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

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; }

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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 = > >

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[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.

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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:

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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

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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>

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[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

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[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

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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...

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[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

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