August 2012 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [sound-unstable:master 94/97] sound/pci/hda/hda_intel.c:3174:6: error: 'struct azx' has no member named 'fw'
by Takashi Iwai 09 Aug '12

09 Aug '12

At Thu, 9 Aug 2012 23:34:06 +0800, Fengguang Wu wrote: > > Hi Takashi, > > FYI, kernel build failed on > > tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master > head: d2dfd82739080fefb704c0d8e4c9b2b2093a601c > commit: 5cb543dba9867588786f87af2e64fca371b69283 [94/97] ALSA: hda - Deferred probing with request_firmware_nowait() > config: i386-defconfig (attached as .config) > > All related error/warning messages: > > sound/pci/hda/hda_intel.c: In function 'azx_firmware_cb': > sound/pci/hda/hda_intel.c:3174:6: error: 'struct azx' has no member named 'fw' > sound/pci/hda/hda_intel.c: At top level: > sound/pci/hda/hda_intel.c:3163:13: warning: 'azx_firmware_cb' defined but not used [-Wunused-function] Oh, a typical mistake, a forgotten ifdef. Fixed now. Thanks! Takashi

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[alsa-devel] [PATCH 10/22] Documentation: Define the MSP Driver Device Tree bindings
by Lee Jones 09 Aug '12

09 Aug '12

Here we add the required documentation for the new Device Tree bindings pertaining to the MSP CPU-side DAI Driver. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- .../devicetree/bindings/sound/ux500-msp.txt | 43 ++++++++++++++++++++ 1 file changed, 43 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ux500-msp.txt diff --git a/Documentation/devicetree/bindings/sound/ux500-msp.txt b/Documentation/devicetree/bindings/sound/ux500-msp.txt new file mode 100644 index 0000000..99acd9c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ux500-msp.txt @@ -0,0 +1,43 @@ +* ux500 MSP (CPU-side Digital Audio Interface) + +Required properties: + - compatible :"stericsson,ux500-msp-i2s" + - reg : Physical base address and length of the device's registers. + +Optional properties: + - interrupts : The interrupt output from the device. + - interrupt-parent : The parent interrupt controller. + - <name>-supply : Phandle to the regulator <name> supply + +Example: + + sound { + compatible = "stericsson,snd-soc-mop500"; + + stericsson,platform-pcm-dma = <&pcm>; + stericsson,cpu-dai = <&msp1 &msp3>; + stericsson,audio-codec = <&codec>; + }; + + pcm: ux500-pcm { + compatible = "stericsson,ux500-pcm"; + }; + + msp1: msp@80124000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80124000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + msp3: msp@80125000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80125000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + codec: ab8500-codec { + compatible = "stericsson,ab8500-codec"; + stericsson,earpeice-cmv = <950>; /* Units in mV. */ + }; -- 1.7.9.5

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[alsa-devel] [PATCH 09/22] Documentation: Define the MOP500 Audio Machine Driver Device Tree bindings
by Lee Jones 09 Aug '12

09 Aug '12

Here we add the required documentation for the new Device Tree bindings pertaining to the MOP500 Audio Machine driver. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- .../devicetree/bindings/sound/ux500-mop500.txt | 39 ++++++++++++++++++++ 1 file changed, 39 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ux500-mop500.txt diff --git a/Documentation/devicetree/bindings/sound/ux500-mop500.txt b/Documentation/devicetree/bindings/sound/ux500-mop500.txt new file mode 100644 index 0000000..48e071c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ux500-mop500.txt @@ -0,0 +1,39 @@ +* MOP500 Audio Machine Driver + +This node is responsible for linking together all ux500 Audio Driver components. + +Required properties: + - compatible : "stericsson,snd-soc-mop500" + +Non-standard properties: + - stericsson,cpu-dai : Phandle to the CPU-side DAI + - stericsson,audio-codec : Phandle to the Audio CODEC + - stericsson,card-name : Over-ride default card name + +Example: + + sound { + compatible = "stericsson,snd-soc-mop500"; + + stericsson,cpu-dai = <&msp1 &msp3>; + stericsson,audio-codec = <&codec>; + }; + + msp1: msp@80124000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80124000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + msp3: msp@80125000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80125000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + codec: ab8500-codec { + compatible = "stericsson,ab8500-codec"; + stericsson,earpeice-cmv = <950>; /* Units in mV. */ + }; -- 1.7.9.5

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[alsa-devel] [PATCH] ucm: Allow cset commands to have values with spaces.
by Tanu Kaskinen 09 Aug '12

09 Aug '12

An example: cset "name='Input Select' Digital Mic" The old parsing code interpreted "name='Input Select' Digital" as the element id, which of course didn't work. Signed-off-by: Tanu Kaskinen <tanu.kaskinen(a)digia.com> --- src/ucm/main.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 4b37776..05a7b0a 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -170,8 +170,23 @@ static int execute_cset(snd_ctl_t *ctl, char *cset) snd_ctl_elem_value_malloc(&value); snd_ctl_elem_info_malloc(&info); - pos = strrchr(cset, ' '); - if (pos == NULL) { + /* Find the space after the element id, taking quoting with + single-quotes into account. */ + for (pos = cset; *pos != '\0'; pos += strcspn(pos, "' ")) { + if (*pos == ' ') + break; + if (*pos == '\'') { + pos++; + pos += strcspn(pos, "'"); + if (*pos == '\0') { + uc_error("invalid element id (closing single-quote not found): %s", cset); + err = -EINVAL; + goto __fail; + } + pos++; + } + } + if (*pos == '\0') { uc_error("undefined value for cset >%s<", cset); err = -EINVAL; goto __fail; -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Defer probe when loading patch firmware
by Thierry Reding 09 Aug '12

09 Aug '12

Recent changes to the firmware loading helpers cause drivers to stall when firmware is loaded during the module_init() call. The snd-hda-intel module requests firmware if the patch= parameter is used to load a patch file. This patch works around the problem by deferring the probe in such cases, which will cause the module to load successfully and the driver binding to the device outside the module_init() call. Signed-off-by: Thierry Reding <thierry.reding(a)avionic-design.de> --- sound/pci/hda/hda_intel.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8aced1..296616e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -69,6 +69,7 @@ static int probe_only[SNDRV_CARDS]; static bool single_cmd; static int enable_msi = -1; #ifdef CONFIG_SND_HDA_PATCH_LOADER +static bool deferred[SNDRV_CARDS]; static char *patch[SNDRV_CARDS]; #endif #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -3156,6 +3157,16 @@ static int __devinit azx_probe(struct pci_dev *pci, if (dev >= SNDRV_CARDS) return -ENODEV; + +#ifdef CONFIG_SND_HDA_PATCH_LOADER + if (patch[dev] && *patch[dev] && !deferred[dev]) { + snd_printk(KERN_ERR SFX "deferring probe for patch %s\n", + patch[dev]); + deferred[dev] = true; + return -EPROBE_DEFER; + } +#endif + if (!enable[dev]) { dev++; return -ENOENT; -- 1.7.11.4

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Re: [alsa-devel] [PATCH v8] media: Add stk1160 new driver
by Ezequiel Garcia 09 Aug '12

09 Aug '12

Hi Mauro, On Mon, Aug 6, 2012 at 10:38 AM, Ezequiel Garcia <elezegarcia(a)gmail.com> wrote: > This driver adds support for stk1160 usb bridge as used in some > video/audio usb capture devices. > It is a complete rewrite of staging/media/easycap driver and > it's expected as a replacement. > --- > I just sent v8, but it looks it wasn't received by patchwork either. What's going on? Regards, Ezequiel.

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[alsa-devel] [PATCH] ASoC: wm8994: Add missing dapm routes for WM8958 rev A
by Mark Brown 09 Aug '12

09 Aug '12

From: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 04ef031..6c9eeca 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4038,6 +4038,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: if (wm8994->revision < 1) { + snd_soc_dapm_add_routes(dapm, wm8994_intercon, + ARRAY_SIZE(wm8994_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: Davinci: McASP: remove unused header include
by Hebbar, Gururaja 09 Aug '12

09 Aug '12

Defines or parameters from <mach/mux.h> isn't used anywhere. Hence remove the header include. Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- This patch was tested on AM18x Board and is based on repo at http://git.kernel.org/?p=linux/kernel/git/broonie/sound.git; a=shortlog;h=refs/heads/for-3.6 :100644 100644 10a2d8c... c80c20a... M sound/soc/davinci/davinci-evm.c sound/soc/davinci/davinci-evm.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 10a2d8c..c80c20a 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -24,7 +24,6 @@ #include <mach/asp.h> #include <mach/edma.h> -#include <mach/mux.h> #include "davinci-pcm.h" #include "davinci-i2s.h" -- 1.7.1

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[alsa-devel] Question about surround default pin config
by David Henningsson 09 Aug '12

09 Aug '12

Hi, Sometimes it's better to just ask than getting confused trying to figure it out yourself. When you have 2 stereo pairs of line-out or speaker, they are meant as either 4.0 (front left, front right, rear left, rear right), and sometimes they are meant as 2.1 (front left, front right, unused, lfe). We should support both variations, I assume. What is really the way to write my pin fixups to specify either 4.0 or 2.1? Are there any de-facto standards? If I look at e g check_output_sfx (in hda_auto_parser.c), it looks like it would be impossible to have 2.1, i e, impossible to have LFE without also having rear left and rear right. But I might be missing something? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] plughw vs default / per-application volume changer
by antony k 09 Aug '12

09 Aug '12

Hello, I am new to ALSA and I cant figure out 2 things. a) if and why should I prefer the "plughw:0,0" over the "default" device name. I am developing a simple audio player which is based on the latter. I made this choice because when I was using plughw:0,0 with other audio programs(flash player,vlc) open in the background I would get the "Device is busy" error code. With default, it's fine. b) can I have an application specific volume changer? So far I have managed to tweak the master volume of the "default" device which has a global effect. Im guessing a) and b) are connected. Thanks for your time

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