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

[alsa-devel] [PATCH] ALSA: hda - show ICT/KAE control bits
by Wang Xingchao 13 Aug '12

13 Aug '12

Enable two debug options for S/PDIF Converter Control. KAE: Keep Alive Enable; ICT: IEC Coding Type. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.h | 4 ++++ sound/pci/hda/hda_proc.c | 4 ++++ 2 files changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d33..7167f37 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -386,6 +386,10 @@ enum { /* DIGITAL2 bits */ #define AC_DIG2_CC (0x7f<<0) +/* DIGITAL3 bits */ +#define AC_DIG3_ICT (0xf<<16) +#define AC_DIG3_KAE (1<<23) + /* Pin widget control - 8bit */ #define AC_PINCTL_EPT (0x3<<0) #define AC_PINCTL_EPT_NATIVE 0 diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7e46258..a222aae 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -419,9 +419,13 @@ static void print_digital_conv(struct snd_info_buffer *buffer, snd_iprintf(buffer, " Pro"); if (digi1 & AC_DIG1_LEVEL) snd_iprintf(buffer, " GenLevel"); + if (digi1 & AC_DIG3_KAE) + snd_iprintf(buffer, " KAE"); snd_iprintf(buffer, "\n"); snd_iprintf(buffer, " Digital category: 0x%x\n", (digi1 >> 8) & AC_DIG2_CC); + snd_iprintf(buffer, " IEC Coding Type: 0x%x\n", + (digi1 >> 16) & AC_DIG3_ICT); } static const char *get_pwr_state(u32 state) -- 1.7.9.5

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[alsa-devel] [PATCH 00/11] MFD/ASoC/Input: twl4030-audio submodule DT support
by Peter Ujfalusi 12 Aug '12

12 Aug '12

Hello, The following series adds DT support for the twl4030 audio submodule which provides audio codec and vibra functionality. The MFD core driver is probed via DT, it will create the needed child devices based on the provided information in the DT blob. Child drivers (vibra, ASoC codec) will parse the core's node if needed to get the needed parameters for their configuration. In the ASoC codec driver the hs_extmute callback (which was used to toggle a GPIO line) has been removed. The codec driver will receive the GPIO number (if it is needed on the platform) and also the level to disable the mute. If the series is OK (and no objections from the maintainers), it would be good if this can go via audio. Changed files are well contained within the twl4030-audio stack so I do not expect merge issues later. The series has been tested on BeagleBoard (with the McBSP DT series, and with the upcoming DT audio support for BeagleBoard). Regards, Peter --- Peter Ujfalusi (11): MFD: twl4030-audio: Clean up MODULE_* and platform_driver part MFD: twl4030-audio: Convert to use devm_kzalloc MFD: twl4030-audio: Rearange and clean-up the probe function MFD: twl4030-audio: Add DT support Input: twl4030-vibra: Support for DT booted kernel ASoC: twl4030: Move hs_extmute GPIO handling to driver ARM: OMAP/ASoC: Zoom2: Let the codec to handle the hs_extmute GPIO ASoC/MFD: twl4030: Remove set_hs_extmute callback from platform data ASoC: twl4030: Convert to use devm_kzalloc ASoC: twl4030: Add pointer to pdata within the private data ASoC: twl4030: Support for DT booted kernel .../devicetree/bindings/mfd/twl4030-audio.txt | 51 +++++++++ arch/arm/mach-omap2/board-zoom-peripherals.c | 9 +- arch/arm/mach-omap2/include/mach/board-zoom.h | 2 - drivers/input/misc/twl4030-vibra.c | 20 +++- drivers/mfd/twl4030-audio.c | 114 +++++++++++++------ include/linux/i2c/twl.h | 4 +- sound/soc/codecs/twl4030.c | 117 ++++++++++++++++---- sound/soc/omap/zoom2.c | 4 - 8 files changed, 250 insertions(+), 71 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/twl4030-audio.txt -- 1.7.8.6

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Re: [alsa-devel] [PATCHv2.1-CA0132 HDA Codec 1/2] Add DSP loader code to CA0132 codec
by Takashi Iwai 11 Aug '12

11 Aug '12

At Fri, 10 Aug 2012 23:17:51 -0700, Ian Minett wrote: > > From: Ian Minett <ian_minett(a)creativelabs.com> > > Hi Takashi, > Great, it's good to hear that you are able to assist with adding the DSP > loader modifications to hda_intel.c. > > I've attached a draft patch as you requested. Please let us know if you have > any questions concerning it. The first patch is the previous update to add DSP > features to patch_ca0132, then the second is the draft patch, to add the > new DMA loader calls. Well, for cleanness, I think we should start rather from scratch: - The first patch to add a new hda_bus_ops for DSP transfer; it won't touch patch_ca0132.c but only the core part (hda_intel.c, hda_codec.[ch]) - Add the DSP transfer to patch_ca0132.c; at this point, just do DMA transfer but no new mixers etc won't be added - Add new features to patch_ca0132.c At the early development stage, the first two items can be a single patch. We can split to two when applied to the upstream. But, mixing all changes in a single patch is pretty bad. So the point is to concentrate only on DMA transfer stuff at first. Once after confirmed it's working, continue to the new DSP features. > We're also currently ascertaining the license information to go along with the > firmware binary release. Good. This is really mandatory for upstreaming. > The project team would like to know, beyond the outstanding items mentioned > above, do you see any other potential issues with the updates to the codec that > could possibly slow the submission process? We're starting to get rather close > to our target date. > Also, once the codec has made it into the ALSA tree, how long do you estimate > it would take before it is upstreamed to the Linux kernel? If you could > provide any info on the timeframe for this process, we'd greatly appreciate it. Heh, this depends on how hard we work :) If it can be quickly done, it'll be merged even in 3.7 kernel. Takashi

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[alsa-devel] [PATCH][trivial] ASoC: isabelle: Remove unneeded include of version.h
by Jesper Juhl 10 Aug '12

10 Aug '12

There is no need to include version.h in sound/soc/codecs/isabelle.c - this patch removes the pointless include. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/soc/codecs/isabelle.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 5d8f39e..1bf5560 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -13,7 +13,6 @@ */ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -- 1.7.11.4 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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Re: [alsa-devel] [PATCH] ASoC: Samsung: Fix build error
by Mark Brown 10 Aug '12

10 Aug '12

On Wed, Aug 08, 2012 at 11:34:43AM +0530, Sachin Kamat wrote: > Fixes the following build error: > In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0, > from arch/arm/plat-samsung/include/plat/dma-ops.h:17, Applied, thanks.

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Re: [alsa-devel] [PATCH] ASoC: core: remove unused variable in soc_probe() in linux-next
by Mark Brown 10 Aug '12

10 Aug '12

On Thu, Aug 09, 2012 at 11:16:26PM -0700, Jerry Snitselaar wrote: > With commit 28d528c8 "ASoC: core: Remove pointless error on card > registration failure", the variable ret is no longer used in > soc_probe() and generates an unused variable warning during a build. Applied, thanks.

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[alsa-devel] [PATCH] ALSA : hda - bug fix on return value check when getting supported power states of a codec
by Lin, Mengdong 10 Aug '12

10 Aug '12

The return value of snd_hda_param_read() is -1 for an error, otherwise it's the supported power states of a codec. The supported power states is a 32-bit value. Bit 31 will be set to 1 if the codec supports EPSS, thus making "sup" negative. And the bit 28:5 is reserved as "0". So a negative value other than -1 shall be further checked. Please refer to High-Definition spec 7.3.4.12 "Supported Power States", thanks! Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20..629131a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3497,7 +3497,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg { int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE); - if (sup < 0) + if (sup == -1) return false; if (sup & power_state) return true; -- 1.7.9.5

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[alsa-devel] [PATCH 00/11] ARM/ASoC: OMAP McBSP device tree support
by Peter Ujfalusi 10 Aug '12

10 Aug '12

Hello, in order to be able to add DT support for the McBSP driver which is used on all OMAP platforms (OMAP1/2/3/4/5) I needed to make some cleanups to the stack: - Removing callbacks used by the driver for - clock re-parenting (code moved to sound driver) - FSR/CLKR mux configuration (from now on it need to be done at board level) This mux is board specific and depends on the wiring. - OMAP3 sidetone ICLK workaround has been reimplemented using the SYSCONFIG register's SIDLEMODE filed to prevent ICLK gating while ST is running. - Replaced the cpu_is_omap* checks with future proof check. - The FIFO was not enabled on OMAP2430 SoC It has been tested on BeagleBoard (with legacy and DT boot). The series changes McBSP related files mostly. It would be great if the whole series could go via audio tree (if the patches are OK and it is fine by Tony). Regards, Peter --- Peter Ujfalusi (11): ARM/ASoC: omap-mcbsp: Move OMAP2+ clock parenting code to ASoC driver ARM: OMAP2+: McBSP: Do not create legacy devices when booting with DT data ARM: OMAP: mcbsp: Enable FIFO use for OMAP2430 ARM: OMAP: board-am3517evm: Configure McBSP1 CLKR/FSR signal source ASoC: am3517evm: Do not configure McBSP1 CLKR/FSR signal muxing ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code ASoC: omap-mcbsp: Sidetone: Use SIDLE bits in SYSCONFIG register to select noidle mode ARM: OMAP3: Remove callback for McBSP sidetone ICLK workaround ASoC: omap-mcbsp: Remove unused defines ASoC: omap-mcbsp: Remove cpu_is_omap* checks from the code ASoC: omap-mcbsp: Add device tree bindings .../devicetree/bindings/sound/omap-mcbsp.txt | 45 ++++++ arch/arm/mach-omap2/board-am3517evm.c | 13 ++ arch/arm/mach-omap2/mcbsp.c | 152 +------------------- arch/arm/plat-omap/include/plat/mcbsp.h | 3 - sound/soc/omap/am3517evm.c | 20 +--- sound/soc/omap/mcbsp.c | 72 +++++----- sound/soc/omap/mcbsp.h | 4 +- sound/soc/omap/omap-mcbsp.c | 106 ++++++++++----- sound/soc/omap/omap-mcbsp.h | 20 --- 9 files changed, 177 insertions(+), 258 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-mcbsp.txt -- 1.7.8.6

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[alsa-devel] [PATCH 1/2] ASoC: wm5102: Add missing input PGA routes
by Mark Brown 10 Aug '12

10 Aug '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 4c2fc36..a00901c 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -643,6 +643,15 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), -- 1.7.10.4

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[alsa-devel] Question about Conversion to S24_LE from S24_3LE
by Fujikawa, Kazutoshi 10 Aug '12

10 Aug '12

Hello. I'd like to play a wav file converted from S24_3LE format to S24_LE by alsa-lib. (I use aplay command.) I recognize the following: <PCM format in wav file> ex. Valid PCM data is 0x123456. S24_LE : 0x12345600(in 4bytes, LSB 0 padding) S24_3LE : 0x123456 (in 3bytes, No padding) When the S24_LE format file is played, the ALSA driver receives data format is as follows: 0x12345600 (LSB 0 padding) However, when the S24_3LE format file which is converted to S24_LE by alsa-lib is played, the ALSA driver receives data formats is as follows: 0x00123456 (MSB 0(or 0xFF) padding) Why is the how to pad 0 different(LSB/MSB)? cf. When "snd_pcm_linear_getput" is called 'get_idx' is set 16. 'put_idx' is set 8. [alsa-lib:pcm_linear.c] 211 void snd_pcm_linear_getput(const snd_pcm_channel_area_t *dst_areas, snd_ pcm_uframes_t dst_offset, 212 const snd_pcm_channel_area_t *src_areas, snd_ pcm_uframes_t src_offset, 213 unsigned int channels, snd_pcm_uframes_t fram es, 214 unsigned int get_idx, unsigned int put_idx) 215 { 216 #define CONV24_LABELS 217 #include "plugin_ops.h" 218 #undef CONV24_LABELS 219 void *get = get32_labels[get_idx]; 220 void *put = put32_labels[put_idx]; 221 unsigned int channel; 222 u_int32_t sample = 0; 223 for (channel = 0; channel < channels; ++channel) { 224 const char *src; When 'get_idx' is 16 and 'put_idx' is 8, it is converted as follows: 0x123456(S24_3LE) -> 0x00123456(MSB 0 padding) I expect the following: 0x123456(S24_3LE) -> 0x12345600(LSB 0 padding) (If 'get_idx' is 16 and 'put_idx' is 12, it is converted like this.) I'm doing something wrong? -------------------- alsa-lib-1.0.24.1 alsa-utils-1.0.24.2 -------------------- Regards. Kaz

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