }, /* R376 - FLL1 NCO Test 0 */ + { 0x00000179, 0x0000 }, /* R377 - FLL1 Control 7 */ { 0x00000181, 0x0000 }, /* R385 - FLL1 Synchroniser 1 */ { 0x00000182, 0x0000 }, /* R386 - FLL1 Synchroniser 2 */ { 0x00000183, 0x0000 }, /* R387 - FLL1 Synchroniser 3 */ { 0x00000184, 0x0000 }, /* R388 - FLL1 Synchroniser 4 */ { 0x00000185, 0x0000 }, /* R389 - FLL1 Synchroniser 5 */ { 0x00000186, 0x0000 }, /* R390 - FLL1 Synchroniser 6 */ + { 0x00000187, 0x0001 }, /* R391 - FLL1 Synchroniser 7 */ { 0x00000189, 0x0000 }, /* R393 - FLL1 Spread Spectrum */ { 0x0000018A, 0x0004 }, /* R394 - FLL1 GPIO Clock */ { 0x00000191, 0x0000 }, /* R401 - FLL2 Control 1 */ @@ -307,12 +309,14 @@ static const struct reg_default wm5102_reg_default[] = { { 0x00000196, 0x0000 }, /* R406 - FLL2 Control 6 */ { 0x00000197, 0x0000 }, /* R407 - FLL2 Loop Filter Test 1 */ { 0x00000198, 0x0000 }, /* R408 - FLL2 NCO Test 0 */ + { 0x00000199, 0x0000 }, /* R409 - FLL2 Control 7 */ { 0x000001A1, 0x0000 }, /* R417 - FLL2 Synchroniser 1 */ { 0x000001A2, 0x0000 }, /* R418 - FLL2 Synchroniser 2 */ { 0x000001A3, 0x0000 }, /* R419 - FLL2 Synchroniser 3 */ { 0x000001A4, 0x0000 }, /* R420 - FLL2 Synchroniser 4 */ { 0x000001A5, 0x0000 }, /* R421 - FLL2 Synchroniser 5 */ { 0x000001A6, 0x0000 }, /* R422 - FLL2 Synchroniser 6 */ + { 0x000001A7, 0x0001 }, /* R423 - FLL2 Synchroniser 7 */ { 0x000001A9, 0x0000 }, /* R425 - FLL2 Spread Spectrum */ { 0x000001AA, 0x0004 }, /* R426 - FLL2 GPIO Clock */ { 0x00000200, 0x0006 }, /* R512 - Mic Charge Pump 1 */ @@ -1056,12 +1060,14 @@ static bool wm5102_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL1_CONTROL_6: case ARIZONA_FLL1_LOOP_FILTER_TEST_1: case ARIZONA_FLL1_NCO_TEST_0: + case ARIZONA_FLL1_CONTROL_7: case ARIZONA_FLL1_SYNCHRONISER_1: case ARIZONA_FLL1_SYNCHRONISER_2: case ARIZONA_FLL1_SYNCHRONISER_3: case ARIZONA_FLL1_SYNCHRONISER_4: case ARIZONA_FLL1_SYNCHRONISER_5: case ARIZONA_FLL1_SYNCHRONISER_6: + case ARIZONA_FLL1_SYNCHRONISER_7: case ARIZONA_FLL1_SPREAD_SPECTRUM: case ARIZONA_FLL1_GPIO_CLOCK: case ARIZONA_FLL2_CONTROL_1: @@ -1072,12 +1078,14 @@ static bool wm5102_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL2_CONTROL_6: case ARIZONA_FLL2_LOOP_FILTER_TEST_1: case ARIZONA_FLL2_NCO_TEST_0: + case ARIZONA_FLL2_CONTROL_7: case ARIZONA_FLL2_SYNCHRONISER_1: case ARIZONA_FLL2_SYNCHRONISER_2: case ARIZONA_FLL2_SYNCHRONISER_3: case ARIZONA_FLL2_SYNCHRONISER_4: case ARIZONA_FLL2_SYNCHRONISER_5: case ARIZONA_FLL2_SYNCHRONISER_6: + case ARIZONA_FLL2_SYNCHRONISER_7: case ARIZONA_FLL2_SPREAD_SPECTRUM: case ARIZONA_FLL2_GPIO_CLOCK: case ARIZONA_MIC_CHARGE_PUMP_1: -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm5102: Apply a SYSCLK patch for later revs
by Mark Brown 06 Mar '13

06 Mar '13

Evaluation has identified some performance improvements to the device. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index deb1097..cef288c 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -573,6 +573,13 @@ static const struct reg_default wm5102_sysclk_reva_patch[] = { { 0x025e, 0x0112 }, }; +static const struct reg_default wm5102_sysclk_revb_patch[] = { + { 0x3081, 0x08FE }, + { 0x3083, 0x00ED }, + { 0x30C1, 0x08FE }, + { 0x30C3, 0x00ED }, +}; + static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -587,6 +594,10 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, patch = wm5102_sysclk_reva_patch; patch_size = ARRAY_SIZE(wm5102_sysclk_reva_patch); break; + default: + patch = wm5102_sysclk_revb_patch; + patch_size = ARRAY_SIZE(wm5102_sysclk_revb_patch); + break; } switch (event) { -- 1.7.10.4

1 0

Re: [alsa-devel] [PATCH v3 09/12] ASoC: dmaengine_pcm: add snd_dmaengine_generic_pcm_open()
by Shawn Guo 06 Mar '13

06 Mar '13

On Wed, Mar 06, 2013 at 12:28:18AM +0100, Marek Vasut wrote: > Hi Shawn, > > > With generic DMA device tree binding and helper function > > dma_request_slave_channel() in place, dmaengine_pcm should support > > that in requesting DMA channel for users that support generic DMA > > device tree binding. > > > > Add a new API snd_dmaengine_generic_pcm_open() as the variant of > > snd_dmaengine_pcm_open(). It takes DMA client struct device pointer > > and slave channel name as arguments and calls generic DMA helper > > dma_request_slave_channel() to request DMA channel from dmaengine. > > > > Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> > > Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > Cc: alsa-devel(a)alsa-project.org > > --- > > Changes since v2: > > - Drop 'const' from argument 'name' to fix the warning below. > > > [...] > > v3 changelog missing ;-) What did you adjust here please? The "Changes since v2" means v3 changelog. Shawn

1 0

[alsa-devel] [PATCH] ASoC: cs42l73: If Internal MCLK is >= 6.4MHz, then set SCLK to 64*Fs.
by Paul Handrigan 06 Mar '13

06 Mar '13

Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- sound/soc/codecs/cs42l73.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 6361dab..3b20c86 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1180,7 +1180,11 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, priv->config[id].mmcc &= 0xC0; priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; priv->config[id].spc &= 0xFC; - priv->config[id].spc |= MCK_SCLK_MCLK; + /* Use SCLK=64*Fs if internal MCLK >= 6.4MHz */ + if (priv->mclk >= 6400000) + priv->config[id].spc |= MCK_SCLK_64FS; + else + priv->config[id].spc |= MCK_SCLK_MCLK; } else { /* CS42L73 Slave */ priv->config[id].spc &= 0xFC; -- 1.7.0.4

3 2

[alsa-devel] [PATCH] ASoC: omap3pandora: Fix compilation error
by Peter Ujfalusi 06 Mar '13

06 Mar '13

Fixes: sound/soc/omap/omap3pandora.c: In function ‘omap3pandora_dac_event’: sound/soc/omap/omap3pandora.c:92:19: error: request for member ‘dev’ in something not a structure or union make[3]: *** [sound/soc/omap/omap3pandora.o] Error 1 Which is introduced by: dd194b4 ASoC: omap: Check regulator enable for DAC on Pandora Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap3pandora.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 10ced9d..9e46e1d 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -89,7 +89,7 @@ static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, if (SND_SOC_DAPM_EVENT_ON(event)) { ret = regulator_enable(omap3pandora_dac_reg); if (ret) { - dev_err(w->dapm.dev, "Failed to power DAC: %d\n", ret); + dev_err(w->dapm->dev, "Failed to power DAC: %d\n", ret); return ret; } mdelay(1); -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ALSA: usb: Work around CM6631 sample rate change bug
by Torstein Hegge 05 Mar '13

05 Mar '13

The C-Media CM6631 USB-to-S/PDIF receiver doesn't respond to changes in sampling frequency while the interface is active. Reset the interface after setting the sampling frequency, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous file, causing distorted sound on sample rate changes. Signed-off-by: Torstein Hegge <hegge(a)resisty.net> --- sound/usb/pcm.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d82e378..01978a6 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -710,6 +710,16 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->need_setup_ep = false; } + /* Some devices doesn't respond to sample rate changes while the + * interface is active. */ + switch (subs->stream->chip->usb_id) { + case USB_ID(0x0d8c, 0x0304): /* C-Media - Schiit USB Interface */ + case USB_ID(0x0d8c, 0x0309): /* C-Media CM6631 */ + usb_set_interface(subs->dev, subs->cur_audiofmt->iface, + subs->cur_audiofmt->altset_idx); + break; + } + /* some unit conversions in runtime */ subs->data_endpoint->maxframesize = bytes_to_frames(runtime, subs->data_endpoint->maxpacksize); -- 1.7.10.4

4 11

Re: [alsa-devel] common way for hardware metering and monitoring between kernel/user land
by Clemens Ladisch 05 Mar '13

05 Mar '13

Takashi Sakamoto wrote: > Now i'm working in kernel land to support Echo's Fireworks (TM) devices > with better integration. > [...] > Well, now I need your advice about common ways for hardware metering and > monitoring between kernel land and user land. For almost all FireWire audio devices, it would not be possible to create ALSA mixer controls that would allow to control the device in a sensible way with a generic mixer application like alsamixer. (Things like matrix mixers, or meters integrated with corresponding volume sliders.) Therefore, custom mixer applications are required. Because devices typically have most of their customizations in the way in which they handle (mixer) controls, the FFADO guys and me have agreed that the kernel driver handles only stuff required for audio streaming; all other controls are accessed *only* from user space by going directly through /dev/fw* (or through a hwdep device if the kernel driver and userspace tools need to share a communication channel). So for now, there's no integration needed in the kernel driver. (I won't have time to test your driver before the next weekend.) Regards, Clemens

1 0

[alsa-devel] ASoC: no-pcm (backend) error propagation
by Frkuska, Joshua 05 Mar '13

05 Mar '13

Hello, I am working off of the 3.5.7 branch of the kernel using a frontend-backend approach for my driver. No additional patches have been added for ASoC/ALSA from the 3.5.7 kernel branch. My question relates to the way the no-pcm (backend) propagates errors to the front end. My FE(frontend) consists of a cpu-dai, a pcm platform driver, and a dummy-codec. My BE(backend) consists of another cpu-dai, a non-pcm platform driver, and a real codec. The non-pcm operates with the dmaengine but just in an autonomous fashion external to the PCM's knowledge. I have the pcm-platform driver passing data to the FE cpu-dai and the backend platform then takes the data from the frontend's cpu-dai and passes it on to the BE cpu-dai, which then configures the codec-dai to get output. In the frontend if/whenever there is an underrun error in the cpu-dai, I can directly call snd_pcm_stop to stop the stream and report the error state. I can do this because the substream->ops have properly been defined in the frontend pcm. However, in the backend, the substream->ops never get initialized and rightfully so since there isn't a valid pcm for the backend. My question is, what is the correct/any way to let the frontend PCM stream know that an underrun has occurred in the backend? Is there a mechanism to handle this in ASoC? Looking through the code, I cannot find such a mechanism. I have a feeling that there should be something like snd_soc_pcm_stop to check to see if we are in a FE/BE configuration and act accordingly to stop the FE PCM stream and report errors. Again, perhaps such a mechanism exists and if so any help would be appreciated. Thanks for your time and any comments/suggestions, Joshua

4 10

[alsa-devel] am335x: mcasp in DIT mode
by Yegor Yefremov 05 Mar '13

05 Mar '13

I've solved the problem with DIT/SPDIF mode (see the issue description here: http://e2e.ti.com/support/arm/sitara_arm/f/791/p/247447/870030.aspx). In davinci_mcasp_hw_params() the DIT or I2S params will be set in the beginning. DIT mode configures the DAVINCI_MCASP_TXMASK_REG and DAVINCI_MCASP_TXFMT_REG. And here comes the problem: at the end of davinci_mcasp_hw_params() the davinci_config_channel_size() will touch the same registers again and thus overwrite the settings necessary for DIT. After I commented this routine I got the sound over S/PDIF and sii9022a HDMI transmitter and I could see the proper bits appearing on my oscilloscope. What were the best way to solve this problem? 1. execute davinci_config_channel_size() only if not in DIT mode? 2. for DIT only change the DAVINCI_MCASP_TXMASK_REG according to channel width? 3. execute if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); else davinci_hw_param(dev, substream->stream); after davinci_config_channel_size() Yegor

3 11

[alsa-devel] [PATCH v2] ALSA: usb: Work around CM6631 sample rate change bug
by Torstein Hegge 05 Mar '13

05 Mar '13

The C-Media CM6631 USB-to-S/PDIF receiver doesn't respond to changes in sample rate while the interface is active. Reset the interface after setting the sampling frequency on sample rate changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous stream, causing distorted sound on sample rate changes. Signed-off-by: Torstein Hegge <hegge(a)resisty.net> --- sound/usb/clock.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 5e634a2..d12a9c4 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -255,6 +255,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, unsigned char data[4]; int err, crate; int clock = snd_usb_clock_find_source(chip, fmt->clock); + unsigned int previous_rate; if (clock < 0) return clock; @@ -266,6 +267,18 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return -ENXIO; } + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data))) < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", + dev->devnum, iface, fmt->altsetting); + return err; + } + + previous_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; @@ -294,6 +307,18 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, if (crate != rate) snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + /* Some devices doesn't respond to sample rate changes while the + * interface is active. */ + if (crate != previous_rate) { + switch (chip->usb_id) { + case USB_ID(0x0d8c, 0x0304): /* C-Media - Schiit USB */ + case USB_ID(0x0d8c, 0x0309): /* C-Media CM6631 */ + usb_set_interface(dev, iface, 0); + usb_set_interface(dev, iface, fmt->altsetting); + break; + } + } + return 0; } -- 1.7.10.4

2 2

[alsa-devel] HDA Beep Device for Asus EeeBox EB1501P
by Harald Becker 05 Mar '13

05 Mar '13

Hi, I own an Asus EeeBox EB1501P with an ALC662 codec and had no HDA mixer channel for the speaker beep device. After digging somewhat in the kernel source I found that I only need to add a single line in the beep_white_list in patch_realtek.c. May be this entry can be included in the mainstream drivers: SND_PCI_QUIRK(0x1043, 0x8376, "EeePC", 1), After recompiling the driver with this entry, the mixer device for the beep channel appears and the digital beep sound is emitted on the connected speakers. ... with one problem: If I use echo -e "\a" in a shell or run the beep command utility the sound appears only on second and following invocations. Whenever the frequency changes with the beep command the first invocation stays silent or just produce some click due to ALSA driver power saving. On all further invocations with same frequency the requested tone is generated on external speakers. There is no difference in behavior if the commands are called in a terminal emulator or directly on a virtual text console. Does this happen only on my machine or is this a known problem? Is there anything I can do to fix this behavior? The kernel I used for testing is 3.8.0 from Gentoo Sources. Feel free to ask for any information you may need for debugging. Thx for any help ahead Harald

2 5

Re: [alsa-devel] [PATCH 1/5 v7][RFC] ASoC: add snd_soc_register_cpu()
by Mark Brown 05 Mar '13

05 Mar '13

On Mon, Mar 04, 2013 at 07:48:36PM +0000, Liam Girdwood wrote: > I think we probably have more in common between platforms and codecs now > than we have that is different (certainly, things were very different in > the past). But you are right, is it really worth the big churn..... Definitely, I'm just not sure that the existing platform object really captures the new bits or if it maps more onto DAIs.

1 0

[alsa-devel] [PATCH V2] ASoC: tegra: fix I2S bit count mask
by Stephen Warren 05 Mar '13

05 Mar '13

From: Stephen Warren <swarren(a)nvidia.com> This register field is 11 bits wide, not 15 bits wide. Given the way this value is currently, used, this patch has no practical effect. However, it's still best if the value is correct. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Apply fix to Tegra20 as well as Tegra30. --- sound/soc/tegra/tegra20_i2s.h | 2 +- sound/soc/tegra/tegra30_i2s.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h index c27069d..7299587 100644 --- a/sound/soc/tegra/tegra20_i2s.h +++ b/sound/soc/tegra/tegra20_i2s.h @@ -121,7 +121,7 @@ #define TEGRA20_I2S_TIMING_NON_SYM_ENABLE (1 << 12) #define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7ff #define TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) /* Fields in TEGRA20_I2S_FIFO_SCR */ diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index 34dc47b..a294d94 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -110,7 +110,7 @@ #define TEGRA30_I2S_TIMING_NON_SYM_ENABLE (1 << 12) #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7ff #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) /* Fields in TEGRA30_I2S_OFFSET */ -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: arizona: If we only have a clock to synchronise with make it REFCLK
by Mark Brown 05 Mar '13

05 Mar '13

If there is only one clock active the FLL should use REFCLK rather than SYNCCLK as the clock to synchronise with since REFCLK is always required. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 76 ++++++++++++++++++++++---------------------- 1 file changed, 38 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e456cb4..0599ff8 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1132,14 +1132,30 @@ static void arizona_enable_fll(struct arizona_fll *fll, struct arizona *arizona = fll->arizona; int ret; - regmap_update_bits(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); - - arizona_apply_fll(arizona, fll->base, ref, fll->ref_src); - if (fll->sync_src >= 0) - arizona_apply_fll(arizona, fll->base + 0x10, sync, + /* + * If we have both REFCLK and SYNCCLK then enable both, + * otherwise apply the SYNCCLK settings to REFCLK. + */ + if (fll->ref_src >= 0 && fll->ref_src != fll->sync_src) { + regmap_update_bits(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + + arizona_apply_fll(arizona, fll->base, ref, fll->ref_src); + if (fll->sync_src >= 0) + arizona_apply_fll(arizona, fll->base + 0x10, sync, + fll->sync_src); + } else if (fll->sync_src >= 0) { + regmap_update_bits(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + + arizona_apply_fll(arizona, fll->base, sync, fll->sync_src); + } else { + arizona_fll_err(fll, "No clocks provided\n"); + return; + } if (!arizona_is_enabled_fll(fll)) pm_runtime_get(arizona->dev); @@ -1149,7 +1165,8 @@ static void arizona_enable_fll(struct arizona_fll *fll, regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); - if (fll->sync_src >= 0) + if (fll->ref_src >= 0 && fll->sync_src >= 0 && + fll->ref_src != fll->sync_src) regmap_update_bits(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, ARIZONA_FLL1_SYNC_ENA); @@ -1180,9 +1197,6 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, struct arizona_fll_cfg ref, sync; int ret; - if (source < 0) - return -EINVAL; - if (fll->ref_src == source && fll->ref_freq == Fref) return 0; @@ -1216,39 +1230,25 @@ int arizona_set_fll(struct arizona_fll *fll, int source, struct arizona_fll_cfg ref, sync; int ret; - if (fll->ref_src < 0 || fll->ref_src == source) { - if (fll->sync_src == ARIZONA_FLL_SRC_NONE && - fll->ref_src == source && fll->ref_freq == Fref && - fll->fout == Fout) - return 0; - - if (Fout) { - ret = arizona_calc_fll(fll, &ref, Fref, Fout); - if (ret != 0) - return ret; - } - - fll->sync_src = ARIZONA_FLL_SRC_NONE; - fll->ref_src = source; - fll->ref_freq = Fref; - } else { - if (fll->sync_src == source && - fll->sync_freq == Fref && fll->fout == Fout) - return 0; - - if (Fout) { - ret = arizona_calc_fll(fll, &ref, fll->ref_freq, Fout); - if (ret != 0) - return ret; + if (fll->sync_src == source && + fll->sync_freq == Fref && fll->fout == Fout) + return 0; - ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (Fout) { + if (fll->ref_src >= 0) { + ret = arizona_calc_fll(fll, &ref, fll->ref_freq, + Fout); if (ret != 0) return ret; } - fll->sync_src = source; - fll->sync_freq = Fref; + ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (ret != 0) + return ret; } + + fll->sync_src = source; + fll->sync_freq = Fref; fll->fout = Fout; if (Fout) { -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: tegra: fix I2S bit count mask
by Stephen Warren 04 Mar '13

04 Mar '13

From: Stephen Warren <swarren(a)nvidia.com> This register field is 11 bits wide, not 15 bits wide. Given the way this value is currently, used, this patch has no practical effect. However, it's still best if the value is correct. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra30_i2s.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index 34dc47b..a294d94 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -110,7 +110,7 @@ #define TEGRA30_I2S_TIMING_NON_SYM_ENABLE (1 << 12) #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 -#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7ff #define TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA30_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) /* Fields in TEGRA30_I2S_OFFSET */ -- 1.7.10.4

1 0

[alsa-devel] [PATCH] hdajackretask: Some more UI fixes
by David Henningsson 04 Mar '13

04 Mar '13

- don't overwrite a set advanced override unless necessary - fix "not present" never selected - remove some trailing whitespaces Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hdajackretask/main-gtk.c | 116 ++++++++++++++++++++++--------------- hdajackretask/sysfs-pin-configs.c | 4 +- 2 files changed, 71 insertions(+), 49 deletions(-) diff --git a/hdajackretask/main-gtk.c b/hdajackretask/main-gtk.c index 4227669..9101af4 100644 --- a/hdajackretask/main-gtk.c +++ b/hdajackretask/main-gtk.c @@ -5,19 +5,22 @@ #include <string.h> #include <stdlib.h> #include <gtk/gtk.h> +#include <stdbool.h> #include "sysfs-pin-configs.h" #include "apply-changes.h" +typedef struct ui_data_t ui_data_t; + typedef struct pin_ui_data_t { pin_configs_t* pin_config; typical_pins_t pins_info[32]; GtkWidget *frame, *override, *jacktype; - GtkWidget* free_override_cb[FREE_OVERRIDES_COUNT]; + ui_data_t* owner; } pin_ui_data_t; -typedef struct ui_data_t { +struct ui_data_t { GList* pin_ui_data; GtkWidget *main_window; GtkWidget *content_scroll_widget; @@ -25,22 +28,35 @@ typedef struct ui_data_t { GtkWidget *codec_selection_combo; codec_name_t* current_codec; - int sysfs_pincount; + int sysfs_pincount; codec_name_t sysfs_codec_names[128]; pin_configs_t sysfs_pins[32]; gboolean free_overrides; gboolean trust_codec; gboolean trust_defcfg; gboolean model_auto; -} ui_data_t; +}; -static void override_toggled(GtkWidget* sender, pin_ui_data_t* data) +static void update_user_pin_config(ui_data_t* ui, pin_configs_t* cfg); + +static void update_override_sensitive(GtkWidget* sender, pin_ui_data_t* data) { int i; gboolean checked = gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(sender)); gtk_widget_set_sensitive(data->jacktype, checked); for (i = 0; i < FREE_OVERRIDES_COUNT; i++) - gtk_widget_set_sensitive(data->free_override_cb[i], checked); + gtk_widget_set_sensitive(data->free_override_cb[i], checked); +} + +static void override_toggled(GtkWidget* sender, pin_ui_data_t* data) +{ + update_override_sensitive(sender, data); + update_user_pin_config(data->owner, data->pin_config); +} + +static void jacktype_changed(GtkWidget* sender, pin_ui_data_t* data) +{ + update_user_pin_config(data->owner, data->pin_config); } static GtkWidget* create_pin_ui(ui_data_t* ui, pin_configs_t* pin_cfg) @@ -56,7 +72,8 @@ static GtkWidget* create_pin_ui(ui_data_t* ui, pin_configs_t* pin_cfg) data = calloc(1, sizeof(pin_ui_data_t)); data->pin_config = pin_cfg; - + data->owner = ui; + { /* Frame */ gchar* d = get_config_description(pin_cfg->init_pin_config); gchar* c = g_strdup_printf("Pin ID: 0x%02x", pin_cfg->nid); @@ -93,6 +110,7 @@ static GtkWidget* create_pin_ui(ui_data_t* ui, pin_configs_t* pin_cfg) gtk_combo_box_set_active(GTK_COMBO_BOX(jacktype), index); gtk_toggle_button_set_active(GTK_TOGGLE_BUTTON(override), pin_cfg->user_override); g_signal_connect(override, "toggled", G_CALLBACK(override_toggled), data); + g_signal_connect(jacktype, "changed", G_CALLBACK(jacktype_changed), data); gtk_container_add(box, override); gtk_container_add(GTK_CONTAINER(jacktype_box), jacktype); @@ -125,6 +143,7 @@ static GtkWidget* create_pin_ui(ui_data_t* ui, pin_configs_t* pin_cfg) } if (index >= 0) gtk_combo_box_set_active(GTK_COMBO_BOX(data->free_override_cb[i]), index); + g_signal_connect(data->free_override_cb[i], "changed", G_CALLBACK(jacktype_changed), data); } gtk_grid_attach(grid, gtk_label_new("Connectivity"), 0, 0, 1, 1); @@ -148,7 +167,7 @@ static GtkWidget* create_pin_ui(ui_data_t* ui, pin_configs_t* pin_cfg) if (ui->free_overrides) gtk_container_add(box, GTK_WIDGET(grid)); } - override_toggled(data->override, data); + update_override_sensitive(data->override, data); gtk_container_add(GTK_CONTAINER(result), GTK_WIDGET(box)); ui->pin_ui_data = g_list_prepend(ui->pin_ui_data, data); @@ -166,45 +185,48 @@ static void free_pin_ui_data(pin_ui_data_t* data) static gint pin_config_find(pin_ui_data_t* pin_ui, pin_configs_t* cfg) { - return pin_ui->pin_config == cfg ? 0 : 1; + return pin_ui->pin_config == cfg ? 0 : 1; } -static void update_user_pin_config(ui_data_t* ui) +static void update_user_pin_config(ui_data_t* ui, pin_configs_t* cfg) { - int i; - for (i = 0; i < ui->sysfs_pincount; i++) { - pin_ui_data_t* pin_ui; - GList *pos = g_list_find_custom(ui->pin_ui_data, &ui->sysfs_pins[i], (GCompareFunc) pin_config_find); - ui->sysfs_pins[i].user_override = FALSE; - if (!pos) - continue; - pin_ui = pos->data; - if (!gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(pin_ui->override))) - continue; + pin_ui_data_t* pin_ui; + GList *pos = g_list_find_custom(ui->pin_ui_data, cfg, (GCompareFunc) pin_config_find); + cfg->user_override = FALSE; + if (!pos) + return; + pin_ui = pos->data; + if (!gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(pin_ui->override))) + return; - if (ui->free_overrides) { - int j; - int index; - unsigned long val = 0; - for (j = 0; j < FREE_OVERRIDES_COUNT; j++) { - index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->free_override_cb[j])); - if (index < 0) - break; - val += get_free_override_list(j)[index].value; - } + if (ui->free_overrides) { + int j; + int index; + unsigned long val = 0; + for (j = 0; j < FREE_OVERRIDES_COUNT; j++) { + index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->free_override_cb[j])); if (index < 0) - continue; - ui->sysfs_pins[i].user_pin_config = val; - } else { - int index; - index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->jacktype)); - if (index < 0) - continue; - ui->sysfs_pins[i].user_pin_config = pin_ui->pins_info[index].pin_set; + break; + val += get_free_override_list(j)[index].value; } - ui->sysfs_pins[i].user_override = TRUE; + if (index < 0) + return; + cfg->user_pin_config = val; + } else { + int index; + index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->jacktype)); + if (index < 0) + return; + cfg->user_pin_config = pin_ui->pins_info[index].pin_set; } + cfg->user_override = TRUE; +} +static void update_all_user_pin_config(ui_data_t* ui) +{ + int i; + for (i = 0; i < ui->sysfs_pincount; i++) + update_user_pin_config(ui, &ui->sysfs_pins[i]); } static gboolean validate_user_pin_config(ui_data_t* ui, GError** err) @@ -215,7 +237,7 @@ static gboolean validate_user_pin_config(ui_data_t* ui, GError** err) g_set_error(err, 0, 0, "You must first select a codec!"); return FALSE; } - update_user_pin_config(ui); + update_all_user_pin_config(ui); if (ui->free_overrides) return TRUE; @@ -325,7 +347,7 @@ static void resize_main_window(ui_data_t* ui) } } -static void update_codec_ui(ui_data_t* ui) +static void update_codec_ui(ui_data_t* ui, bool codec_change) { int codec_index = gtk_combo_box_get_active(GTK_COMBO_BOX(ui->codec_selection_combo)); int i; @@ -337,7 +359,8 @@ static void update_codec_ui(ui_data_t* ui) if (codec_index < 0) return; ui->current_codec = &ui->sysfs_codec_names[codec_index]; - ui->sysfs_pincount = get_pin_configs_list(ui->sysfs_pins, 32, ui->current_codec->card, ui->current_codec->device); + if (codec_change) + ui->sysfs_pincount = get_pin_configs_list(ui->sysfs_pins, 32, ui->current_codec->card, ui->current_codec->device); for (i = 0; i < ui->sysfs_pincount; i++) { GtkWidget *w = create_pin_ui(ui, &ui->sysfs_pins[i]); if (w) @@ -350,14 +373,14 @@ static void update_codec_ui(ui_data_t* ui) static void codec_selected(GtkComboBox* combo, gpointer user_data) { - update_codec_ui(user_data); + update_codec_ui(user_data, true); } static void showallpins_toggled(GtkWidget* sender, ui_data_t* ui_data) { gboolean checked = gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(sender)); ui_data->trust_defcfg = !checked; - update_codec_ui(ui_data); + update_codec_ui(ui_data, false); } static void automodel_toggled(GtkWidget* sender, ui_data_t* ui_data) @@ -367,9 +390,8 @@ static void automodel_toggled(GtkWidget* sender, ui_data_t* ui_data) static void free_override_toggled(GtkWidget* sender, ui_data_t* ui_data) { - update_user_pin_config(ui_data); ui_data->free_overrides = gtk_toggle_button_get_active(GTK_TOGGLE_BUTTON(sender)); - update_codec_ui(ui_data); + update_codec_ui(ui_data, false); } static const char* readme_text = @@ -517,7 +539,7 @@ int main(int argc, char *argv[]) ui = create_ui(); gtk_widget_show_all(ui->main_window); if (ui->codec_selection_combo) - update_codec_ui(ui); + update_codec_ui(ui, true); gtk_main(); return 0; } diff --git a/hdajackretask/sysfs-pin-configs.c b/hdajackretask/sysfs-pin-configs.c index cd2897e..4044989 100644 --- a/hdajackretask/sysfs-pin-configs.c +++ b/hdajackretask/sysfs-pin-configs.c @@ -460,7 +460,7 @@ static free_override_t channel_arr[] = { static free_override_t* type_order[FREE_OVERRIDES_COUNT] = { - pc_arr, location_arr, device_arr, jack_arr, + pc_arr, location_arr, device_arr, jack_arr, color_arr, no_presence_arr, group_nr_arr, channel_arr }; @@ -472,7 +472,7 @@ unsigned long get_free_override_mask(int type) AC_DEFCFG_DEVICE, AC_DEFCFG_CONN_TYPE, AC_DEFCFG_COLOR, - AC_DEFCFG_MISC & 1, + AC_DEFCFG_MISC & (AC_DEFCFG_MISC_NO_PRESENCE << AC_DEFCFG_MISC_SHIFT), AC_DEFCFG_DEF_ASSOC, AC_DEFCFG_SEQUENCE, }; -- 1.7.9.5

2 1

[alsa-devel] [patch] [ALSA] seq_oss_event: missing range checks
by Dan Carpenter 04 Mar '13

04 Mar '13

The "dev" variable could be out of bounds. Calling snd_seq_oss_synth_is_valid() checks that it is is a valid device which has been opened. We check this inside set_note_event() so this function can't succeed without a valid "dev". But we need to do the check earlier to prevent invalid dereferences and memory corruption. One call tree where "dev" could be out of bounds is: -> snd_seq_oss_oob_user() -> snd_seq_oss_process_event() -> extended_event() -> note_on_event() Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c index 066f5f3..c390886 100644 --- a/sound/core/seq/oss/seq_oss_event.c +++ b/sound/core/seq/oss/seq_oss_event.c @@ -285,7 +285,12 @@ local_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev static int note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev) { - struct seq_oss_synthinfo *info = &dp->synths[dev]; + struct seq_oss_synthinfo *info; + + if (!snd_seq_oss_synth_is_valid(dp, dev)) + return -ENXIO; + + info = &dp->synths[dev]; switch (info->arg.event_passing) { case SNDRV_SEQ_OSS_PROCESS_EVENTS: if (! info->ch || ch < 0 || ch >= info->nr_voices) { @@ -340,7 +345,12 @@ note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, st static int note_off_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev) { - struct seq_oss_synthinfo *info = &dp->synths[dev]; + struct seq_oss_synthinfo *info; + + if (!snd_seq_oss_synth_is_valid(dp, dev)) + return -ENXIO; + + info = &dp->synths[dev]; switch (info->arg.event_passing) { case SNDRV_SEQ_OSS_PROCESS_EVENTS: if (! info->ch || ch < 0 || ch >= info->nr_voices) {

2 1

[alsa-devel] [PATCH v4 1/2] ALSA: ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> AFSX won't be used in DIT mode. The related pins are AHCLKX and the data pins. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> Acked-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> --- v4: change subjuct to include "ALSA: ASoC: " sound/soc/davinci/davinci-mcasp.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9321e5c..90a9b99 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -772,9 +772,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* Set the PDIR for Serialiser as output */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); - /* TXMASK for 24 bits */ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); -- 1.7.7

3 5

[alsa-devel] [PATCH v3] davinci-mcasp: don't overwrite DIT settings
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> Channel size settings will be made at the end of davinci_mcasp_hw_params() routine and thus overwrite frame format settings made for DIT mode. This patch fixes this issue by taking op_mode into account. Tested with official PSP 3.2 kernel and sii9022a HDMI transmitter. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> --- v3: don't set mask in davinci_hw_dit_param() v2: fix checkpatch.pl warnings sound/soc/davinci/davinci-mcasp.c | 25 +++++++++++++------------ 1 files changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 90a9b99..dcd56f2 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -643,16 +643,20 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), - TXROT(7)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), - RXROT(7)); + if (dev->op_mode != DAVINCI_MCASP_DIT_MODE) { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(rotate), TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXROT(rotate), RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, + mask); + } + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); return 0; } @@ -772,9 +776,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* TXMASK for 24 bits */ - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); - /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, -- 1.7.7

3 3

[alsa-devel] Issue with Dmix
by mateen 04 Mar '13

04 Mar '13

HI We have enabled DMIX plugin for sample rate conversion and mixing of 2 pcm streams in our alsa package (Alsa lib 1.0.22 and alsa driver 1.0.22.1) to 48K output Sample rate.However we are seeing a noise issue with 44100 Sample rate input streams or Unsigned Stereo 48K stream. This is how our current design is: We call snd_pcm_lib_malloc_page() in which alsa driver internally allocates a dma buffer and map it when Dmix is invoked. We currently reads from that mmaped buffer to our hardware buffer inside a transfer_ack_begin() function which is called from snd_pcm_period_elapsed(). However we are getting a problem in understanding how do we know how much data has been written by dmix lib in mapped buffer? As we see no ioctl or corresponding hardware function call where dmix may pass information on Size of data being written to our driver. We do see appl_ptr and hw_ptr in runtime->mmap->control and status struct. Out architecture doesn’t support defining and mapping of these status and control structure at hardware level.Thus Dmix lib allocates them and copy them to runtime substream via ioctl. As far as I could understand both appl_ptr and hw_ptr are initialised to 0 and then appl_ptr is the one adjusted by lib after every write. Should we tap this value to update hw_ptr value in status information? Will difference of hw_ptr and appl_ptr will let us know amount of frames been written? Can anyone please help us to understand criteria and concept of data copy from intermediate to hardware buffer? Any information on this would be of immense help. Regards, Shaikh Mateen

1 1

[alsa-devel] [PATCH] davinci-mcasp: don't overwrite DIT settings
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> Channel size settings will be made at the end of davinci_mcasp_hw_params() routine and thus overwrite frame format settings made for DIT mode. This patch fixes this issue by taking op_mode into account. Tested with official PSP 3.2 kernel and sii9022a HDMI transmitter. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> --- sound/soc/davinci/davinci-mcasp.c | 27 +++++++++++++++++---------- 1 files changed, 17 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 90a9b99..de59075 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -643,16 +643,23 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), - TXROT(7)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), - RXROT(7)); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + } + else + { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), + TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), + RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); + } return 0; } -- 1.7.7

3 2

[alsa-devel] [PATCH v2] davinci-mcasp: don't overwrite DIT settings
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> Channel size settings will be made at the end of davinci_mcasp_hw_params() routine and thus overwrite frame format settings made for DIT mode. This patch fixes this issue by taking op_mode into account. Tested with official PSP 3.2 kernel and sii9022a HDMI transmitter. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> --- v2: fix checkpatch.pl warnings sound/soc/davinci/davinci-mcasp.c | 22 +++++++++++++--------- 1 files changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 90a9b99..ce32b75 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -643,16 +643,20 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXSSZ(fmt), RXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXSSZ(fmt), TXSSZ(0x0F)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), - TXROT(7)); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), - RXROT(7)); + if (dev->op_mode != DAVINCI_MCASP_DIT_MODE) { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(rotate), TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXROT(rotate), RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, + mask); + } + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); return 0; } -- 1.7.7

1 0

[alsa-devel] [PATCH 09/12] ASoC: dmaengine_pcm: support use of generic DMA helper
by Shawn Guo 04 Mar '13

04 Mar '13

With generic DMA device tree binding and helper function dma_request_slave_channel() in place, dmaengine_pcm should support that in requesting DMA channel for users that support generic DMA device tree binding. Instead of inventing a new API, it defines the parameters needed by dma_request_slave_channel() into struct snd_dma_channel_params, interprets filter_data into snd_dma_channel_params, and calls the helper in case that dmaengine_pcm users pass in a NULL filter_fn. Then, dmaengine_pcm users can call snd_dmaengine_pcm_open() with NULL filter_fn and snd_dma_channel_params being filter_data to direct the API to request DMA channel using generic DMA helper. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- include/sound/dmaengine_pcm.h | 5 +++++ sound/soc/soc-dmaengine-pcm.c | 19 +++++++++++++------ 2 files changed, 18 insertions(+), 6 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index b877334..6c0f795 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -18,6 +18,11 @@ #include <sound/pcm.h> #include <linux/dmaengine.h> +struct snd_dma_channel_params { + struct device *dev; + char *name; +}; + /** * snd_pcm_substream_to_dma_direction - Get dma_transfer_direction for a PCM * substream diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 111b7d92..7ef8034 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -247,12 +247,19 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd, dma_filter_fn filter_fn, void *filter_data) { - dma_cap_mask_t mask; - - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - dma_cap_set(DMA_CYCLIC, mask); - prtd->dma_chan = dma_request_channel(mask, filter_fn, filter_data); + if (filter_fn) { + dma_cap_mask_t mask; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dma_cap_set(DMA_CYCLIC, mask); + prtd->dma_chan = dma_request_channel(mask, filter_fn, + filter_data); + } else { + struct snd_dma_channel_params *params = filter_data; + prtd->dma_chan = dma_request_slave_channel(params->dev, + params->name); + } if (!prtd->dma_chan) return -ENXIO; -- 1.7.9.5

3 4

[alsa-devel] [PATCH] davinci-mcasp: don't configure AFSX direction in DIT mode
by yegorslists＠googlemail.com 04 Mar '13

04 Mar '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> AFSX won't be used in DIT mode. The related pins are AHCLKX and the data pins. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> --- sound/soc/davinci/davinci-mcasp.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9321e5c..90a9b99 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -772,9 +772,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* Set the PDIR for Serialiser as output */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); - /* TXMASK for 24 bits */ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); -- 1.7.7

2 1

[alsa-devel] [PATCH 2/2] ALSA: snd-usb-caiaq: switch to dev_*() logging
by Daniel Mack 03 Mar '13

03 Mar '13

Get rid of the proprietary functions log() and debug() and use the generic dev_*() approach. A macro is needed to cast a cdev to a struct device *. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/caiaq/audio.c | 61 ++++++++++++++++++++++++++++++----------------- sound/usb/caiaq/control.c | 1 + sound/usb/caiaq/device.c | 42 +++++++++++++++++--------------- sound/usb/caiaq/device.h | 10 +------- sound/usb/caiaq/input.c | 8 +++++-- sound/usb/caiaq/midi.c | 9 +++---- 6 files changed, 75 insertions(+), 56 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 75d8ba9..67330af 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -16,6 +16,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/spinlock.h> #include <linux/slab.h> #include <linux/init.h> @@ -101,8 +102,9 @@ all_substreams_zero(struct snd_pcm_substream **subs) static int stream_start(struct snd_usb_caiaqdev *cdev) { int i, ret; + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, cdev); + dev_dbg(dev, "%s(%p)\n", __func__, cdev); if (cdev->streaming) return -EINVAL; @@ -118,7 +120,8 @@ static int stream_start(struct snd_usb_caiaqdev *cdev) for (i = 0; i < N_URBS; i++) { ret = usb_submit_urb(cdev->data_urbs_in[i], GFP_ATOMIC); if (ret) { - log("unable to trigger read #%d! (ret %d)\n", i, ret); + dev_err(dev, "unable to trigger read #%d! (ret %d)\n", + i, ret); cdev->streaming = 0; return -EPIPE; } @@ -130,8 +133,9 @@ static int stream_start(struct snd_usb_caiaqdev *cdev) static void stream_stop(struct snd_usb_caiaqdev *cdev) { int i; + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, cdev); + dev_dbg(dev, "%s(%p)\n", __func__, cdev); if (!cdev->streaming) return; @@ -150,17 +154,21 @@ static void stream_stop(struct snd_usb_caiaqdev *cdev) static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(substream); - debug("%s(%p)\n", __func__, substream); + struct device *dev = caiaqdev_to_dev(cdev); + + dev_dbg(dev, "%s(%p)\n", __func__, substream); substream->runtime->hw = cdev->pcm_info; snd_pcm_limit_hw_rates(substream->runtime); + return 0; } static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(substream); + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, substream); + dev_dbg(dev, "%s(%p)\n", __func__, substream); if (all_substreams_zero(cdev->sub_playback) && all_substreams_zero(cdev->sub_capture)) { /* when the last client has stopped streaming, @@ -175,14 +183,12 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) static int snd_usb_caiaq_pcm_hw_params(struct snd_pcm_substream *sub, struct snd_pcm_hw_params *hw_params) { - debug("%s(%p)\n", __func__, sub); return snd_pcm_lib_malloc_pages(sub, params_buffer_bytes(hw_params)); } static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(sub); - debug("%s(%p)\n", __func__, sub); deactivate_substream(cdev, sub); return snd_pcm_lib_free_pages(sub); } @@ -201,8 +207,9 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) int index = substream->number; struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p)\n", __func__, substream); + dev_dbg(dev, "%s(%p)\n", __func__, substream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { int out_pos; @@ -283,8 +290,9 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) static int snd_usb_caiaq_pcm_trigger(struct snd_pcm_substream *sub, int cmd) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(sub); + struct device *dev = caiaqdev_to_dev(cdev); - debug("%s(%p) cmd %d\n", __func__, sub, cmd); + dev_dbg(dev, "%s(%p) cmd %d\n", __func__, sub, cmd); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -443,6 +451,7 @@ static void read_in_urb_mode3(struct snd_usb_caiaqdev *cdev, const struct usb_iso_packet_descriptor *iso) { unsigned char *usb_buf = urb->transfer_buffer + iso->offset; + struct device *dev = caiaqdev_to_dev(cdev); int stream, i; /* paranoia check */ @@ -479,8 +488,8 @@ static void read_in_urb_mode3(struct snd_usb_caiaqdev *cdev, if (usb_buf[i] != ((stream << 1) | c) && !cdev->first_packet) { if (!cdev->input_panic) - printk(" EXPECTED: %02x got %02x, c %d, stream %d, i %d\n", - ((stream << 1) | c), usb_buf[i], c, stream, i); + dev_warn(dev, " EXPECTED: %02x got %02x, c %d, stream %d, i %d\n", + ((stream << 1) | c), usb_buf[i], c, stream, i); cdev->input_panic = 1; } @@ -497,6 +506,8 @@ static void read_in_urb(struct snd_usb_caiaqdev *cdev, const struct urb *urb, const struct usb_iso_packet_descriptor *iso) { + struct device *dev = caiaqdev_to_dev(cdev); + if (!cdev->streaming) return; @@ -516,7 +527,7 @@ static void read_in_urb(struct snd_usb_caiaqdev *cdev, } if ((cdev->input_panic || cdev->output_panic) && !cdev->warned) { - debug("streaming error detected %s %s\n", + dev_warn(dev, "streaming error detected %s %s\n", cdev->input_panic ? "(input)" : "", cdev->output_panic ? "(output)" : ""); cdev->warned = 1; @@ -619,6 +630,7 @@ static void read_completed(struct urb *urb) { struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *cdev; + struct device *dev; struct urb *out = NULL; int i, frame, len, send_it = 0, outframe = 0; size_t offset = 0; @@ -627,6 +639,7 @@ static void read_completed(struct urb *urb) return; cdev = info->cdev; + dev = caiaqdev_to_dev(cdev); if (!cdev->streaming) return; @@ -639,7 +652,7 @@ static void read_completed(struct urb *urb) } if (!out) { - log("Unable to find an output urb to use\n"); + dev_err(dev, "Unable to find an output urb to use\n"); goto requeue; } @@ -708,6 +721,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) int i, frame; struct urb **urbs; struct usb_device *usb_dev = cdev->chip.dev; + struct device *dev = caiaqdev_to_dev(cdev); unsigned int pipe; pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? @@ -716,7 +730,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) urbs = kmalloc(N_URBS * sizeof(*urbs), GFP_KERNEL); if (!urbs) { - log("unable to kmalloc() urbs, OOM!?\n"); + dev_err(dev, "unable to kmalloc() urbs, OOM!?\n"); *ret = -ENOMEM; return NULL; } @@ -724,7 +738,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) for (i = 0; i < N_URBS; i++) { urbs[i] = usb_alloc_urb(FRAMES_PER_URB, GFP_KERNEL); if (!urbs[i]) { - log("unable to usb_alloc_urb(), OOM!?\n"); + dev_err(dev, "unable to usb_alloc_urb(), OOM!?\n"); *ret = -ENOMEM; return urbs; } @@ -732,7 +746,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) urbs[i]->transfer_buffer = kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); if (!urbs[i]->transfer_buffer) { - log("unable to kmalloc() transfer buffer, OOM!?\n"); + dev_err(dev, "unable to kmalloc() transfer buffer, OOM!?\n"); *ret = -ENOMEM; return urbs; } @@ -783,6 +797,7 @@ static void free_urbs(struct urb **urbs) int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) { int i, ret; + struct device *dev = caiaqdev_to_dev(cdev); cdev->n_audio_in = max(cdev->spec.num_analog_audio_in, cdev->spec.num_digital_audio_in) / @@ -792,12 +807,12 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) CHANNELS_PER_STREAM; cdev->n_streams = max(cdev->n_audio_in, cdev->n_audio_out); - debug("cdev->n_audio_in = %d\n", cdev->n_audio_in); - debug("cdev->n_audio_out = %d\n", cdev->n_audio_out); - debug("cdev->n_streams = %d\n", cdev->n_streams); + dev_dbg(dev, "cdev->n_audio_in = %d\n", cdev->n_audio_in); + dev_dbg(dev, "cdev->n_audio_out = %d\n", cdev->n_audio_out); + dev_dbg(dev, "cdev->n_streams = %d\n", cdev->n_streams); if (cdev->n_streams > MAX_STREAMS) { - log("unable to initialize device, too many streams.\n"); + dev_err(dev, "unable to initialize device, too many streams.\n"); return -EINVAL; } @@ -805,7 +820,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) cdev->n_audio_out, cdev->n_audio_in, &cdev->pcm); if (ret < 0) { - log("snd_pcm_new() returned %d\n", ret); + dev_err(dev, "snd_pcm_new() returned %d\n", ret); return ret; } @@ -880,7 +895,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) void snd_usb_caiaq_audio_free(struct snd_usb_caiaqdev *cdev) { - debug("%s(%p)\n", __func__, cdev); + struct device *dev = caiaqdev_to_dev(cdev); + + dev_dbg(dev, "%s(%p)\n", __func__, cdev); stream_stop(cdev); free_urbs(cdev->data_urbs_in); free_urbs(cdev->data_urbs_out); diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index 2c51959..ae6b50f 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -17,6 +17,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/init.h> #include <linux/usb.h> #include <sound/control.h> diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 45c3853..d898f73 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -20,6 +20,7 @@ */ #include <linux/moduleparam.h> +#include <linux/device.h> #include <linux/interrupt.h> #include <linux/module.h> #include <linux/init.h> @@ -159,10 +160,11 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) { int ret; struct snd_usb_caiaqdev *cdev = urb->context; + struct device *dev = caiaqdev_to_dev(cdev); unsigned char *buf = urb->transfer_buffer; if (urb->status || !cdev) { - log("received EP1 urb->status = %i\n", urb->status); + dev_warn(dev, "received EP1 urb->status = %i\n", urb->status); return; } @@ -170,7 +172,7 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) case EP1_CMD_GET_DEVICE_INFO: memcpy(&cdev->spec, buf+1, sizeof(struct caiaq_device_spec)); cdev->spec.fw_version = le16_to_cpu(cdev->spec.fw_version); - debug("device spec (firmware %d): audio: %d in, %d out, " + dev_dbg(dev, "device spec (firmware %d): audio: %d in, %d out, " "MIDI: %d in, %d out, data alignment %d\n", cdev->spec.fw_version, cdev->spec.num_analog_audio_in, @@ -209,7 +211,7 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb) cdev->ep1_in_urb.actual_length = 0; ret = usb_submit_urb(&cdev->ep1_in_urb, GFP_ATOMIC); if (ret < 0) - log("unable to submit urb. OOM!?\n"); + dev_err(dev, "unable to submit urb. OOM!?\n"); } int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *cdev, @@ -239,6 +241,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, { int ret; char tmp[5]; + struct device *dev = caiaqdev_to_dev(cdev); switch (rate) { case 44100: tmp[0] = SAMPLERATE_44100; break; @@ -259,7 +262,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, tmp[3] = bpp >> 8; tmp[4] = 1; /* packets per microframe */ - debug("setting audio params: %d Hz, %d bits, %d bpp\n", + dev_dbg(dev, "setting audio params: %d Hz, %d bits, %d bpp\n", rate, depth, bpp); cdev->audio_parm_answer = -1; @@ -274,7 +277,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, return -EPIPE; if (cdev->audio_parm_answer != 1) - debug("unable to set the device's audio params\n"); + dev_dbg(dev, "unable to set the device's audio params\n"); else cdev->bpp = bpp; @@ -293,6 +296,7 @@ static void setup_card(struct snd_usb_caiaqdev *cdev) { int ret; char val[4]; + struct device *dev = caiaqdev_to_dev(cdev); /* device-specific startup specials */ switch (cdev->chip.usb_id) { @@ -346,32 +350,32 @@ static void setup_card(struct snd_usb_caiaqdev *cdev) cdev->spec.num_digital_audio_in > 0) { ret = snd_usb_caiaq_audio_init(cdev); if (ret < 0) - log("Unable to set up audio system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up audio system (ret=%d)\n", ret); } if (cdev->spec.num_midi_in + cdev->spec.num_midi_out > 0) { ret = snd_usb_caiaq_midi_init(cdev); if (ret < 0) - log("Unable to set up MIDI system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up MIDI system (ret=%d)\n", ret); } #ifdef CONFIG_SND_USB_CAIAQ_INPUT ret = snd_usb_caiaq_input_init(cdev); if (ret < 0) - log("Unable to set up input system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up input system (ret=%d)\n", ret); #endif /* finally, register the card and all its sub-instances */ ret = snd_card_register(cdev->chip.card); if (ret < 0) { - log("snd_card_register() returned %d\n", ret); + dev_err(dev, "snd_card_register() returned %d\n", ret); snd_card_free(cdev->chip.card); } ret = snd_usb_caiaq_control_init(cdev); if (ret < 0) - log("Unable to set up control system (ret=%d)\n", ret); + dev_err(dev, "Unable to set up control system (ret=%d)\n", ret); } static int create_card(struct usb_device *usb_dev, @@ -412,10 +416,11 @@ static int init_card(struct snd_usb_caiaqdev *cdev) char *c, usbpath[32]; struct usb_device *usb_dev = cdev->chip.dev; struct snd_card *card = cdev->chip.card; + struct device *dev = caiaqdev_to_dev(cdev); int err, len; if (usb_set_interface(usb_dev, 0, 1) != 0) { - log("can't set alt interface.\n"); + dev_err(dev, "can't set alt interface.\n"); return -EIO; } @@ -473,8 +478,7 @@ static int init_card(struct snd_usb_caiaqdev *cdev) } usb_make_path(usb_dev, usbpath, sizeof(usbpath)); - snprintf(card->longname, sizeof(card->longname), - "%s %s (%s)", + snprintf(card->longname, sizeof(card->longname), "%s %s (%s)", cdev->vendor_name, cdev->product_name, usbpath); setup_card(cdev); @@ -496,7 +500,7 @@ static int snd_probe(struct usb_interface *intf, usb_set_intfdata(intf, card); ret = init_card(caiaqdev(card)); if (ret < 0) { - log("unable to init card! (ret=%d)\n", ret); + dev_err(&usb_dev->dev, "unable to init card! (ret=%d)\n", ret); snd_card_free(card); return ret; } @@ -506,15 +510,16 @@ static int snd_probe(struct usb_interface *intf, static void snd_disconnect(struct usb_interface *intf) { - struct snd_usb_caiaqdev *cdev; struct snd_card *card = usb_get_intfdata(intf); - - debug("%s(%p)\n", __func__, intf); + struct snd_usb_caiaqdev *cdev = caiaqdev(card); + struct device *dev; if (!card) return; - cdev = caiaqdev(card); + dev = caiaqdev_to_dev(cdev); + dev_dbg(dev, "%s(%p)\n", __func__, intf); + snd_card_disconnect(card); #ifdef CONFIG_SND_USB_CAIAQ_INPUT @@ -539,4 +544,3 @@ static struct usb_driver snd_usb_driver = { }; module_usb_driver(snd_usb_driver); - diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index 7176a0e..ad102fa 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -25,16 +25,7 @@ #define CAIAQ_USB_STR_LEN 0xff #define MAX_STREAMS 32 -//#define SND_USB_CAIAQ_DEBUG - #define MODNAME "snd-usb-caiaq" -#define log(x...) snd_printk(KERN_WARNING MODNAME" log: " x) - -#ifdef SND_USB_CAIAQ_DEBUG -#define debug(x...) snd_printk(KERN_WARNING MODNAME " debug: " x) -#else -#define debug(x...) do { } while(0) -#endif #define EP1_CMD_GET_DEVICE_INFO 0x1 #define EP1_CMD_READ_ERP 0x2 @@ -129,6 +120,7 @@ struct snd_usb_caiaq_cb_info { }; #define caiaqdev(c) ((struct snd_usb_caiaqdev*)(c)->private_data) +#define caiaqdev_to_dev(d) (d->chip.card->dev) int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *cdev, int rate, int depth, int bbp); int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *cdev, int digital, int analog, int erp); diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index a32ad7c..fe8f4b4 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -16,6 +16,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/gfp.h> #include <linux/init.h> #include <linux/usb.h> @@ -340,6 +341,8 @@ static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *cdev, const unsigned char *buf, unsigned int len) { + struct device *dev = caiaqdev_to_dev(cdev); + while (len) { unsigned int i, block_id = (buf[0] << 8) | buf[1]; @@ -466,7 +469,7 @@ static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *cdev, break; default: - debug("%s(): bogus block (id %d)\n", + dev_dbg(dev, "%s(): bogus block (id %d)\n", __func__, block_id); return; } @@ -500,6 +503,7 @@ static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *cdev, static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) { struct snd_usb_caiaqdev *cdev = urb->context; + struct device *dev = caiaqdev_to_dev(cdev); unsigned char *buf = urb->transfer_buffer; int ret; @@ -535,7 +539,7 @@ requeue: cdev->ep4_in_urb->actual_length = 0; ret = usb_submit_urb(cdev->ep4_in_urb, GFP_ATOMIC); if (ret < 0) - log("unable to submit urb. OOM!?\n"); + dev_err(dev, "unable to submit urb. OOM!?\n"); } static int snd_usb_caiaq_input_open(struct input_dev *idev) diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 63c5a2c..2d75884 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -16,6 +16,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <linux/device.h> #include <linux/usb.h> #include <linux/gfp.h> #include <sound/rawmidi.h> @@ -65,6 +66,7 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *cdev, struct snd_rawmidi_substream *substream) { int len, ret; + struct device *dev = caiaqdev_to_dev(cdev); cdev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; cdev->midi_out_buf[1] = 0; /* port */ @@ -79,9 +81,9 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *cdev, ret = usb_submit_urb(&cdev->midi_out_urb, GFP_ATOMIC); if (ret < 0) - log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," - "ret=%d, len=%d\n", - substream, ret, len); + dev_err(dev, + "snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," + "ret=%d, len=%d\n", substream, ret, len); else cdev->midi_out_active = 1; } @@ -171,4 +173,3 @@ void snd_usb_caiaq_midi_output_done(struct urb* urb) snd_usb_caiaq_midi_send(cdev, cdev->midi_out_substream); } - -- 1.8.1.4

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[alsa-devel] usb: Endpoints sharing the same interface
by Mark Hills 03 Mar '13

03 Mar '13

Novation Twitch [1] is a USB device with audio capture and playback endpoints on the same bInterfaceNumber. This has thrown up some tricky cases in the USB audio driver, and I'd appreciate some advice. I'll follow up with my patches so far as an RFC. This device has endpoints for 2 channels capture (0x01), 4 channels playback (0x82), both on interface #0. Interface #1 is raw MIDI. I seem to be finding code that assumes pcm to always use the first endpoint on an interface. eg. problems with maxpacksize (see patch in thread) Now I'm in a position where playback _or_ capture works reliably. But to start capture stops playback, and vice-versa (or sometimes causes hangs) It seems like snd_pcm_prepare() concludes that need_setup_ep is required, but I think this may be based on something going on in set_sample_rate_v1() also looking to the first endpoint -- usb_sndctrlpipe(dev,0). Before I do any more work on this, are there any hints where to go with this? If we can overcome thse limitations then the same would likely apply to Saffire 6 USB and other Focusrite devices [2], and maybe some of the other devices with quirks that only support playback. Many thanks [1] http://uk.novationmusic.com/digital-dj/twitch/ [2] http://focusritedevelopmentteam.wordpress.com/category/drivers/ -- Mark --- Bus 005 Device 010: ID 1235:0018 Novation EMS Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x1235 Novation EMS idProduct 0x0018 bcdDevice 1.00 iManufacturer 1 Focusrite A.E. Ltd iProduct 2 Twitch iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 64 bNumInterfaces 2 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 498mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x024c 1x 588 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0126 1x 294 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 1 Device Status: 0x0000 (Bus Powered)

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[alsa-devel] amdtp: syt handling
by Robert von Chamier 03 Mar '13

03 Mar '13

Hello, i am working on the firewire amdtp module. My aim is to add the capture functionality. Already in my prototype the module captures up to 192k sampling rate. But ... How i have to handle the syt in the CIP packages? Do i compare the syt with the corresponding cycle? or ... Do i take the syt as timestamp for alsa? Is it possible to commit a timestamp to alsa for the corresponding samples? Can you help me? Robert

2 1

[alsa-devel] [PATCH] ASoC: max98090: Convert to devm_regmap_init_i2c
by Axel Lin 03 Mar '13

03 Mar '13

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98090.c | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index fc17604..6dd36da 100755 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2324,28 +2324,20 @@ static int max98090_i2c_probe(struct i2c_client *i2c, max98090->pdata = i2c->dev.platform_data; max98090->irq = i2c->irq; - max98090->regmap = regmap_init_i2c(i2c, &max98090_regmap); + max98090->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap); if (IS_ERR(max98090->regmap)) { ret = PTR_ERR(max98090->regmap); dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); - goto err_enable; + return ret; } - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_max98090, max98090_dai, - ARRAY_SIZE(max98090_dai)); - if (ret < 0) - regmap_exit(max98090->regmap); - -err_enable: - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98090, + max98090_dai, ARRAY_SIZE(max98090_dai)); } static int max98090_i2c_remove(struct i2c_client *client) { - struct max98090_priv *max98090 = dev_get_drvdata(&client->dev); snd_soc_unregister_codec(&client->dev); - regmap_exit(max98090->regmap); return 0; } -- 1.7.9.5

2 2

[alsa-devel] [PATCH RFT] ASoC: wm8350: Use jiffies rather than msecs in schedule_delayed_work()
by Axel Lin 03 Mar '13

03 Mar '13

The delay parameter of schedule_delayed_work() is number of jiffies to wait rather than miliseconds. Before commit 6d3c26bcb "ASoC: Use delayed work to debounce WM8350 jack IRQs", the debounce time is 200 miliseconds in wm8350_hp_jack_handler(). So I think this is a bug when convert to use delayed work. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8350.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index ec0efc1..0e8b3aa 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1301,7 +1301,7 @@ static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpl.work, 200); + schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } @@ -1318,7 +1318,7 @@ static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpr.work, 200); + schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: omap: Check regulator enable for DAC on Pandora
by Mark Brown 03 Mar '13

03 Mar '13

This will probably never fail but it's better style. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/omap/omap3pandora.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 805512f..953483e 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -85,7 +85,11 @@ static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, * VCC power on/off and /PD pin high/low */ if (SND_SOC_DAPM_EVENT_ON(event)) { - regulator_enable(omap3pandora_dac_reg); + ret = regulator_enable(omap3pandora_dac_reg); + if (ret != 0) { + dev_err(w->dapm.dev, "Failed to power DAC: %d\n", ret); + return ret; + } mdelay(1); gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1); } else { -- 1.7.10.4

3 2

Re: [alsa-devel] [PATCH v2 3/4] spi: s3c64xx: add dma maxburst size initialization
by Grant Likely 02 Mar '13

02 Mar '13

On Fri, 22 Feb 2013 17:09:18 +0900, Boojin Kim <boojin.kim(a)samsung.com> wrote: > This patch adds dma maxburst size initialization. > The maxburst should be set by MODE_CFGn.DMA_TYPE, > because the pl330 dma driver supports burst mode. > > Signed-off-by: Hyeonkook Kim <hk619.kim(a)samsung.com> Acked-by: Grant Likely <grant.likely(a)secretlab.ca> This series probably needs to be merged together through the arm-soc tree. g. > --- > drivers/spi/spi-s3c64xx.c | 18 ++++++++++++++++++ > 1 files changed, 18 insertions(+), 0 deletions(-) > > diff --git a/drivers/spi/spi-s3c64xx.c b/drivers/spi/spi-s3c64xx.c > index e862ab8..5e6cafe 100644 > --- a/drivers/spi/spi-s3c64xx.c > +++ b/drivers/spi/spi-s3c64xx.c > @@ -287,10 +287,20 @@ static void prepare_dma(struct s3c64xx_spi_dma_data *dma, > struct s3c64xx_spi_driver_data *sdd; > struct samsung_dma_prep info; > struct samsung_dma_config config; > + void __iomem *regs; > + unsigned int mode_cfg; > > if (dma->direction == DMA_DEV_TO_MEM) { > sdd = container_of((void *)dma, > struct s3c64xx_spi_driver_data, rx_dma); > + > + regs = sdd->regs; > + mode_cfg = readl(regs + S3C64XX_SPI_MODE_CFG); > + if (mode_cfg & S3C64XX_SPI_MODE_4BURST) > + config.maxburst = 4; > + else > + config.maxburst = 1; > + > config.direction = sdd->rx_dma.direction; > config.fifo = sdd->sfr_start + S3C64XX_SPI_RX_DATA; > config.width = sdd->cur_bpw / 8; > @@ -298,6 +308,14 @@ static void prepare_dma(struct s3c64xx_spi_dma_data *dma, > } else { > sdd = container_of((void *)dma, > struct s3c64xx_spi_driver_data, tx_dma); > + > + regs = sdd->regs; > + mode_cfg = readl(regs + S3C64XX_SPI_MODE_CFG); > + if (mode_cfg & S3C64XX_SPI_MODE_4BURST) > + config.maxburst = 4; > + else > + config.maxburst = 1; > + > config.direction = sdd->tx_dma.direction; > config.fifo = sdd->sfr_start + S3C64XX_SPI_TX_DATA; > config.width = sdd->cur_bpw / 8; > -- > 1.7.5.4 > > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ -- Grant Likely, B.Sc, P.Eng. Secret Lab Technologies, Ltd.

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Re: [alsa-devel] [PATCH v2 3/4] spi: s3c64xx: add dma maxburst size initialization
by Grant Likely 02 Mar '13

02 Mar '13

On Fri, 22 Feb 2013 17:09:18 +0900, Boojin Kim <boojin.kim(a)samsung.com> wrote: > This patch adds dma maxburst size initialization. > The maxburst should be set by MODE_CFGn.DMA_TYPE, > because the pl330 dma driver supports burst mode. > > Signed-off-by: Hyeonkook Kim <hk619.kim(a)samsung.com> Hello Hyeonkook, I applied this patch but then discovered that it relys on the previous patches that you've posted. When patched depend on each other like this please post them as a series with [PATCH x/4] in the subject line. 'git send-email' will format this correctly for you. g. > --- > drivers/spi/spi-s3c64xx.c | 18 ++++++++++++++++++ > 1 files changed, 18 insertions(+), 0 deletions(-) > > diff --git a/drivers/spi/spi-s3c64xx.c b/drivers/spi/spi-s3c64xx.c > index e862ab8..5e6cafe 100644 > --- a/drivers/spi/spi-s3c64xx.c > +++ b/drivers/spi/spi-s3c64xx.c > @@ -287,10 +287,20 @@ static void prepare_dma(struct s3c64xx_spi_dma_data *dma, > struct s3c64xx_spi_driver_data *sdd; > struct samsung_dma_prep info; > struct samsung_dma_config config; > + void __iomem *regs; > + unsigned int mode_cfg; > > if (dma->direction == DMA_DEV_TO_MEM) { > sdd = container_of((void *)dma, > struct s3c64xx_spi_driver_data, rx_dma); > + > + regs = sdd->regs; > + mode_cfg = readl(regs + S3C64XX_SPI_MODE_CFG); > + if (mode_cfg & S3C64XX_SPI_MODE_4BURST) > + config.maxburst = 4; > + else > + config.maxburst = 1; > + > config.direction = sdd->rx_dma.direction; > config.fifo = sdd->sfr_start + S3C64XX_SPI_RX_DATA; > config.width = sdd->cur_bpw / 8; > @@ -298,6 +308,14 @@ static void prepare_dma(struct s3c64xx_spi_dma_data *dma, > } else { > sdd = container_of((void *)dma, > struct s3c64xx_spi_driver_data, tx_dma); > + > + regs = sdd->regs; > + mode_cfg = readl(regs + S3C64XX_SPI_MODE_CFG); > + if (mode_cfg & S3C64XX_SPI_MODE_4BURST) > + config.maxburst = 4; > + else > + config.maxburst = 1; > + > config.direction = sdd->tx_dma.direction; > config.fifo = sdd->sfr_start + S3C64XX_SPI_TX_DATA; > config.width = sdd->cur_bpw / 8; > -- > 1.7.5.4 > > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ -- Grant Likely, B.Sc, P.Eng. Secret Lab Technologies, Ltd.

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Re: [alsa-devel] [PATCH v2 3/4] spi: s3c64xx: add dma maxburst size initialization
by Grant Likely 02 Mar '13

02 Mar '13

On Fri, 22 Feb 2013 17:09:18 +0900, Boojin Kim <boojin.kim(a)samsung.com> wrote: > This patch adds dma maxburst size initialization. > The maxburst should be set by MODE_CFGn.DMA_TYPE, > because the pl330 dma driver supports burst mode. > > Signed-off-by: Hyeonkook Kim <hk619.kim(a)samsung.com> Applied, thanks. g.

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Re: [alsa-devel] [Alsa-user] Status of CM6631 USB
by Daniel Mack 02 Mar '13

02 Mar '13

Hi Torstein, (+ alsa-devel) On 02.03.2013 12:09, Torstein Hegge wrote: > Daniel Mack <zonque <at> gmail.com> writes: >>>> This is a known bug, also most probably flaw in the CMEDIA chip, >>>> and not yet properly worked around in the snd-usb driver. If you >>>> want to investigate, have a look at the feedback format >>>> autodetection code in sound/usb/endpoint.c. >>> >>> Thanks, I'll have a look. >> >> I believe what's going on here is that the device is confused after >> sample rate changes and keeps sending back feedback data for the old >> rate on its feedback endpoint. That causes the autodetection algorithm, >> which kicks in on every first packet of a stream, to go nuts, which >> causes the part of the driver that sends the data assume a very wrong >> rate. Some printk() there should give you a hint. > > I had a look at this using another C-Media CM6631 based device, the Corda > Daccord (usbid 0x0d8c, 0x0309). As far as I can see, what happens in the > feedback format auto-detection is: > > When starting playback of a file with a different sampling frequency than the > previous file, the reported feedback frequency is in the same range as the > previously played file, varying within +-1%. > > The feedback format detection gives a shift of -1 and reports the feedback > frequency as ~48kHz when playing a 44.1kHz file after a 96kHz file. Same thing > happens when playing a 96kHz file after a 44.1kHz file, the device reports > feedback frequency in the 44.1kHz range, the feedback format detection gives a > shift of +1 and reports the feedback frequency as ~88kHz. That's what I suspected. > Bypassing the feedback format detection by hard-coding the frequency shift to > zero and setting the feedback frequency ep->freqm equal to the nominal > frequency ep->freqn if the feedback frequency is far off from the expected > value (like 48kHz when expecting 44.1kHz) does not change the distortion. > > Similarly, forcing the feedback detection to run multiple times until it > reaches an acceptable value just keeps the feedback detection to repeatedly > until playback is stopped. This probably means any amount of skipping packages > early in the stream won't work. Ok. > It seems like the device stays in the previous frequency until the interface > is reset. One possible workaround is to call usb_set_interface() again after > snd_usb_init_sample_rate(), like this: I see - that also explains why explicitly stopping and restarting the stream fixes the problem in most cases, right? > > diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c > index d82e378..01978a6 100644 > --- a/sound/usb/pcm.c > +++ b/sound/usb/pcm.c > @@ -710,6 +710,16 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream > *substream) > subs->need_setup_ep = false; > } > > + /* Some devices doesn't respond to sample rate changes while the > + * interface is active. */ > + switch (subs->stream->chip->usb_id) { > + case USB_ID(0x0d8c, 0x0304): /* C-Media - Schiit USB Interface */ > + case USB_ID(0x0d8c, 0x0309): /* C-Media CM6631 */ > + usb_set_interface(subs->dev, subs->cur_audiofmt->iface, > + subs->cur_audiofmt->altset_idx); > + break; > + } > + > /* some unit conversions in runtime */ > subs->data_endpoint->maxframesize = > bytes_to_frames(runtime, subs->data_endpoint->maxpacksize); > > This fixes all problems related to sample rate changes for me. I have only > tested with the Daccord, but it should work for the Schiit and other CM6631 > based devices as well. Many thanks for looking into this! Much appreciated. Could you cook a real patch please and send it to alsa-devel? Have a look at Documentation/SubmittingPatches if that's your first submission. Best, Daniel

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[alsa-devel] [PATCH 1/2] ASoC: max98090: Fix checkpatch errors related to spacing
by Sachin Kamat 02 Mar '13

02 Mar '13

Fixes the following type of checkpatch errors: ERROR: "foo * bar" should be "foo *bar" Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/max98090.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1cf017f..89f83f8 100755 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -507,16 +507,16 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, return 0; } -static const char * max98090_perf_pwr_text[] = +static const char *max98090_perf_pwr_text[] = { "High Performance", "Low Power" }; -static const char * max98090_pwr_perf_text[] = +static const char *max98090_pwr_perf_text[] = { "Low Power", "High Performance" }; static const struct soc_enum max98090_vcmbandgap_enum = SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT, ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); -static const char * max98090_osr128_text[] = { "64*fs", "128*fs" }; +static const char *max98090_osr128_text[] = { "64*fs", "128*fs" }; static const struct soc_enum max98090_osr128_enum = SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT, @@ -533,28 +533,28 @@ static const struct soc_enum max98090_filter_dmic34mode_enum = M98090_FLT_DMIC34MODE_SHIFT, ARRAY_SIZE(max98090_mode_text), max98090_mode_text); -static const char * max98090_drcatk_text[] = +static const char *max98090_drcatk_text[] = { "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" }; static const struct soc_enum max98090_drcatk_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT, ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text); -static const char * max98090_drcrls_text[] = +static const char *max98090_drcrls_text[] = { "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" }; static const struct soc_enum max98090_drcrls_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT, ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text); -static const char * max98090_alccmp_text[] = +static const char *max98090_alccmp_text[] = { "1:1", "1:1.5", "1:2", "1:4", "1:INF" }; static const struct soc_enum max98090_alccmp_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT, ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text); -static const char * max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; +static const char *max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; static const struct soc_enum max98090_drcexp_enum = SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT, @@ -857,7 +857,7 @@ static const struct soc_enum mic2_mux_enum = static const struct snd_kcontrol_new max98090_mic2_mux = SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); -static const char * max98090_micpre_text[] = { "Off", "On" }; +static const char *max98090_micpre_text[] = { "Off", "On" }; static const struct soc_enum max98090_pa1en_enum = SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: soc-core: use snd_soc_register_dais() on codec
by Kuninori Morimoto 02 Mar '13

02 Mar '13

snd_soc_register_dais() considers dai counts inside. snd_soc_register_codec() does not need to care for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ceac893..5f44d22 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4047,11 +4047,10 @@ int snd_soc_register_codec(struct device *dev, mutex_unlock(&client_mutex); /* register any DAIs */ - if (num_dai) { - ret = snd_soc_register_dais(dev, dai_drv, num_dai); - if (ret < 0) - dev_err(codec->dev, "ASoC: Failed to regster" - " DAIs: %d\n", ret); + ret = snd_soc_register_dais(dev, dai_drv, num_dai); + if (ret < 0) { + dev_err(codec->dev, "ASoC: Failed to regster DAIs: %d\n", ret); + goto fail_codec_name; } dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name); @@ -4073,7 +4072,6 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); void snd_soc_unregister_codec(struct device *dev) { struct snd_soc_codec *codec; - int i; list_for_each_entry(codec, &codec_list, list) { if (dev == codec->dev) @@ -4082,9 +4080,7 @@ void snd_soc_unregister_codec(struct device *dev) return; found: - if (codec->num_dai) - for (i = 0; i < codec->num_dai; i++) - snd_soc_unregister_dai(dev); + snd_soc_unregister_dais(dev, codec->num_dai); mutex_lock(&client_mutex); list_del(&codec->list); -- 1.7.9.5

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Re: [alsa-devel] [PATCH 1/5 v7][RFC] ASoC: add snd_soc_register_cpu()
by Mark Brown 02 Mar '13

02 Mar '13

On Fri, Mar 01, 2013 at 09:42:10PM +0000, Liam Girdwood wrote: > On Fri, 2013-03-01 at 21:50 +0100, Lars-Peter Clausen wrote: > > object. I mean the distinction is mostly for historic reasons from a time where > > it was only possible to bind a the cpu_dai of a link to a CPU DAI and the > > codec_dai to a CODEC DAI. But these dais both can for example be bound CODEC > > DAIs. I think to have a common base here would allow us to do some cleanups to > > the ASoC core and also avoid the duplication with snd_soc_of_get_cpu_dai_name > > and snd_soc_of_get_codec_dai_name which was introduced in this series. > I've been thinking about this distinction for some time too (since > multi-component support). You are right, the division with CODEC and CPU > DAIs is entirely historical. The same logic also applies to platform and > codec drivers too. The DAIs are now the same of course, it's just the containing object that differs. Platform drivers are a bit different to CODEC drivers for me though in that they encapsulate the DMA equivalent of a DAI - I can see a SoC having multiple platform drivers if it's got a couple of different DMA controllers for example (I've seen some which have a fancier one on faster buses for example). > This does need some mechanical cleanup where we have a generic audio > component driver object that contains generic DAIs. i.e. codec and > platform drivers would become generic component drivers with generic > DAIs. There's also the perennial questions about code churn and if it's worth it of course. Should we be adding a chip driver type rather than a CPU one here and then going through and doing the work to make sure that you can use a chip driver wherever you use a CODEC driver so we can start converting things over?

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Re: [alsa-devel] ASoC: BeagleBoard driver development (PCM3168)
by wendelin klimann 01 Mar '13

01 Mar '13

Hello and thanks for the support I would advice to update. Eventually you want to upstream the codec driver, > so > it is going to be easier for you. > Well i followed your advice and updated to kernel version 3.7.4+ and to ALSA version k3.7.4+, and i will be glad to upstram the finalized driver. As you recommended i am using your omap_twl4030.c file as a starting-point for my driver development, and i am trying to get the adapted machine driver to run but without success. The machine driver is not even going into * static __devinit int omap_pcm3168_probe(struct platform_device *pdev)* and i am really stuck at this point. It would be really nice if you could help me. Regards Wendelin File: /* * omap-pcm3168.c -- PCM3168 ASoC driver for the BeagleBoard. * * based on: * > omap-twl4030.c -- SoC audio for TI SoC based boards with twl4030 codec * > * > Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com * > All rights reserved. * > * > Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> * * adapted by Klimann Wendelin <wklimann(a)hotmail.com> * * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * version 2 as published by the Free Software Foundation. * * This program is distributed in the hope that it will be useful, but * WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA * 02110-1301 USA * */ #include <linux/platform_device.h> //#include <linux/platform_data/omap-twl4030.h> #include <linux/module.h> #include <linux/of.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include "omap-mcbsp.h" #include "omap-pcm.h" static int omap_pcm3168_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 *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; unsigned int fmt; int ret; fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF; /* Set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { dev_err(card->dev, "can't set codec DAI configuration\n"); return ret; } /* Set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, fmt); if (ret < 0) { dev_err(card->dev, "can't set cpu DAI configuration\n"); return ret; } return 0; } static struct snd_soc_ops omap_pcm3168_ops = { .hw_params = omap_pcm3168_hw_params, }; /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link omap_pcm3168_dai_links[] = { { .name = "PCM3168", .stream_name = "PCM3168", .cpu_dai_name = "omap-mcbsp.3", .codec_dai_name = "pcm3168-hifi", .platform_name = "omap-pcm-audio", .codec_name = "pcm3168-codec.0", .ops = &omap_pcm3168_ops, }, }; /* Audio machine driver */ static struct snd_soc_card omap_pcm3168_card = { .owner = THIS_MODULE, .dai_link = omap_pcm3168_dai_links, .num_links = ARRAY_SIZE(omap_pcm3168_dai_links), }; static __devinit int omap_pcm3168_probe(struct platform_device *pdev) { printk("in __devinit omap_pcm3168_probe -> by kli"); struct omap_tw4030_pdata *pdata = dev_get_platdata(&pdev->dev); struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_pcm3168_card; int ret = 0; card->dev = &pdev->dev; if (node) { struct device_node *dai_node; if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; } dai_node = of_parse_phandle(node, "ti,mcbsp", 0); if (!dai_node) { dev_err(&pdev->dev, "McBSP node is not provided\n"); return -EINVAL; } omap_pcm3168_dai_links[0].cpu_dai_name = NULL; omap_pcm3168_dai_links[0].cpu_of_node = dai_node; } /*else if (pdata) { if (pdata->card_name) { card->name = pdata->card_name; } else { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; } }*/ else { dev_err(&pdev->dev, "Missing pdata\n"); return -ENODEV; } ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } return 0; } static int __devexit omap_pcm3168_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); snd_soc_unregister_card(card); return 0; } static const struct of_device_id omap_pcm3168_of_match[] = { {.compatible = "ti,omap-pcm3168", }, { }, }; MODULE_DEVICE_TABLE(of, omap_pcm3168_of_match); static struct platform_driver omap_pcm3168_driver = { .driver = { .name = "omap-pcm3168", .owner = THIS_MODULE, // .pm = &snd_soc_pm_ops, // .of_match_table = omap_pcm3168_of_match, }, .probe = omap_pcm3168_probe, .remove = __devexit_p(omap_pcm3168_remove), }; module_platform_driver(omap_pcm3168_driver); MODULE_AUTHOR("Klimann Wendelin <wklimann(a)hotmail.com>"); MODULE_DESCRIPTION("ALSA SoC PCM3168 add on Soundcard"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:omap-pcm3168");

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[alsa-devel] Thinkpad T400 - internal mic doesn't work, can't set CAPTURE - schematics included
by Arkadiusz Miśkiewicz 01 Mar '13

01 Mar '13

Hi, My Thinkpad T400 (2764CTO model) doesn't have ability to set CAPTURE on microphone inputs with alsa. Most likely some quirk is missing. The laptop has internal microphone, microphone in input, headphones out output and 2 internal speakers. That's all. Is uses Conexant CX20561_15Z internally and there is detailed T400 schematic available: http://kythuatphancung.com/download/lenovo-thinkpad-t400-schematic- diagram.html?dl (for audio look around page 43). Kernels 3.7.x, fresh 3.8 - no luck. Tried even linux-next from today but it dies early on alsa unrelated issues. I've tried model options laptop, hp, hp-dv6736, hp-700, toshiba, ideapad, auto. None of these work fine. With model=hp-700 I can set CAPTURE but it doesn't work (silent audio is captured using arecord or in google hangout). External mic connected also doesn't work with these setups. http://www.alsa-project.org/db/?f=394ff4cdca33e75c8a47203d42b89d8de36deca7 [3.8 kernel] http://www.alsa-project.org/db/?f=302877e50f99835075d038207f6db0dfd89873b0 [3.7 kernel] Thinkpad is 2764CTO model exactly. I can test various hacks, patches etc easily. Could someone look at schematic and provide quirks to test? Thanks! -- Arkadiusz Miśkiewicz, arekm / maven.pl

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[alsa-devel] [PATCH v3] davinci-mcasp: Add support for multichannel playback
by Michal Bachraty 01 Mar '13

01 Mar '13

Davinci McASP has support for I2S multichannel playback. For I2S playback/receive, each serializer is capable to play 2 channels (L/R) audio data.Serializer function (Playback-receive-none) is configured in DT, depending on hardware specification. It is possible to play less channels than configured in DT. For that purpose,only specific number of active serializers are enabled. McASP FIFO need to have DMA transfer Bcnt set to number of enabled serializers, otherwise no data are transfered to McASP and Alsa generates "DMA/IRQ playback write error (DMA or IRQ trouble?)" error. For TDM mode, McASP is capable to play or receive 32 channels for one serializer. McAsp has support for max 16 serializer, therefore max channels is 32 * 8. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 66 ++++++++++++++++++++++++++++++++----- sound/soc/davinci/davinci-pcm.c | 16 +++++---- sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 68 insertions(+), 15 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d2ca682..0a1af1a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -235,6 +235,10 @@ #define DISMOD (val)(val<<2) #define TXSTATE BIT(4) #define RXSTATE BIT(5) +#define SRMOD_MASK 3 +#define SRMOD_INACTIVE 0 +#define SRMOD_TX 1 +#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -657,12 +661,15 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, return 0; } -static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, + int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; - + u8 ser; + u8 slots = dev->tdm_slots; + u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +687,42 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) } for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) + tx_ser++; + if (dev->serial_dir[i] == RX_MODE) + rx_ser++; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ser = tx_ser; + else + ser = rx_ser; + + if (ser < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + return -EINVAL; + } + + tx_ser = 0; + rx_ser = 0; + + for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) { + if (dev->serial_dir[i] == TX_MODE && + tx_ser < max_active_serializers) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE) { + } else if (dev->serial_dir[i] == RX_MODE && + rx_ser < max_active_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } @@ -729,6 +762,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } + + return 0; } static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) @@ -812,8 +847,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + u8 slots = dev->tdm_slots; + int channels; + struct snd_interval *pcm_channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels = pcm_channels->min; - davinci_hw_common_param(dev, substream->stream); + if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +903,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_params->active_serializers = (channels + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +978,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, @@ -1015,8 +1057,16 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( pdata->op_mode = val; ret = of_property_read_u32(np, "tdm-slots", &val); - if (ret >= 0) + if (ret >= 0) { + if (val < 2 || val > 32) { + dev_err(&pdev->dev, + "tdm-slots must be in rage [2-32]\n"); + ret = -EINVAL; + goto nodata; + } + pdata->tdm_slots = val; + } ret = of_property_read_u32(np, "num-serializer", &val); if (ret >= 0) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bb57552..3af8b50 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,6 +182,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; + unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -195,14 +196,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level; + count /= fifo_level * serializers; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; - dst_bidx = 0; - src_cidx = data_type * fifo_level; + dst_bidx = 4; + src_cidx = data_type * fifo_level * serializers; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -210,7 +211,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level; + dst_cidx = data_type * fifo_level * serializers; } acnt = prtd->params->acnt; @@ -224,9 +225,10 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, - ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, + fifo_level * serializers, + count, fifo_level * serializers, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fbb710c..0d84d32 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,6 +27,7 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; + unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

3 3

[alsa-devel] [PATCH 1/2] ASoC: wm5102: Correct OUT2 volume and switch names
by Charles Keepax 01 Mar '13

01 Mar '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index ab69c83..deb1097 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -755,7 +755,7 @@ SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), @@ -767,7 +767,7 @@ SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ASoC: arizona: Don't enable FLL on REFCLK configuration
by Charles Keepax 01 Mar '13

01 Mar '13

Enabling the FLL when REFCLK is being configured is not what the user would expect and can cause issues if SYNCCLK has no specified frequency. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 15 ++++++--------- 1 files changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6837863..eb25ab9 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1148,17 +1148,17 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, if (source < 0) return -EINVAL; - if (fll->ref_src == source && fll->ref_freq == Fref && - fll->fout == Fout) + if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (Fout) { - ret = arizona_calc_fll(fll, &ref, Fref, Fout); + if (fll->fout) { + ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); if (ret != 0) return ret; if (fll->sync_src >= 0) { - ret = arizona_calc_fll(fll, &sync, fll->sync_freq, Fout); + ret = arizona_calc_fll(fll, &sync, fll->sync_freq, + fll->fout); if (ret != 0) return ret; } @@ -1166,12 +1166,9 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, fll->ref_src = source; fll->ref_freq = Fref; - fll->fout = Fout; - if (Fout) { + if (fll->fout) { arizona_enable_fll(fll, &ref, &sync); - } else { - arizona_disable_fll(fll); } return 0; -- 1.7.2.5

2 1

[alsa-devel] [PATCH] hdajackretask: Fix "advanced override" bug
by David Henningsson 01 Mar '13

01 Mar '13

The values were not correctly retreived from the comboboxes. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hdajackretask/main-gtk.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/hdajackretask/main-gtk.c b/hdajackretask/main-gtk.c index 65d0f3a..4227669 100644 --- a/hdajackretask/main-gtk.c +++ b/hdajackretask/main-gtk.c @@ -183,14 +183,14 @@ static void update_user_pin_config(ui_data_t* ui) continue; if (ui->free_overrides) { - int i; + int j; int index; unsigned long val = 0; - for (i = 0; i < FREE_OVERRIDES_COUNT; i++) { - index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->jacktype)); - if (index < 0) + for (j = 0; j < FREE_OVERRIDES_COUNT; j++) { + index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->free_override_cb[j])); + if (index < 0) break; - val += get_free_override_list(i)[index].value; + val += get_free_override_list(j)[index].value; } if (index < 0) continue; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: snd-core: tidyup snd_soc_register_codec() fail case
by Kuninori Morimoto 01 Mar '13

01 Mar '13

kfree() on snd_soc_register_codec() was summarized to one place. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 689eb04..ceac893 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3983,8 +3983,8 @@ int snd_soc_register_codec(struct device *dev, /* create CODEC component name */ codec->name = fmt_single_name(dev, &codec->id); if (codec->name == NULL) { - kfree(codec); - return -ENOMEM; + ret = -ENOMEM; + goto fail_codec; } if (codec_drv->compress_type) @@ -4023,7 +4023,7 @@ int snd_soc_register_codec(struct device *dev, reg_size, GFP_KERNEL); if (!codec->reg_def_copy) { ret = -ENOMEM; - goto fail; + goto fail_codec_name; } } } @@ -4057,8 +4057,9 @@ int snd_soc_register_codec(struct device *dev, dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name); return 0; -fail: +fail_codec_name: kfree(codec->name); +fail_codec: kfree(codec); return ret; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/3] ASoC: wm8960: Correct register 0 and 1 defaults
by Mark Brown 01 Mar '13

01 Mar '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8960.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9bb9273..4cb49eb 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -53,8 +53,8 @@ * using 2 wire for device control, so we cache them instead. */ static const struct reg_default wm8960_reg_defaults[] = { - { 0x0, 0x0097 }, - { 0x1, 0x0097 }, + { 0x0, 0x00a7 }, + { 0x1, 0x00a7 }, { 0x2, 0x0000 }, { 0x3, 0x0000 }, { 0x4, 0x0000 }, -- 1.7.10.4

1 2

[alsa-devel] [PATCH] ALSA: hda - Use LPIB position fix for Acer AO751h
by David Henningsson 28 Feb '13

28 Feb '13

The user reports a regression (choppy audio), and that position_fix=1 (or position_fix=4) fixes the issue. This is a Poulsbo controller (8086:811b (rev 07)), so might be worth checking if other Poulsbo machines suffer from the same bug. Cc: stable(a)vger.kernel.org [v3.7+] BugLink: https://bugs.launchpad.net/bugs/1132562 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4cea6bb6..3505f1e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3114,6 +3114,7 @@ static bool check_hdmi_disabled(struct pci_dev *pci) * white/black-listing for position_fix */ static struct snd_pci_quirk position_fix_list[] = { + SND_PCI_QUIRK(0x1025, 0x0244, "Acer AO751h", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), -- 1.7.9.5

2 5

[alsa-devel] Query about SNDRV_PCM_FORMAT_S24_LE
by Patrick Lai 28 Feb '13

28 Feb '13

Hi, The DSP I am working with currently supports 24-bit PCM sample in 32-bit data space but with LSB padded. Based on the discussion I found on line, SNDRV_PCM_FORMAT_S24_LE is defined to be 24-bit PCM sample in 32-bit data space with MSB padded. Also, there is no format macro defined for the format which my DSP supports. Please correct me if I am wrong. Furthermore, which software layer in audio stack of a typical linux distribution such as ubuntu handles the padding. I don't think it's handled by the alsa library. Does application or pulseaudio handle the padding? I would think it's done in pulseaudio. I have not study pulseaudio codec extensively but would think someone on the mailing list can give a quick answer. Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

2 1

[alsa-devel] [PATCH] davinci-mcasp: Add support for multichannel playback
by Michal Bachraty 27 Feb '13

27 Feb '13

Davinci McASP has support for I2S multichannel playback. For I2S playback/receive, each serializer is capable to play 2 channels (L/R) audio data.Serializer function (Playback-receive-none) is configured in DT, depending on hardware specification. It is possible to play less channels than configured in DT. For that purpose,only specific number of active serializers are enabled. McASP FIFO need to have DMA transfer Bcnt set to number of enabled serializers, otherwise no data are transfered to McASP and Alsa generates "DMA/IRQ playback write error (DMA or IRQ trouble?)" error. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 56 ++++++++++++++++++++++++++++++++----- sound/soc/davinci/davinci-pcm.c | 16 ++++++----- sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 59 insertions(+), 14 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d2ca682..9f33d90 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -235,6 +235,10 @@ #define DISMOD (val)(val<<2) #define TXSTATE BIT(4) #define RXSTATE BIT(5) +#define SRMOD_MASK 3 +#define SRMOD_INACTIVE 0 +#define SRMOD_TX 1 +#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -657,12 +661,15 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, return 0; } -static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, + int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; - + u8 ser; + u8 slots = dev->tdm_slots; + u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +687,42 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) } for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) + tx_ser++; + if (dev->serial_dir[i] == RX_MODE) + rx_ser++; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ser = tx_ser; + else + ser = rx_ser; + + if (ser < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + return -EINVAL; + } + + tx_ser = 0; + rx_ser = 0; + + for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) { + if (dev->serial_dir[i] == TX_MODE && + tx_ser < max_active_serializers) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE) { + } else if (dev->serial_dir[i] == RX_MODE && + rx_ser < max_active_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } @@ -729,6 +762,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } + + return 0; } static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) @@ -812,8 +847,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + u8 slots = dev->tdm_slots; + int channels; + struct snd_interval *pcm_channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels = pcm_channels->min; - davinci_hw_common_param(dev, substream->stream); + if (davinci_hw_common_param(dev, substream->stream, channels)) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +903,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_params->active_serializers = (channels + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +978,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bb57552..3af8b50 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,6 +182,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; + unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -195,14 +196,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level; + count /= fifo_level * serializers; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; - dst_bidx = 0; - src_cidx = data_type * fifo_level; + dst_bidx = 4; + src_cidx = data_type * fifo_level * serializers; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -210,7 +211,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level; + dst_cidx = data_type * fifo_level * serializers; } acnt = prtd->params->acnt; @@ -224,9 +225,10 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, - ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, + fifo_level * serializers, + count, fifo_level * serializers, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fbb710c..0d84d32 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,6 +27,7 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; + unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 10/12] ASoC: mxs: move to use generic DMA helper
by Shawn Guo 27 Feb '13

27 Feb '13

With the generic DMA device tree helper supported by mxs-dma driver, client devices only need to call dma_request_slave_channel() for requesting a DMA channel from dmaengine. Call snd_dmaengine_pcm_open() with NULL filter_fn, and having parameters needed by dma_request_slave_channel() wrap into snd_dma_channel_params and pass it in as filter_data, so that dmaengine_pcm will call the helper rather than dma_request_channel() to request DMA channel. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- .../devicetree/bindings/sound/mxs-saif.txt | 18 ++++---- sound/soc/mxs/mxs-pcm.c | 43 ++------------------ sound/soc/mxs/mxs-pcm.h | 5 --- sound/soc/mxs/mxs-saif.c | 28 ++----------- sound/soc/mxs/mxs-saif.h | 3 +- 5 files changed, 20 insertions(+), 77 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mxs-saif.txt b/Documentation/devicetree/bindings/sound/mxs-saif.txt index c37ba61..da221b4 100644 --- a/Documentation/devicetree/bindings/sound/mxs-saif.txt +++ b/Documentation/devicetree/bindings/sound/mxs-saif.txt @@ -3,8 +3,11 @@ Required properties: - compatible: Should be "fsl,<chip>-saif" - reg: Should contain registers location and length -- interrupts: Should contain ERROR and DMA interrupts -- fsl,saif-dma-channel: APBX DMA channel for the SAIF +- interrupts: Should contain ERROR interrupt number +- dmas: DMA specifier, consisting of a phandle to DMA controller node + and SAIF DMA channel ID. + Refer to dma.txt and fsl-mxs-dma.txt for details. +- dma-names: Must be "saif". Optional properties: - fsl,saif-master: phandle to the master SAIF. It's only required for @@ -23,14 +26,15 @@ aliases { saif0: saif@80042000 { compatible = "fsl,imx28-saif"; reg = <0x80042000 2000>; - interrupts = <59 80>; - fsl,saif-dma-channel = <4>; + interrupts = <59>; + dmas = <&dma_apbx 4>; + dma-names = "saif"; }; saif1: saif@80046000 { compatible = "fsl,imx28-saif"; reg = <0x80046000 2000>; - interrupts = <58 81>; - fsl,saif-dma-channel = <5>; - fsl,saif-master = <&saif0>; + interrupts = <58>; + dmas = <&dma_apbx 5>; + dma-names = "saif"; }; diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 564b5b6..74cb4b5 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -28,7 +28,6 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/dmaengine.h> -#include <linux/fsl/mxs-dma.h> #include <sound/core.h> #include <sound/initval.h> @@ -39,11 +38,6 @@ #include "mxs-pcm.h" -struct mxs_pcm_dma_data { - struct mxs_dma_data dma_data; - struct mxs_pcm_dma_params *dma_params; -}; - static struct snd_pcm_hardware snd_mxs_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -64,22 +58,6 @@ static struct snd_pcm_hardware snd_mxs_hardware = { }; -static bool filter(struct dma_chan *chan, void *param) -{ - struct mxs_pcm_dma_data *pcm_dma_data = param; - struct mxs_pcm_dma_params *dma_params = pcm_dma_data->dma_params; - - if (!mxs_dma_is_apbx(chan)) - return false; - - if (chan->chan_id != dma_params->chan_num) - return false; - - chan->private = &pcm_dma_data->dma_data; - - return true; -} - static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -91,35 +69,22 @@ static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, static int snd_mxs_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct mxs_pcm_dma_data *pcm_dma_data; + struct snd_dma_channel_params *params; int ret; - pcm_dma_data = kzalloc(sizeof(*pcm_dma_data), GFP_KERNEL); - if (pcm_dma_data == NULL) - return -ENOMEM; - - pcm_dma_data->dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - pcm_dma_data->dma_data.chan_irq = pcm_dma_data->dma_params->chan_irq; - - ret = snd_dmaengine_pcm_open(substream, filter, pcm_dma_data); - if (ret) { - kfree(pcm_dma_data); + params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + ret = snd_dmaengine_pcm_open(substream, NULL, params); + if (ret) return ret; - } snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware); - snd_dmaengine_pcm_set_data(substream, pcm_dma_data); - return 0; } static int snd_mxs_close(struct snd_pcm_substream *substream) { - struct mxs_pcm_dma_data *pcm_dma_data = snd_dmaengine_pcm_get_data(substream); - snd_dmaengine_pcm_close(substream); - kfree(pcm_dma_data); return 0; } diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index 35ba2ca..bc685b6 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -19,11 +19,6 @@ #ifndef _MXS_PCM_H #define _MXS_PCM_H -struct mxs_pcm_dma_params { - int chan_irq; - int chan_num; -}; - int mxs_pcm_platform_register(struct device *dev); void mxs_pcm_platform_unregister(struct device *dev); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 3a2aa1d..9f6abb8 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -26,7 +26,6 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/time.h> -#include <linux/fsl/mxs-dma.h> #include <linux/pinctrl/consumer.h> #include <sound/core.h> #include <sound/pcm.h> @@ -659,7 +658,7 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) static int mxs_saif_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct resource *iores, *dmares; + struct resource *iores; struct mxs_saif *saif; struct pinctrl *pinctrl; int ret = 0; @@ -721,22 +720,6 @@ static int mxs_saif_probe(struct platform_device *pdev) if (IS_ERR(saif->base)) return PTR_ERR(saif->base); - dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmares) { - /* - * TODO: This is a temporary solution and should be changed - * to use generic DMA binding later when the helplers get in. - */ - ret = of_property_read_u32(np, "fsl,saif-dma-channel", - &saif->dma_param.chan_num); - if (ret) { - dev_err(&pdev->dev, "failed to get dma channel\n"); - return ret; - } - } else { - saif->dma_param.chan_num = dmares->start; - } - saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { ret = saif->irq; @@ -753,13 +736,8 @@ static int mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->dma_param.chan_irq = platform_get_irq(pdev, 1); - if (saif->dma_param.chan_irq < 0) { - ret = saif->dma_param.chan_irq; - dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", - ret); - return ret; - } + saif->dma_param.dev = saif->dev; + saif->dma_param.name = "saif"; platform_set_drvdata(pdev, saif); diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 3cb342e..32b4310 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -108,6 +108,7 @@ #define MXS_SAIF_MCLK 0 +#include <sound/dmaengine_pcm.h> #include "mxs-pcm.h" struct mxs_saif { @@ -117,7 +118,7 @@ struct mxs_saif { unsigned int mclk_in_use; void __iomem *base; int irq; - struct mxs_pcm_dma_params dma_param; + struct snd_dma_channel_params dma_param; unsigned int id; unsigned int master_id; unsigned int cur_rate; -- 1.7.9.5

1 0

[alsa-devel] [PATCH] davinci-mcasp: Add support for multichannel playback
by Michal Bachraty 27 Feb '13

27 Feb '13

Davinci McASP has support for I2S multichannel playback. For I2S playback/receive, each serializer is capable to play 2 channels (L/R) audio data.Serializer function (Playback-receive-none) is configured in DT, depending on hardware specification. It is possible to play less channels than configured in DT. For that purpose,only specific number of active serializers are enabled. McASP FIFO need to have DMA transfer Bcnt set to number of enabled serializers, otherwise no data are transfered to McASP and Alsa generates "DMA/IRQ playback write error (DMA or IRQ trouble?)" error. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 49 +++++++++++++++++++++++++++++++------ sound/soc/davinci/davinci-pcm.c | 15 ++++++------ sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 51 insertions(+), 14 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d2ca682..ebd6ee0 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -657,12 +657,14 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, return 0; } -static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, + int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; - + u8 ser; + u8 stream_serializers = channels/2 + channels%2; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +682,44 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) } for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) + tx_ser++; + if (dev->serial_dir[i] == RX_MODE) + rx_ser++; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + ser = tx_ser; + } else { + ser = rx_ser; + } + + if (ser < stream_serializers) { + printk(KERN_WARNING "davinci-mcasp: stream has more channels " + "(%d) than are enabled in mcasp (%d)", + channels, ser * 2); + return -EINVAL; + } + + tx_ser = 0; + rx_ser = 0; + + for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) { + if (dev->serial_dir[i] == TX_MODE && + tx_ser < stream_serializers) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE) { + } else if (dev->serial_dir[i] == RX_MODE && + rx_ser < stream_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + 3); } } @@ -812,8 +842,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + int channels; + struct snd_interval *pcm_channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels = pcm_channels->min; - davinci_hw_common_param(dev, substream->stream); + davinci_hw_common_param(dev, substream->stream, channels); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +896,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_params->active_serializers = channels/2 + channels%2; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +971,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bb57552..9951973 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,6 +182,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; + unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -193,7 +194,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) period_size); data_type = prtd->params->data_type; - count = period_size / data_type; + count = period_size / (data_type * serializers); if (fifo_level) count /= fifo_level; @@ -201,8 +202,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; - dst_bidx = 0; - src_cidx = data_type * fifo_level; + dst_bidx = 4; + src_cidx = data_type * fifo_level * serializers; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -210,7 +211,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level; + dst_cidx = data_type * fifo_level * serializers; } acnt = prtd->params->acnt; @@ -224,9 +225,9 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, - ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, serializers, + count, fifo_level * serializers, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fbb710c..0d84d32 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,6 +27,7 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; + unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

2 2

[alsa-devel] [PATCH v4 0/3] ASoC davinci/tlv320aic3x updates, fixes & DT support
by Hebbar Gururaja 27 Feb '13

27 Feb '13

1. Convert tlv320aic3x mic bias to a supply widget and related machine driver 2. Add DT support for Davinci machine platform 3. Remove __dev* attributes This patch-set is tested on Davinci platform (DA850 EVM). This series applies on top of tag next-20130128 git tree https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git These patches cannot be tested directly on mainline kernel (DT based). This is because dma is not yet completely ported to Davinci ASoC. I had to add few hacks to get Audio working on Davinci DA850 platform. For reference, I have placed the entire working tree @ Tree : https://github.com/ghebbar/linux.git Branch : dev_next-20130128_davinci_asoc_dt_v3 Changes in v4 - updated pending machine driver to change in micbias -> supply widget - merge codec & machine driver change into a single commit Changes in v3 - drop mcasp pinctrl handler patch from series as it is now handled centrally. - Convert tlv320aic3x mic bias to a supply widget and related machine driver - remove __dev* attributes Changes in v2 - Remove reference to Linux & software details from DT binding Hebbar Gururaja (2): ASoC: tlv320aic3x: Convert mic bias to a supply widget ASoC: davinci: remove __dev* attributes Hebbar, Gururaja (1): ASoC: davinci: machine: Add device tree binding .../bindings/sound/davinci-evm-audio.txt | 53 ++++++ .../devicetree/bindings/sound/tlv320aic3x.txt | 6 + include/sound/tlv320aic3x.h | 10 ++ sound/soc/codecs/tlv320aic3x.c | 83 ++++++++- sound/soc/codecs/tlv320aic3x.h | 4 + sound/soc/davinci/davinci-evm.c | 185 +++++++++++++++++--- sound/soc/omap/n810.c | 4 +- sound/soc/omap/rx51.c | 8 +- 8 files changed, 318 insertions(+), 35 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt -- 1.7.9.5

3 5

[alsa-devel] [PATCH 1/2] ASoC: wm5102: Correct AEC loopback enable definiton
by Mark Brown 26 Feb '13

26 Feb '13

Reported-by: Ryo Tsutsui <Ryo.Tsutsui(a)wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index c1ea75c..efd7317 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1083,7 +1083,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5102_aec_loopback_mux), SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, -- 1.7.10.4

1 4

[alsa-devel] [PATCH - alsa-lib/mixer 1/1] snd_mixer_poll_descriptors_revents should loop over pollfd
by s.jansen＠gmail.com 26 Feb '13

26 Feb '13

From: Sander Jansen <s.jansen(a)gmail.com> Signed-off-by: Sander Jansen <s.jansen(a)gmail.com> diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 85d843f..8deceac 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -743,7 +743,7 @@ int snd_mixer_poll_descriptors_revents(snd_mixer_t *mixer, struct pollfd *pfds, if (nfds == 0) return -EINVAL; res = 0; - for (idx = 0; idx < nfds; idx++) + for (idx = 0; idx < nfds; idx++, pfds++) res |= pfds->revents & (POLLIN|POLLERR|POLLNVAL); *revents = res; return 0; -- 1.8.1.3

2 1

[alsa-devel] [PATCH] Fix: pulse_pcm_poll_revents should return 0 on success
by s.jansen＠gmail.com 26 Feb '13

26 Feb '13

From: Sander Jansen <s.jansen(a)gmail.com> --- pulse/pcm_pulse.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index 4226d30..0d3704b 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -651,6 +651,8 @@ static int pulse_pcm_poll_revents(snd_pcm_ioplug_t * io, else *revents = 0; + err = 0; + finish: pa_threaded_mainloop_unlock(pcm->p->mainloop); -- 1.8.1.4

1 0

[alsa-devel] [PATCH] pulse_pcm_poll_revents should return 0 on success
by s.jansen＠gmail.com 26 Feb '13

26 Feb '13

From: Sander Jansen <s.jansen(a)gmail.com> According to the DOCS, poll_revents should return 0 on success. Fix the pulse plugin to do so. Sander Jansen (1): Fix: pulse_pcm_poll_revents should return 0 on success pulse/pcm_pulse.c | 2 ++ 1 file changed, 2 insertions(+) -- 1.8.1.4

1 0

[alsa-devel] [PATCH] ASoC: dapm: Provide early event callbacks for power up and down
by Mark Brown 26 Feb '13

26 Feb '13

Some devices may benefit from being able to start some parts of the widget power up/down sequence earlier on in the sequence than the point at which the final power state is committed. Support these by providing events which are called before any power state changes are done. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 2 ++ sound/soc/soc-dapm.c | 16 ++++++++++++++++ 2 files changed, 18 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 38fd652..7e4efa5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -311,6 +311,8 @@ struct device; #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ +#define SND_SOC_DAPM_WILL_PMU 0x40 /* called at start of sequence */ +#define SND_SOC_DAPM_WILL_PMD 0x80 /* called at start of sequence */ #define SND_SOC_DAPM_PRE_POST_PMD \ (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 11047ba..b69e9de 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1255,6 +1255,14 @@ static void dapm_seq_check_event(struct snd_soc_dapm_context *dapm, ev_name = "POST_PMD"; power = 0; break; + case SND_SOC_DAPM_WILL_PMU: + ev_name = "WILL_PMU"; + power = 1; + break; + case SND_SOC_DAPM_WILL_PMD: + ev_name = "WILL_PMD"; + power = 0; + break; default: BUG(); return; @@ -1715,6 +1723,14 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) &async_domain); async_synchronize_full_domain(&async_domain); + list_for_each_entry(w, &down_list, list) { + dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMD); + } + + list_for_each_entry(w, &up_list, list) { + dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMU); + } + /* Power down widgets first; try to avoid amplifying pops. */ dapm_seq_run(dapm, &down_list, event, false); -- 1.7.10.4

2 1

[alsa-devel] [PATCH] Allow to switch hardware sampe rate on EMU
by Mihail Zenkov 25 Feb '13

25 Feb '13

Commit 57e5c63007955838043e34c732d224b2cbbb128f "emu10k1: allow to disable the SRC" force hardware use only one rate (48000 hz), but EMU have support two hardware sampling rates (44100 and 48000 hz). Signed-off-by: Mihail Zenkov <mihail.zenkov(a)gmail.com> --- linux-3.8/sound/pci/emu10k1/emupcm.c +++ linux-3.8-1/sound/pci/emu10k1/emupcm.c @@ -1127,7 +1127,7 @@ struct snd_emu10k1_pcm *epcm; struct snd_emu10k1_pcm_mixer *mix; struct snd_pcm_runtime *runtime = substream->runtime; - int i, err; + int i, err, sample_rate; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); if (epcm == NULL) @@ -1146,7 +1146,11 @@ kfree(epcm); return err; } - err = snd_pcm_hw_rule_noresample(runtime, 48000); + if (emu->card_capabilities->emu_model && emu->emu1010.internal_clock == 0) + sample_rate = 44100; + else + sample_rate = 48000; + err = snd_pcm_hw_rule_noresample(runtime, sample_rate); if (err < 0) { kfree(epcm); return err;

3 6

[alsa-devel] M-Audio Audiophile 192 (ice1724)'s broken spdif capture
by Jonas Petersen 25 Feb '13

25 Feb '13

Hi there, I'm trying to get spdif capture to work properly on the M-Audio Audiophile 192 (VT1724/Envy24HT). It is generally working, but I don't get a clean signal. I have "Multi Track Internal Clock" set to "IEC958 In" (spdif in). When I capture via Jack, arecord or Audacity I experience the following two phenomenons (always): 1.) The captured signal ends up 6 dB(FS) to loud (200%)! Everything above -6 dB is distorting. A 1 kHz sine test signal at -12 dB (25%) will be captured as -6 dB (50%). 2.) The left and right channels are shifted by one sample! When I feed a 1 kHz test signal (L+R are _exactly_ the same signal), the right channel will be offset by exactly one sample. Zooming into the waveform clearly shows that. Analyzing the signal with a goniometer shows a (slight but obvious) vertical eliptical shape and not the expected single vertical line. To make sure the hardware actually works fine, I did install a Windows 7 on the exact same machine, installed the drivers from M-Audio and did some recording with audacity. The result is as expected: the -12 dB signal ends up as -12 dB and the left and right channel exactly match each other. So the hardware is willing. I am running a Lubuntu 12.10 and I'm am able to compile and run the current alsa-driver source with the 3.5.0-22 kernel. I played around a bit with the alsa driver source (e.g. pci/ice1712/ice1724.c, pci/ice1712/revo.c) and I am able to compile and load a modified driver. So far I only was able to make the problem worse though. I'd now like to ask for some advice on how to approach the problem. I guess the fact that the left and right channel differ - even though they shouldn't - might be a thing to look for. This must be happening at some stage in the capturing. Is there a way to hook in at different places to narrow down what causes this? Maybe this even already rings somebody's bells? I'll be glad to deliver more information when needed. Thanks and regards, Jonas

3 29

[alsa-devel] [PATCH 1/1] M-Audio Audiophile192: Fix SPDIF input
by Jonas Petersen 25 Feb '13

25 Feb '13

This patch fixes initialization of the AK4114 chip so spdif capture is working properly. Worked out together with Pavel Hofman. Signed-off-by: Jonas Petersen <jnsptrsn1(a)gmail.com> diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 7641080..1112ec1 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -35,6 +35,7 @@ struct revo51_spec { struct snd_i2c_device *dev; struct snd_pt2258 *pt2258; + struct ak4114 *ak4114; }; static void revo_i2s_mclk_changed(struct snd_ice1712 *ice) @@ -359,9 +360,9 @@ static struct snd_ak4xxx_private akm_ap192_priv = { .cif = 0, .data_mask = VT1724_REVO_CDOUT, .clk_mask = VT1724_REVO_CCLK, - .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1, - .cs_addr = VT1724_REVO_CS1, - .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1, + .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS3, + .cs_addr = VT1724_REVO_CS3, + .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS3, .add_flags = VT1724_REVO_CCLK, /* high at init */ .mask_flags = 0, }; @@ -372,7 +373,7 @@ static struct snd_ak4xxx_private akm_ap192_priv = { * CCLK (pin 34) -- GPIO1 pin 51 (shared with AK4358) * CSN (pin 35) -- GPIO7 pin 59 */ -#define AK4114_ADDR 0x02 +#define AK4114_ADDR 0x00 static void write_data(struct snd_ice1712 *ice, unsigned int gpio, unsigned int data, int idx) @@ -426,7 +427,7 @@ static unsigned int ap192_4wire_start(struct snd_ice1712 *ice) tmp = snd_ice1712_gpio_read(ice); tmp |= VT1724_REVO_CCLK; /* high at init */ tmp |= VT1724_REVO_CS0; - tmp &= ~VT1724_REVO_CS1; + tmp &= ~VT1724_REVO_CS3; snd_ice1712_gpio_write(ice, tmp); udelay(1); return tmp; @@ -434,7 +435,7 @@ static unsigned int ap192_4wire_start(struct snd_ice1712 *ice) static void ap192_4wire_finish(struct snd_ice1712 *ice, unsigned int tmp) { - tmp |= VT1724_REVO_CS1; + tmp |= VT1724_REVO_CS3; tmp |= VT1724_REVO_CS0; snd_ice1712_gpio_write(ice, tmp); udelay(1); @@ -470,27 +471,32 @@ static unsigned char ap192_ak4114_read(void *private_data, unsigned char addr) static int ap192_ak4114_init(struct snd_ice1712 *ice) { static const unsigned char ak4114_init_vals[] = { - AK4114_RST | AK4114_PWN | AK4114_OCKS0 | AK4114_OCKS1, + AK4114_RST | AK4114_PWN | AK4114_OCKS0, AK4114_DIF_I24I2S, AK4114_TX1E, - AK4114_EFH_1024 | AK4114_DIT | AK4114_IPS(1), + AK4114_EFH_1024 | AK4114_DIT | AK4114_IPS(0), 0, 0 }; static const unsigned char ak4114_init_txcsb[] = { 0x41, 0x02, 0x2c, 0x00, 0x00 }; - struct ak4114 *ak; int err; + struct revo51_spec *spec; + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + err = snd_ak4114_create(ice->card, ap192_ak4114_read, ap192_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, - ice, &ak); + ice, &spec->ak4114); /* AK4114 in Revo cannot detect external rate correctly. * No reason to stop capture stream due to incorrect checks */ - ak->check_flags = AK4114_CHECK_NO_RATE; + spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; return 0; /* error ignored; it's no fatal error */ } @@ -562,6 +568,9 @@ static int revo_init(struct snd_ice1712 *ice) ice); if (err < 0) return err; + err = ap192_ak4114_init(ice); + if (err < 0) + return err; /* unmute all codecs */ snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE, @@ -575,7 +584,7 @@ static int revo_init(struct snd_ice1712 *ice) static int revo_add_controls(struct snd_ice1712 *ice) { - struct revo51_spec *spec; + struct revo51_spec *spec = ice->spec; int err; switch (ice->eeprom.subvendor) { @@ -597,7 +606,9 @@ static int revo_add_controls(struct snd_ice1712 *ice) err = snd_ice1712_akm4xxx_build_controls(ice); if (err < 0) return err; - err = ap192_ak4114_init(ice); + /* only capture SPDIF over AK4114 */ + err = snd_ak4114_build(spec->ak4114, NULL, + ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); if (err < 0) return err; break; -- 1.7.10.4

2 1

Re: [alsa-devel] [PATCH] ALSA: emu10k1: Load firmware when it was already cached
by Takashi Iwai 25 Feb '13

25 Feb '13

At Sat, 23 Feb 2013 15:36:29 +0100, Florian Zeitz wrote: > > This expands the regression fix from > d28215996b0c3a900411769039aa3c54cf7008ab. > The firmware also needs to be loaded when it was already cached. > > Signed-off-by: Florian Zeitz <florob(a)babelmonkeys.de> Thanks, applied now. Takashi > --- > sound/pci/emu10k1/emu10k1_main.c | 11 ++++++----- > 1 file changed, 6 insertions(+), 5 deletions(-) > > > [2 0001-ALSA-emu10k1-Load-firmware-when-it-was-already-cache.patch <text/x-patch (7bit)>] > diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c > index 5115f11..e6b0166 100644 > --- a/sound/pci/emu10k1/emu10k1_main.c > +++ b/sound/pci/emu10k1/emu10k1_main.c > @@ -860,11 +860,12 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) > } > snd_printk(KERN_INFO "emu1010: firmware file = %s, size = 0x%zx\n", > filename, emu->firmware->size); > - err = snd_emu1010_load_firmware(emu); > - if (err != 0) { > - snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", filename); > - return err; > - } > + } > + > + err = snd_emu1010_load_firmware(emu); > + if (err != 0) { > + snd_printk(KERN_INFO "emu1010: Loading Firmware failed\n"); > + return err; > } > > /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ >

1 0

[alsa-devel] Bug#701498: hda: Internal mic does not work on Thinkpad T420s (CX20590 codec)
by Ben Hutchings 25 Feb '13

25 Feb '13

On Sat, 2013-02-23 at 20:42 +0100, Adrian Lang wrote: > Package: src:linux > Version: 3.2.35-2 > Severity: normal > > There are two microphones in alsamixer and Gnome 3 audio settings (i. e., > pulseaudio), but I can't get any sound using either of them. I think that's the > same problem as reported to [openSUSE] and [Ubuntu]. > > The problem persists with 3.7 (3.7.8-1~experimental.1) from > experimental. > > [openSUSE] > http://forums.opensuse.org/english/get-technical-help-here/laptop/469210-op… > [Ubuntu] > https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/938644 I happen to have a Thinkpad T420 with the same audio codec, and the internal mic works fine. (There does seem to be a problem with external mics, now that I check, but let's ignore that for now.) So this is more board-specific, not chip-specific, behaviour. [...] > ** Model information > sys_vendor: LENOVO > product_name: 4171CTO > product_version: ThinkPad T420s [...] Does the T420s also have a single socket that can be used with either headphones or headset? [I'm copying your alsa-info.txt as an attachment for the benefit of alsa-devel.] Ben. -- Ben Hutchings Absolutum obsoletum. (If it works, it's out of date.) - Stafford Beer

3 2

[alsa-devel] [PATCH 1/5] ASoC: max98090: Remove unneeded version.h header include
by Sachin Kamat 25 Feb '13

25 Feb '13

version.h header file inclusion is not required as detected by versioncheck. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- All patches in this series are build tested and based on for-next branch of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tree. --- sound/soc/codecs/max98090.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index fc17604..9ea73aa 100755 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -23,8 +23,6 @@ #include <sound/max98090.h> #include "max98090.h" -#include <linux/version.h> - #define DEBUG #define EXTMIC_METHOD #define EXTMIC_METHOD_TEST -- 1.7.4.1

2 10

[alsa-devel] [ANNOUNCE]: tinycompress: release 0.1.0
by Vinod Koul 24 Feb '13

24 Feb '13

Hi, tinycompress library - the user mode library for using the ALSA compressed APIs has seen some good testing, which brings few fixes and new feature for gapless playback. This library supports both Android and Linux OSes. Along with we have a player, cplay- which can be used for mp3 offload playback Number of fixes and cleanups along the way as seen by below changelog and diffstat. So time to bump up the version to 0.1.0 Charles Keepax (4): compress: Extend limit on error messages compress: Return error when failing to allocate compress struct compress: Return error messages correctly from compress_open compress: Do not put newlines on error messages Namarta Kohli (1): compress: add compress_resume Richard Fitzgerald (12): compress: Add function to get timestamp in samples compress: Change compress_write() to take buffer as const void * compress: Remove unused buffer_size member from struct compress compress: Change running from bitfield to plain int compress: Fix incorrect error message on compress_resume() compress: Remove hardcoded limit on length of poll() wait compress: Handle case of pause() during compress_write() compress: Allow client to pass "don't care" for fragment size/number compress: Block if unable to write all data in single write() compress: Add missing header includes compress: Add makefile for building under standard linux cplay: support auto-configuration of fragment size and count Vinod Koul (9): initial creation compress: fix the timestamp calculation compress: write when we have a fragment to write compress: make variable 'running' as unsigned compress: remove unused variables compress: cache the config values passed tinycompress: add gapless meta data APIs tinycompres: remove unused variable tinycompress: version 0.1.0 Xuemin Su (1): compress: fix arithmetic exception in compress_get_hpointer Android.mk | 22 ++ README | 31 ++- compress.c | 539 +++++++++++++++++++++++++++++++++++ cplay.c | 368 ++++++++++++++++++++++++ include/sound/compress_offload.h | 95 ++++++ include/sound/compress_params.h | 230 +++++++++++++++ include/tinycompress/tinycompress.h | 270 ++++++++++++++++++ include/tinycompress/tinymp3.h | 106 +++++++ include/tinycompress/version.h | 70 +++++ makefile.linux | 35 +++ 10 files changed, 1765 insertions(+), 1 deletions(-) create mode 100644 Android.mk create mode 100644 compress.c create mode 100644 cplay.c create mode 100644 include/sound/compress_offload.h create mode 100644 include/sound/compress_params.h create mode 100644 include/tinycompress/tinycompress.h create mode 100644 include/tinycompress/tinymp3.h create mode 100644 include/tinycompress/version.h create mode 100644 makefile.linux -- ~Vinod

1 0

[alsa-devel] [PATCH] tinycompres: remove unused variable
by Vinod Koul 24 Feb '13

24 Feb '13

cplay.c:210: warning: unused variable i Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- cplay.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/cplay.c b/cplay.c index e52d641..5b4e83e 100644 --- a/cplay.c +++ b/cplay.c @@ -207,7 +207,7 @@ void play_samples(char *name, unsigned int card, unsigned int device, FILE *file; char *buffer; int size, num_read, wrote; - unsigned int i, channels, rate, bits; + unsigned int channels, rate, bits; if (verbose) printf("%s: entry\n", __func__); -- 1.7.0.4

1 1

[alsa-devel] [PATCH] tinycompress: add gapless meta data APIs
by Vinod Koul 24 Feb '13

24 Feb '13

Based on newly introduced kernel API, we add gapless metatdata to the library. Since kernel can recieve only key/value pairs for metadata we add a structure for gapless_metdata and handle this in library Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- compress.c | 71 +++++++++++++++++++++++++++++++++++ include/sound/compress_offload.h | 18 ++++++++- include/tinycompress/tinycompress.h | 37 ++++++++++++++++++ include/tinycompress/version.h | 4 +- 4 files changed, 126 insertions(+), 4 deletions(-) diff --git a/compress.c b/compress.c index d44d5fe..5b600c3 100644 --- a/compress.c +++ b/compress.c @@ -85,6 +85,8 @@ struct compress { char error[COMPR_ERR_MAX]; struct compr_config *config; unsigned int running:1; + unsigned int gapless_metadata; + unsigned int next_track; }; static int oops(struct compress *compress, int e, const char *fmt, ...) @@ -121,6 +123,17 @@ int is_compress_ready(struct compress *compress) return (compress->fd > 0) ? 1 : 0; } +static int get_compress_version(struct compress *compress) +{ + int version = 0; + + if (ioctl(compress->fd, SNDRV_COMPRESS_IOCTL_VERSION, &version)) { + oops(compress, errno, "cant read version"); + return -1; + } + return version; +} + static bool _is_codec_supported(struct compress *compress, struct compr_config *config) { struct snd_compr_caps caps; @@ -212,6 +225,8 @@ struct compress *compress_open(unsigned int card, unsigned int device, return &bad_compress; } + compress->next_track = 0; + compress->gapless_metadata = 0; compress->config = calloc(1, sizeof(*config)); if (!compress->config) goto input_fail; @@ -401,6 +416,62 @@ int compress_drain(struct compress *compress) return 0; } +int compress_partial_drain(struct compress *compress) +{ + if (!is_compress_running(compress)) + return oops(compress, -ENODEV, "device not ready"); + + if (!compress->next_track) + return oops(compress, -EPERM, "next track not signalled"); + if (ioctl(compress->fd, SNDRV_COMPRESS_PARTIAL_DRAIN)) + return oops(compress, errno, "cannot drain the stream\n"); + compress->next_track = 0; + return 0; +} + +int compress_next_track(struct compress *compress) +{ + if (!is_compress_running(compress)) + return oops(compress, -ENODEV, "device not ready"); + + if (!compress->gapless_metadata) + return oops(compress, -EPERM, "metadata not set"); + if (ioctl(compress->fd, SNDRV_COMPRESS_NEXT_TRACK)) + return oops(compress, errno, "cannot set next track\n"); + compress->next_track = 1; + compress->gapless_metadata = 0; + return 0; +} + +int compress_set_gapless_metadata(struct compress *compress, + struct compr_gapless_mdata *mdata) +{ + struct snd_compr_metadata metadata; + int version; + + if (!is_compress_ready(compress)) + return oops(compress, -ENODEV, "device not ready"); + + version = get_compress_version(compress); + if (version <= 0) + return -1; + + if (version < SNDRV_PROTOCOL_VERSION(0, 1, 1)) + return oops(compress, -ENXIO, "gapless apis not supported in kernel"); + + metadata.key = SNDRV_COMPRESS_ENCODER_PADDING; + metadata.value[0] = mdata->encoder_padding; + if (ioctl(compress->fd, SNDRV_COMPRESS_SET_METADATA, &metadata)) + return oops(compress, errno, "can't set metadata for stream\n"); + + metadata.key = SNDRV_COMPRESS_ENCODER_DELAY; + metadata.value[0] = mdata->encoder_delay; + if (ioctl(compress->fd, SNDRV_COMPRESS_SET_METADATA, &metadata)) + return oops(compress, errno, "can't set metadata for stream\n"); + compress->gapless_metadata = 1; + return 0; +} + bool is_codec_supported(unsigned int card, unsigned int device, unsigned int flags, struct snd_codec *codec) { diff --git a/include/sound/compress_offload.h b/include/sound/compress_offload.h index bbb09bc..17377df 100644 --- a/include/sound/compress_offload.h +++ b/include/sound/compress_offload.h @@ -16,7 +16,7 @@ #include <sound/asound.h> //#include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) struct snd_compressed_buffer { __u32 fragment_size; @@ -64,11 +64,23 @@ struct snd_compr_codec_caps { struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS]; }; +enum { + SNDRV_COMPRESS_ENCODER_PADDING = 1, + SNDRV_COMPRESS_ENCODER_DELAY = 2, +}; + +struct snd_compr_metadata { + __u32 key; + __u32 value[8]; +}; + #define SNDRV_COMPRESS_IOCTL_VERSION _IOR('C', 0x00, int) #define SNDRV_COMPRESS_GET_CAPS _IOWR('C', 0x10, struct snd_compr_caps) #define SNDRV_COMPRESS_GET_CODEC_CAPS _IOWR('C', 0x11, struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14, struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOWR('C', 0x15, struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -76,6 +88,8 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) - +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) #define SND_COMPR_TRIGGER_DRAIN 7 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 8 #endif diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 13dea3e..e129817 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -69,6 +69,11 @@ struct compr_config { struct snd_codec *codec; }; +struct compr_gapless_mdata { + __u32 encoder_delay; + __u32 encoder_padding; +}; + #define COMPRESS_OUT 0x00000000 #define COMPRESS_IN 0x10000000 @@ -165,6 +170,38 @@ int compress_resume(struct compress *compress); int compress_drain(struct compress *compress); /* + * compress_next_track: set the next track for stream + * + * return 0 on success, negative on error + * + * @compress: compress stream to be transistioned to next track + */ +int compress_next_track(struct compress *compress); + +/* + * compress_partial_drain: drain will return after the last frame is decoded + * by DSP and will play the , All the data written into compressed + * ring buffer is decoded + * + * return 0 on success, negative on error + * + * @compress: compress stream to be drain + */ +int compress_partial_drain(struct compress *compress); + +/* + * compress_set_gapless_metadata: set gapless metadata of a compress strem + * + * return 0 on success, negative on error + * + * @compress: compress stream for which metadata has to set + * @mdata: metadata encoder delay and padding + */ + +int compress_set_gapless_metadata(struct compress *compress, + struct compr_gapless_mdata *mdata); + +/* * is_codec_supported:check if the given codec is supported * returns true when supported, false if not * diff --git a/include/tinycompress/version.h b/include/tinycompress/version.h index 89f0f4e..db191d6 100644 --- a/include/tinycompress/version.h +++ b/include/tinycompress/version.h @@ -56,7 +56,7 @@ #define TINYCOMPRESS_LIB_MAJOR 0 /* major number of library version */ #define TINYCOMPRESS_LIB_MINOR 0 /* minor number of library version */ -#define TINYCOMPRESS_LIB_SUBMINOR 1 /* subminor number of library version */ +#define TINYCOMPRESS_LIB_SUBMINOR 2 /* subminor number of library version */ /** library version */ #define TINYCOMPRESS_LIB_VERSION \ @@ -65,6 +65,6 @@ TINYCOMPRESS_LIB_SUBMINOR) /** library version (string) */ -#define TINYCOMPRESS_LIB_VERSION_STR "0.0.1" +#define TINYCOMPRESS_LIB_VERSION_STR "0.0.2" #endif -- 1.7.0.4

2 3

[alsa-devel] [PATCH] ASoC: dapm: Fix handling of loops
by Mark Brown 22 Feb '13

22 Feb '13

Currently if a path loops back on itself we correctly skip over it to avoid going into an infinite loop but this causes us to ignore the need to power up the path as we don't count the loop for the purposes of counting inputs and outputs. This means that internal loopbacks within a device that have powered devices on them won't be powered up. Fix this by treating any path that is currently in the process of being recursed as having a single input or output so that it is counted for the purposes of power decisions. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 1 + sound/soc/soc-dapm.c | 15 +++++++++++++++ 2 files changed, 16 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 617ac36..38fd652 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -491,6 +491,7 @@ struct snd_soc_dapm_path { /* status */ u32 connect:1; /* source and sink widgets are connected */ u32 walked:1; /* path has been walked */ + u32 walking:1; /* path is in the process of being walked */ u32 weak:1; /* path ignored for power management */ int (*connected)(struct snd_soc_dapm_widget *source, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1dc5b72..11047ba 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -821,6 +821,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources))) { widget->outputs = snd_soc_dapm_suspend_check(widget); + path->walking = 0; return widget->outputs; } } @@ -831,6 +832,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -838,6 +842,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->sink && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -847,11 +852,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_output_ep(path->sink, list); + + path->walking = 0; } } @@ -931,6 +939,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -938,6 +949,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->source && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -947,11 +959,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_input_ep(path->source, list); + + path->walking = 0; } } -- 1.7.10.4

2 1

[alsa-devel] [PATCH TINYCOMPRESS 1/14] compress: Add function to get timestamp in samples
by Richard Fitzgerald 22 Feb '13

22 Feb '13

The compress_get_hpointer() converts the timestamp into actual time. But Android needs it in samples. To avoid the inefficiency of using compress_get_hpointer() and converting into time and then back into samples, this change adds compress_get_tstamp() which returns the raw struct snd_compr_tstamp. diff --git a/compress.c b/compress.c index faef982..19b9366 100644 --- a/compress.c +++ b/compress.c @@ -287,6 +287,18 @@ int compress_get_hpointer(struct compress *compress, return 0; } +int compress_get_tstamp(struct compress *compress, + struct snd_compr_tstamp *tstamp ) +{ + if (!is_compress_ready(compress)) + return oops(compress, -ENODEV, "device not ready"); + + if (ioctl(compress->fd, SNDRV_COMPRESS_TSTAMP, tstamp)) + return oops(compress, errno, "cannot get avail"); + + return 0; +} + int compress_write(struct compress *compress, char *buf, unsigned int size) { struct snd_compr_avail avail; diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 13dea3e..8a98527 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -73,6 +73,7 @@ struct compr_config { #define COMPRESS_IN 0x10000000 struct compress; +struct snd_compr_tstamp; /* * compress_open: open a new compress stream @@ -103,6 +104,18 @@ void compress_close(struct compress *compress); */ int compress_get_hpointer(struct compress *compress, unsigned int *avail, struct timespec *tstamp); + + +/* + * compress_get_tstamp: get the raw hw timestamp + * return 0 on success, negative on error + * + * @compress: compress stream on which query is made + * @tstamp: timestamp information + */ +int compress_get_tstamp(struct compress *compress, + struct snd_compr_tstamp *tstamp ); + /* * compress_write: write data to the compress stream * return bytes written on success, negative on error -- 1.7.2.5

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