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

[alsa-devel] [PATCH] ALSA: hda - fix inverted internal mic on Acer AOA150/ZG5
by David Henningsson 28 Jan '13

28 Jan '13

This patch enables internal mic input on the machine. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1107477 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dc75607..de512c5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2363,6 +2363,7 @@ static const struct hda_model_fixup alc268_fixup_models[] = { }; static const struct snd_pci_quirk alc268_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x015b, "Acer AOA 150 (ZG5)", ALC268_FIXUP_INV_DMIC), /* below is codec SSID since multiple Toshiba laptops have the * same PCI SSID 1179:ff00 */ -- 1.7.9.5

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Re: [alsa-devel] [PATCH 3/3] ALSA: intel8x0: Check return value before assigning
by Takashi Iwai 28 Jan '13

28 Jan '13

At Sun, 27 Jan 2013 17:40:02 +0100, Emil Goode wrote: > > The first patch in this series changes the return type of function > kvm_para_available to bool. In the function snd_intel8x0_inside_vm > we now need to check it's return value before assigning a value to > the result variable. > > Signed-off-by: Emil Goode <emilgoode(a)gmail.com> > --- > sound/pci/intel8x0.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c > index 3b9be75..fa9337e 100644 > --- a/sound/pci/intel8x0.c > +++ b/sound/pci/intel8x0.c > @@ -2982,7 +2982,7 @@ static int snd_intel8x0_inside_vm(struct pci_dev *pci) > } > > /* detect KVM and Parallels virtual environments */ > - result = kvm_para_available(); > + result = kvm_para_available() ? 1 : 0; This is superfluous. The conversion from bool to int works implicitly like that. Takashi

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[alsa-devel] [PATCH 1/2] Revert "ASoC: fsl: fix multiple definition of init_module"
by Shawn Guo 28 Jan '13

28 Jan '13

This reverts commit 25b8d31488a3fb3611651991969526b2ea475764. While the commit fixes multiple init_module definition error with module build, it breaks build when both imx-pcm-fiq and imx-pcm-dma are built in as below. LD sound/soc/fsl/snd-soc-fsl-ssi.o LD sound/soc/fsl/snd-soc-fsl-utils.o LD sound/soc/fsl/snd-soc-imx-ssi.o LD sound/soc/fsl/snd-soc-imx-audmux.o LD sound/soc/fsl/snd-soc-imx-pcm-fiq.o LD sound/soc/fsl/snd-soc-imx-pcm-dma.o LD sound/soc/fsl/snd-soc-eukrea-tlv320.o LD sound/soc/fsl/snd-soc-imx-sgtl5000.o LD sound/soc/fsl/snd-soc-imx-mc13783.o LD sound/soc/fsl/built-in.o sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `imx_pcm_free': imx-pcm.c:(.text+0x464): multiple definition of `imx_pcm_free' sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0x1a8): first defined here sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `snd_imx_pcm_mmap': imx-pcm.c:(.text+0x35c): multiple definition of `snd_imx_pcm_mmap' sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0xa0): first defined here sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `imx_pcm_new': imx-pcm.c:(.text+0x3dc): multiple definition of `imx_pcm_new' sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0x120): first defined here make[4]: *** [sound/soc/fsl/built-in.o] Error 1 Let's revert the commit and find a proper fix for multiple init_module definition error later. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/fsl/Kconfig | 9 +++++++-- sound/soc/fsl/Makefile | 5 +---- sound/soc/fsl/imx-pcm.c | 3 +++ 3 files changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index a210c8d..3b98159 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -108,13 +108,18 @@ if SND_IMX_SOC config SND_SOC_IMX_SSI tristate -config SND_SOC_IMX_PCM_FIQ +config SND_SOC_IMX_PCM tristate + +config SND_SOC_IMX_PCM_FIQ + bool select FIQ + select SND_SOC_IMX_PCM config SND_SOC_IMX_PCM_DMA - tristate + bool select SND_SOC_DMAENGINE_PCM + select SND_SOC_IMX_PCM config SND_SOC_IMX_AUDMUX tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index ec14579..afd3479 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -41,10 +41,7 @@ endif obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o -obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += snd-soc-imx-pcm-fiq.o -snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o imx-pcm.o -obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += snd-soc-imx-pcm-dma.o -snd-soc-imx-pcm-dma-y := imx-pcm-dma.o imx-pcm.o +obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c index 0c9f188..d5cd9ef 100644 --- a/sound/soc/fsl/imx-pcm.c +++ b/sound/soc/fsl/imx-pcm.c @@ -31,6 +31,7 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); return ret; } +EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap); static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { @@ -79,6 +80,7 @@ int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) out: return ret; } +EXPORT_SYMBOL_GPL(imx_pcm_new); void imx_pcm_free(struct snd_pcm *pcm) { @@ -100,6 +102,7 @@ void imx_pcm_free(struct snd_pcm *pcm) buf->area = NULL; } } +EXPORT_SYMBOL_GPL(imx_pcm_free); MODULE_DESCRIPTION("Freescale i.MX PCM driver"); MODULE_AUTHOR("Sascha Hauer <s.hauer(a)pengutronix.de>"); -- 1.7.9.5

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[alsa-devel] [PATCH - TINYCOMPRESS 1/4] compress: Extend limit on error messages
by Charles Keepax 28 Jan '13

28 Jan '13

32 characters is a very short size for the error message, extend this to 128 characters to prevent clipping on the majority of messages. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> diff --git a/compress.c b/compress.c index faef982..05e54ae 100644 --- a/compress.c +++ b/compress.c @@ -75,7 +75,7 @@ #include "sound/compress_offload.h" #include "tinycompress/tinycompress.h" -#define COMPR_ERR_MAX 32 +#define COMPR_ERR_MAX 128 struct compress { int fd; -- 1.7.2.5

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Re: [alsa-devel] [PATCH 3/3] ALSA: intel8x0: Check return value before assigning
by H. Peter Anvin 27 Jan '13

27 Jan '13

Wtf? This patch seems semibogus at best and the description is even more ridiculous... Emil Goode <emilgoode(a)gmail.com> wrote: >The first patch in this series changes the return type of function >kvm_para_available to bool. In the function snd_intel8x0_inside_vm >we now need to check it's return value before assigning a value to >the result variable. > >Signed-off-by: Emil Goode <emilgoode(a)gmail.com> >--- > sound/pci/intel8x0.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > >diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c >index 3b9be75..fa9337e 100644 >--- a/sound/pci/intel8x0.c >+++ b/sound/pci/intel8x0.c >@@ -2982,7 +2982,7 @@ static int snd_intel8x0_inside_vm(struct pci_dev >*pci) > } > > /* detect KVM and Parallels virtual environments */ >- result = kvm_para_available(); >+ result = kvm_para_available() ? 1 : 0; > #ifdef X86_FEATURE_HYPERVISOR > result = result || boot_cpu_has(X86_FEATURE_HYPERVISOR); > #endif -- Sent from my mobile phone. Please excuse brevity and lack of formatting.

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[alsa-devel] ASoC updates for v3.8
by Mark Brown 27 Jan '13

27 Jan '13

The following changes since commit 7d1f9aeff1ee4a20b1aeb377dd0f579fe9647619: Linux 3.8-rc4 (2013-01-17 19:25:45 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.8-rc4 for you to fetch changes up to a8c136d2eba7b0b0c872e59eb404b1fe36504112: Merge remote-tracking branch 'asoc/fix/wm2200' into tmp (2013-01-22 16:26:33 +0800) ---------------------------------------------------------------- ASoC: Updates for v3.8-rc4 The usual set of driver updates, nothing too thrilling in here - one core change for the regulator bypass mode which was just not doing the right thing at all and a bunch of driver specifics. ---------------------------------------------------------------- Charles Keepax (2): ASoC: arizona: Disable free-running mode on FLL1 ASoC: wm_adsp: Release firmware on error Chris Rattray (1): ASoC: wm2200: correct mixer values and text Liam Girdwood (1): ASoC: MAINTAINERS: Update email address. Mark Brown (10): ASoC: dapm: Fix sense of regulator bypass mode ASoC: wm5102: Correct AEC loopback mask ASoC: wm5110: Correct AEC loopback mask ASoC: arizona: Use actual rather than desired BCLK when calculating LRCLK ASoC: wm_adsp: Use GFP_DMA for things that may be DMAed Merge remote-tracking branch 'asoc/fix/arizona' into tmp Merge remote-tracking branch 'asoc/fix/core' into tmp Merge remote-tracking branch 'asoc/fix/fsl' into tmp Merge remote-tracking branch 'asoc/fix/lrg' into tmp Merge remote-tracking branch 'asoc/fix/wm2200' into tmp Shawn Guo (1): ASoC: fsl: fix multiple definition of init_module MAINTAINERS | 2 +- sound/soc/codecs/arizona.c | 5 ++++- sound/soc/codecs/wm2200.c | 3 --- sound/soc/codecs/wm5102.c | 3 +-- sound/soc/codecs/wm5110.c | 3 +-- sound/soc/codecs/wm_adsp.c | 6 +++--- sound/soc/fsl/Kconfig | 9 ++------- sound/soc/fsl/Makefile | 5 ++++- sound/soc/fsl/imx-pcm.c | 3 --- sound/soc/soc-dapm.c | 12 ++++++++++-- 10 files changed, 26 insertions(+), 25 deletions(-)

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[alsa-devel] [git pull] usb-audio: RME Babyface/Fireface fix
by Clemens Ladisch 27 Jan '13

27 Jan '13

The following change since commit 29594404d7fe73cd80eaa4ee8c43dcc53970c60e: Linux 3.7 (2012-12-10 19:30:57 -0800) is available in the git repository at: git://git.alsa-project.org/alsa-kprivate.git usb-audio-fix for you to fetch changes up to 5fcbad428a81efa318f15bfcfbcd8f0a98ec0b11: ALSA: usb-audio: fix invalid length check for RME and other UAC 2 devices (2013-01-26 17:49:35 +0100) ---------------------------------------------------------------- Clemens Ladisch (1): ALSA: usb-audio: fix invalid length check for RME and other UAC 2 devices sound/usb/mixer.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) ALSA: usb-audio: fix invalid length check for RME and other UAC 2 devices Commit 23caaf19b11e (ALSA: usb-mixer: Add support for Audio Class v2.0) forgot to adjust the length check for UAC 2.0 feature unit descriptors. This would make the code abort on encountering a feature unit without per-channel controls, and thus prevented the driver to work with any device having such a unit, such as the RME Babyface or Fireface UCX. Reported-by: Florian Hanisch <fhanisch(a)uni-potsdam.de> Tested-by: Matthew Robbetts <wingfeathera(a)gmail.com> Tested-by: Michael Beer <beerml(a)sigma6audio.de> Cc: Daniel Mack <daniel(a)caiaq.de> Cc: 2.6.35+ <stable(a)vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 298070e..41e8bfb 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1259,16 +1259,23 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void } channels = (hdr->bLength - 7) / csize - 1; bmaControls = hdr->bmaControls; + if (hdr->bLength < 7 + csize) { + snd_printk(KERN_ERR "usbaudio: unit %u: " + "invalid UAC_FEATURE_UNIT descriptor\n", + unitid); + return -EINVAL; + } } else { struct uac2_feature_unit_descriptor *ftr = _ftr; csize = 4; channels = (hdr->bLength - 6) / 4 - 1; bmaControls = ftr->bmaControls; - } - - if (hdr->bLength < 7 || !csize || hdr->bLength < 7 + csize) { - snd_printk(KERN_ERR "usbaudio: unit %u: invalid UAC_FEATURE_UNIT descriptor\n", unitid); - return -EINVAL; + if (hdr->bLength < 6 + csize) { + snd_printk(KERN_ERR "usbaudio: unit %u: " + "invalid UAC_FEATURE_UNIT descriptor\n", + unitid); + return -EINVAL; + } } /* parse the source unit */

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Re: [alsa-devel] [PATCH v2] ALSA: ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback
by Mark Brown 27 Jan '13

27 Jan '13

On Fri, Jan 18, 2013 at 10:17:00AM +0100, Michal Bachraty wrote: > u32 rotate = (32 - word_length) / 4; > This implementation is wrong, but it works only for 16, or 32 bit audio data. Applied, thanks.

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Re: [alsa-devel] [PATCH V6 10/10] dmaengine: Fix compilation error in non-DT case
by Mark Brown 27 Jan '13

27 Jan '13

On Fri, Jan 18, 2013 at 05:17:09PM +0530, Padmavathi Venna wrote: > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> This should be sent separately to Vinod for merge into v3.8, it's not obviously related to the rest of the series and is a bug fix.

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[alsa-devel] upsampling
by Ran Shalit 26 Jan '13

26 Jan '13

Hello, I would like to ask if I can use alsa-lib to resample (downsample or sample) input audio and playback audio. I searched for appropriate API but found none. Regards, Ran

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