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

[alsa-devel] [PATCH] ASoC: arizona: Fix debug logging level for FLLs and AIFs
by Mark Brown 06 Feb '13

06 Feb '13

Use _dbg for debug messages. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ba3bd03..5e77a60 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -57,14 +57,14 @@ #define arizona_fll_warn(_fll, fmt, ...) \ dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_fll_dbg(_fll, fmt, ...) \ - dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) + dev_dbg(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_aif_err(_dai, fmt, ...) \ dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) #define arizona_aif_warn(_dai, fmt, ...) \ dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) #define arizona_aif_dbg(_dai, fmt, ...) \ - dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) + dev_dbg(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) static int arizona_spk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, -- 1.7.10.4

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[alsa-devel] PandaBoard ES Audio Problems
by Brent Weatherall 06 Feb '13

06 Feb '13

Hello. While developing with OpenMax and ALSA on the PandaBoard ES (OMAP 4460 armhf), I ran into a strange problem. At a high level, when configuring the built in audio device (3.5 mm jack), the device accepts ANY rate requested (via snd_pcm_set_rate_near). it accepts unsupported rates without alteration. If the rate is actually not supported, then 'snd_pcm_hw_params' fails with error '-22'. After this happens, it is not possible to reconfigure the rate without closing the device and starting over to completely re-configure from the start (calling snd_pcm_open, etc...). It is not possible to simply try another rate. Technical details below: When 'snd_pcm_set_rate_near' is called, during the min and max rate probing, the code reaches interval_inline.h:71 INTERVAL_INLINE int snd_interval_min(const snd_interval_t *i) During this call, line '74 return i->min;' executes. This function ALWAYS returns the same rate value requested in 'snd_pcm_set_rate_near' as the minimum possible rate. If I request a rate near 44100, the minimum rates comes back as 44100. If I request a rate near 8000, the minimum is returned as 8000 and so forth. To recap, the 'snd_pcm_set_rate_near' call ALWAYS succeeds without altering the rate passed to it the first time. The only way to find out if the rate is actually supported is to call 'snd_pcm_hw_params' and then see if that fails with err == -22 (invalid argument). Furthermore, once the failure occurs, another rate value cannot be tried unless the PCM device is closed and the entire configuration process is repeated. After the failure, subsequent calls to 'snd_pcm_set_rate_near' with ANY different value always get set back to the original bad value. Example, I request 44100 and it fails. So I try 8000 and it returns 44100 (and of course fails again if I try 'snd_pcm_hw_params'). I am not sure yet, but is the value at 'interval_inline.h:74 return i->min;' determined directly by the hardware? It seems like it is, but I don't know the code well enough yet to make that assertion. I also haven't yet examined the 'snd_interval_max' command, but my gut tells me it is behaving the same way, since after a bad rate is set, any higher or lower rate setting attempts ALWAYS come back as the original rate. How would i determine if it is hardware or maybe something bad with the hardware.software interface? I checked the PCM state value and it was correct the entire time. Here are short snippets of two gdb sessions where I found the issue and below that is test program output showing the state values as I try to configure two different rates: Code snippet in ALSA lib file interval_inline.h: 71 INTERVAL_INLINE int snd_interval_min(const snd_interval_t *i) 72 { 73 assert(!snd_interval_empty(i)); 74 return i->min; 75 } Run #1 with 44100 'snd_pcm_set_rate_near' call: (gdb) s 74 return i->min; (gdb) print i->min $6 = 44100 Run #2 with 8000 'snd_pcm_set_rate_near' call: (gdb) s 74 return i->min; (gdb) print i->min $8 = 8000 Here is a log of the high level test program: snd_pcm_hw_params_set_rate_near succeeded. Rate set to 44100. snd_pcm_hw_params_set_channels succeeded. cannot set buffer time (Invalid argument) cannot set buffer period time (Invalid argument) Before commit, state is: SND_PCM_STATE_OPEN cannot set parameters (Invalid argument) After failure, state is: SND_PCM_STATE_OPEN Trying to re-configure rate param. Before calling 'snd_pcm_drop', state is: SND_PCM_STATE_OPEN cannot drop (Input/output error) After calling 'snd_pcm_drop', state is: SND_PCM_STATE_OPEN Trying rate 8000. snd_pcm_hw_params_set_rate_near succeeded. Rate set to 44100. snd_pcm_hw_params_set_buffer_time_near succeeded. Buffer size actually set to 22050. cannot set buffer period time (Invalid argument) cannot set parameters (Invalid argument) cannot prepare audio interface for use (Input/output error) Rate 44100 supported: FAIL Please let me know your thoughts on this and if there is a good way to prove if this is a hardware error or a software error. I am running 'Linux version 3.4.0-1487-omap4' and using the standard repositories with apt-get to install ALSA. I used 'apt-get source' to obtain 'alsa-lib-1.0.25' source files and I am running 'libasound2-dbg' to enable GDB debugging. Regards, Brent

6 17

[alsa-devel] [RFC] compress: add support for gapless playback
by Vinod Koul 06 Feb '13

06 Feb '13

From: Jeeja KP <jeeja.kp(a)intel.com> this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Documentation/sound/alsa/compress_offload.txt | 22 +++++++++++ include/sound/compress_driver.h | 2 + include/uapi/sound/compress_offload.h | 18 ++++++++- sound/core/compress_offload.c | 51 +++++++++++++++++++++++++ 4 files changed, 92 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..071c4b6 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,28 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +The decoder needs to know the encoder delay and encoder padding. So we need to +pass this to DSP. Also DSP and userspace needs to switch from one track to +another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..42d6d7c 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -110,6 +110,8 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..65ca2f1 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #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: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -122,6 +122,18 @@ struct snd_compr_codec_caps { }; /** + * struct snd_compr_metadata: compressed stream metadata + * @encoder_delay: no of samples inserted by the encoder at the beginning + * of the track + * @encoder_padding: no of samples appended by the encoder at the end + * of the track + */ +struct snd_compr_metadata { + __u32 encoder_delay; + __u32 encoder_padding; +}; + +/** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec @@ -145,6 +157,8 @@ struct snd_compr_codec_caps { 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_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) @@ -152,10 +166,12 @@ 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_PARTIAL_DRAIN _IO('C', 0x35) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 8 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..2928971 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -514,6 +514,37 @@ out: return retval; } +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *metadata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + metadata = kmalloc(sizeof(*metadata), GFP_KERNEL); + if (!metadata) + return -ENOMEM; + if (copy_from_user(metadata, (void __user *)arg, + sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + + pr_debug("metadata encoder delay=%x encoder padding=%x\n", + metadata->encoder_delay, metadata->encoder_padding); + + retval = stream->ops->set_metadata(stream, metadata); + +out: + kfree(metadata); + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +625,20 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval) + pr_err("Partial drain returned failure\n"); + else + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + return retval; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +668,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +692,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

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[alsa-devel] ASoC updates for v3.8
by Mark Brown 06 Feb '13

06 Feb '13

The following changes since commit 949db153b6466c6f7cad5a427ecea94985927311: Linux 3.8-rc5 (2013-01-25 11:57:28 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.8-rc6 for you to fetch changes up to 5074e8eca643dd19e5397e638c7e9ca0ba9b4e24: Merge remote-tracking branch 'asoc/fix/wm2200' into tmp (2013-02-06 10:54:35 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.8-rc7 I've been a bit slack sending the pull requests for this so there's rather more driver specific updates here than is desirable but everything here is nicely localised so hopefully we're OK. ---------------------------------------------------------------- Alexey Galakhov (1): ASoC: samsung: Fix compilation error on S3C2440 Benoît Thébaudeau (1): ASoC: tlv320aic3x: Remove mono support Charles Keepax (1): ASoC: wm_adsp: Release firmware on error Chris Rattray (1): ASoC: wm2200: correct IN2L and IN3L digital mute Dan Carpenter (1): ASoC: dwc: fix support for more than two channels Lee Jones (1): ASoC: Ux500: Fix build error pertaining to missing include file Mark Brown (13): ASoC: wm_adsp: Ensure ADSP2 DMAs are quiesced when DSP is halted ASoC: wm2000: Lock state machine updates Merge remote-tracking branch 'asoc/fix/arizona' into tmp Merge remote-tracking branch 'asoc/fix/core' into tmp Merge remote-tracking branch 'asoc/fix/davinci' into tmp Merge remote-tracking branch 'asoc/fix/dwc' 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/samsung' into tmp Merge remote-tracking branch 'asoc/fix/tlv320aic3x' into tmp Merge remote-tracking branch 'asoc/fix/ux500' into tmp Merge remote-tracking branch 'asoc/fix/wm2000' into tmp Merge remote-tracking branch 'asoc/fix/wm2200' into tmp Michal Bachraty (1): ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback Ryo Tsutsui (1): ASoC: arizona: Fixed a bug in FLL fractional calculation Shawn Guo (2): Revert "ASoC: fsl: fix multiple definition of init_module" ASoC: fsl: fix snd-soc-imx-pcm module build MAINTAINERS | 2 +- sound/soc/codecs/arizona.c | 16 ++++++++++++++-- sound/soc/codecs/tlv320aic3x.c | 4 ++-- sound/soc/codecs/wm2000.c | 31 ++++++++++++++++++++++++++++--- sound/soc/codecs/wm2200.c | 7 ++----- sound/soc/codecs/wm5102.c | 3 +-- sound/soc/codecs/wm5110.c | 3 +-- sound/soc/codecs/wm_adsp.c | 20 ++++++++++++++------ sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/dwc/designware_i2s.c | 4 ++++ sound/soc/fsl/imx-pcm-dma.c | 21 +-------------------- sound/soc/fsl/imx-pcm-fiq.c | 22 +--------------------- sound/soc/fsl/imx-pcm.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/fsl/imx-pcm.h | 18 ++++++++++++++++++ sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/soc-dapm.c | 12 ++++++++++-- sound/soc/ux500/mop500.c | 2 +- 17 files changed, 132 insertions(+), 69 deletions(-)

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[alsa-devel] hda_analyzer failed
by Raymond Yau 06 Feb '13

06 Feb '13

>> >> > File "/dev/shm/hda-analyzer/hda_proc.py", line 143, in wrongfile >> > raise ValueError, "wrong proc file format (%s)" % msg >> > ValueError: wrong proc file format (integer decode 'State of AFG node >> 0x01:' ('Node ')) >> > >> > Any ideas? >> >> This patch add lines which make the parser of hda_proc.py fail >> >> http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=f4… >> > > Sigh, this is not the first time new stuff in the proc file crashes hda_analyzer. > > Jaroslav, can you please fix hda_analyzer so that it ignores new stuff in the proc file instead of crashing, so we don't have this problem over and over again? > I don`t understand why the parser have to ignore new stuff BTW, why hda_proc query the connection list of vendor defined widget ? http://git.kernel.org/?p=linux/kernel/git/tiwai/hda-emu.git;a=blob;f=codecs… Node 0x23 [Vendor Defined Widget] wcaps 0xf00100: Mono Connection: -22

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Re: [alsa-devel] 3.8-rc1 patch_cirrus 4.0 regression...
by Takashi Iwai 06 Feb '13

06 Feb '13

At Wed, 6 Feb 2013 08:47:41 +0800, Daniel J Blueman wrote: > > On 6 February 2013 01:40, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 05 Feb 2013 17:34:15 +0100, > > Takashi Iwai wrote: > >> > >> At Wed, 6 Feb 2013 00:29:54 +0800, > >> Daniel J Blueman wrote: > >> > > >> > On 6 February 2013 00:16, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > > At Wed, 6 Feb 2013 00:10:30 +0800, > >> > > Daniel J Blueman wrote: > >> > >> > >> > >> Hi Takashi, > >> > >> > >> > >> The v3.8-rc kernels have regressed from v3.7 with the quad-speaker > >> > >> arrangement on my Macbook Pro 10,1 - only the higher-frequency > >> > >> speakers work despite the front and rear channels being exposed in the > >> > >> mixer. > >> > > > >> > > Interesting. So you have a machine with 4.0 built-in speaker instead > >> > > of 2.1? Then we need to add a device-specific flag for it. Currently > >> > > the driver assumes 2.1 system blindly because majority of machines > >> > > have that. > >> > > > >> > > FWIW, the codec parser code has been totally rewritten for 3.9, so any > >> > > patch to 3.8 won't be applied to 3.9 (and vice versa)... > >> > > > >> > > Could you give alsa-info.sh output on 3.8 kernel? Then I'll try to > >> > > cook it for 3.9 (and maybe backport to 3.8). > >> > > >> > Here's the output from the current alsa-info.sh on 3.8-rc6 with the > >> > two cited patches reverted [1]; let me know if you'd like 3.8-rc6 > >> > pure. > >> > >> Thanks. > >> > >> > The sound is pretty impressive for a laptop when the low-frequecy > >> > speakers are enabled. > >> > >> Which program are you using for testing the surrounds? > >> I'm interested in it because the commit you reverted is basically > >> providing only an additional information for the channel map, and it > >> doesn't change anything else. It implies that some applications are > >> really referring to the chmap info. > > > > Or, it might be that the mixer value is simply not set correct. > > > > To be sure, could you try again 3.8-rc6 without reversing patches, > > adjust "Speaker" and "Bass Speaker" volumes properly, and retest? > > If it still doesn't work, please take alsa-info.sh snapshot at this > > state for comparing with the previous result. > > My apologies! We do now have "Bass Speaker" which affects both bass > speakers. It was always being restored to level 0 and works when set > up. OK, good to know. > There is a "Subwoofer" slider in addition to "Balance" and "Fade" in > the GNOME mixer UI which is greyed out; presumably this is intended as > the same mixer control? I have no idea about GNOME stuff... David, do you know of this? Takashi

2 1

[alsa-devel] [PATCH 00/15] ARM build regressions in v3.8
by Arnd Bergmann 06 Feb '13

06 Feb '13

I know this comes late, but we have a number of broken configurations in ARM in v3.8 that were still building in v3.7, and I'd like to get them all fixed in the final 3.8 release. It would be nice if the respective maintainers could have a look at these patches and apply them directly when they are happy with them. The first patch in the series is strictly speaking not a build error but just a warning, but it is a particularly annoying one that came in through the latest binutils release rather than a kernel change. The same binutils update also broke the samsung and w90x900 platforms. A few of the other changes are the result of the imx multiplatform conversion. I'm not really fixing those here, just picking up the pieces. It would be much nicer if we could actually get those drivers to work again with CONFIG_MULTIPLATFORM enabled rather than just disabling them, but it may be much too late for that. At least the drivers don't seem to be too essential, as they are only built in allyesconfig but not in any of the defconfigs. Arnd Arnd Bergmann (15): ARM: compressed/head.S: work around new binutils warning ARM: mvebu: build coherency_ll.S for arch=armv7-a ARM: samsung: fix assembly syntax for new gas ARM: w90x900: fix legacy assembly syntax ASoC: fsl: fiq and dma cannot both be modules clk: export __clk_get_name drm/exynos: don't include plat/gpio-cfg.h drm/exynos: fimd and ipp are broken on multiplatform media: coda: don't build on multiplatform mfd/vexpress: export vexpress_config_func_{put,get} mtd: davinci_nand: fix OF support USB: gadget/freescale: disable non-multiplatform drivers USB: ehci: make orion and mxc bus glues coexist samples/seccomp: be less stupid about cross compiling staging/omapdrm: don't build on multiplatform arch/arm/boot/compressed/Makefile | 2 +- arch/arm/boot/compressed/head.S | 12 ++++++++++++ arch/arm/mach-mvebu/coherency_ll.S | 1 + arch/arm/mach-s3c24xx/include/mach/debug-macro.S | 12 ++++++------ arch/arm/mach-s3c24xx/include/mach/entry-macro.S | 4 ++-- arch/arm/mach-s3c24xx/pm-h1940.S | 2 +- arch/arm/mach-s3c24xx/sleep-s3c2410.S | 12 ++++++------ arch/arm/mach-s3c24xx/sleep-s3c2412.S | 12 ++++++------ arch/arm/mach-w90x900/include/mach/entry-macro.S | 4 ++-- arch/arm/plat-samsung/include/plat/debug-macro.S | 18 +++++++++--------- drivers/clk/clk.c | 1 + drivers/gpu/drm/exynos/Kconfig | 4 ++-- drivers/gpu/drm/exynos/exynos_hdmi.c | 1 - drivers/media/platform/Kconfig | 2 +- drivers/mfd/vexpress-config.c | 3 ++- drivers/mtd/nand/davinci_nand.c | 2 +- drivers/staging/omapdrm/Kconfig | 2 +- drivers/usb/gadget/Kconfig | 3 ++- drivers/usb/host/ehci-hcd.c | 16 +++++++++++++++- samples/seccomp/Makefile | 2 ++ sound/soc/fsl/Kconfig | 3 +++ 21 files changed, 76 insertions(+), 42 deletions(-) -- 1.7.10.4 Cc: Artem Bityutskiy <artem.bityutskiy(a)linux.intel.com> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: David Airlie <airlied(a)linux.ie> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: James Morris <james.l.morris(a)oracle.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Mauro Carvalho Chehab <mchehab(a)redhat.com> Cc: Mike Turquette <mturquette(a)linaro.org> Cc: Rob Clark <rob(a)ti.com> Cc: Russell King <rmk+kernel(a)arm.linux.org.uk> Cc: Shawn Guo <shawn.guo(a)linaro.org> Cc: alsa-devel(a)alsa-project.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-media(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org

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Re: [alsa-devel] 3.8-rc1 patch_cirrus 4.0 regression...
by Takashi Iwai 05 Feb '13

05 Feb '13

At Wed, 6 Feb 2013 00:29:54 +0800, Daniel J Blueman wrote: > > On 6 February 2013 00:16, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Wed, 6 Feb 2013 00:10:30 +0800, > > Daniel J Blueman wrote: > >> > >> Hi Takashi, > >> > >> The v3.8-rc kernels have regressed from v3.7 with the quad-speaker > >> arrangement on my Macbook Pro 10,1 - only the higher-frequency > >> speakers work despite the front and rear channels being exposed in the > >> mixer. > > > > Interesting. So you have a machine with 4.0 built-in speaker instead > > of 2.1? Then we need to add a device-specific flag for it. Currently > > the driver assumes 2.1 system blindly because majority of machines > > have that. > > > > FWIW, the codec parser code has been totally rewritten for 3.9, so any > > patch to 3.8 won't be applied to 3.9 (and vice versa)... > > > > Could you give alsa-info.sh output on 3.8 kernel? Then I'll try to > > cook it for 3.9 (and maybe backport to 3.8). > > Here's the output from the current alsa-info.sh on 3.8-rc6 with the > two cited patches reverted [1]; let me know if you'd like 3.8-rc6 > pure. Thanks. > The sound is pretty impressive for a laptop when the low-frequecy > speakers are enabled. Which program are you using for testing the surrounds? I'm interested in it because the commit you reverted is basically providing only an additional information for the channel map, and it doesn't change anything else. It implies that some applications are really referring to the chmap info. Takashi

1 1

Re: [alsa-devel] 3.8-rc1 patch_cirrus 4.0 regression...
by Takashi Iwai 05 Feb '13

05 Feb '13

At Wed, 6 Feb 2013 00:10:30 +0800, Daniel J Blueman wrote: > > Hi Takashi, > > The v3.8-rc kernels have regressed from v3.7 with the quad-speaker > arrangement on my Macbook Pro 10,1 - only the higher-frequency > speakers work despite the front and rear channels being exposed in the > mixer. Interesting. So you have a machine with 4.0 built-in speaker instead of 2.1? Then we need to add a device-specific flag for it. Currently the driver assumes 2.1 system blindly because majority of machines have that. FWIW, the codec parser code has been totally rewritten for 3.9, so any patch to 3.8 won't be applied to 3.9 (and vice versa)... Could you give alsa-info.sh output on 3.8 kernel? Then I'll try to cook it for 3.9 (and maybe backport to 3.8). Takashi > > Reverting f37bc7 [1] restored the correct behaviour ([2] was reverted > to compile, but the problem still occurs with only [2] reverted). > > How can I help to debug this and find the right approach? > > Thanks, > Daniel > > --- [1] > > commit f37bc7a88d374448a1f4bba9267d308606d78bf2 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Thu Nov 8 15:59:23 2012 +0100 > > ALSA: hda - Give standard "Bass Speaker" mixer for 2.1 speakers > > When two built-in speakers are found on the machine, we can suppose > it's rather a 2.1 speaker system with a bass output instead of > front/surround channels. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- [2] > > commit ee81abb623cb5e03c182d16871bb4fb34fdc9b4f > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Thu Nov 8 17:12:10 2012 +0100 > > ALSA: hda - Apply a proper chmap for built-in 2.1 speakers > > When 2.1 speakers are detected, use the corresponding channel map > instead of the standard map with front+rear surrounds. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > -- > Daniel J Blueman >

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[alsa-devel] [PATCH] ASoC: wm2200: Provide platform data for MICBIAS configuration
by Chris Rattray 05 Feb '13

05 Feb '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- include/sound/wm2200.h | 22 +++++++++++++++++++++- sound/soc/codecs/wm2200.c | 31 +++++++++++++++++++++++++++++++ 2 files changed, 52 insertions(+), 1 deletions(-) diff --git a/include/sound/wm2200.h b/include/sound/wm2200.h index 79bf55b..bc7ab1a 100644 --- a/include/sound/wm2200.h +++ b/include/sound/wm2200.h @@ -12,6 +12,7 @@ #define __LINUX_SND_WM2200_H #define WM2200_GPIO_SET 0x10000 +#define WM2200_MAX_MICBIAS 2 enum wm2200_in_mode { WM2200_IN_SE = 0, @@ -25,6 +26,24 @@ enum wm2200_dmic_sup { WM2200_DMIC_SUP_MICBIAS2 = 2, }; +enum wm2200_mbias_lvl { + WM2200_MBIAS_LVL_1V5 = 1, + WM2200_MBIAS_LVL_1V8 = 2, + WM2200_MBIAS_LVL_1V9 = 3, + WM2200_MBIAS_LVL_2V0 = 4, + WM2200_MBIAS_LVL_2V2 = 5, + WM2200_MBIAS_LVL_2V4 = 6, + WM2200_MBIAS_LVL_2V5 = 7, + WM2200_MBIAS_LVL_2V6 = 8, +}; + +struct wm2200_micbias { + enum wm2200_mbias_lvl mb_lvl; /** Regulated voltage */ + unsigned int discharge:1; /** Actively discharge */ + unsigned int fast_start:1; /** Enable aggressive startup ramp rate */ + unsigned int bypass:1; /** Use bypass mode */ +}; + struct wm2200_pdata { int reset; /** GPIO controlling /RESET, if any */ int ldo_ena; /** GPIO controlling LODENA, if any */ @@ -35,7 +54,8 @@ struct wm2200_pdata { enum wm2200_in_mode in_mode[3]; enum wm2200_dmic_sup dmic_sup[3]; - int micbias_cfg[2]; /** Register value to configure MICBIAS */ + /** MICBIAS configurations */ + struct wm2200_micbias micbias[WM2200_MAX_MICBIAS]; }; #endif diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index bdd3d80..bbd8383 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2192,6 +2192,7 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, struct wm2200_priv *wm2200; unsigned int reg; int ret, i; + int val; wm2200 = devm_kzalloc(&i2c->dev, sizeof(struct wm2200_priv), GFP_KERNEL); @@ -2342,6 +2343,36 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, regmap_write(wm2200->regmap, WM2200_AUDIO_IF_1_16 + i, i); } + for (i = 0; i < WM2200_MAX_MICBIAS; i++) { + if (!wm2200->pdata.micbias[i].mb_lvl && + !wm2200->pdata.micbias[i].bypass) + continue; + + /* Apply default for bypass mode */ + if (!wm2200->pdata.micbias[i].mb_lvl) + wm2200->pdata.micbias[i].mb_lvl + = WM2200_MBIAS_LVL_1V5; + + val = (wm2200->pdata.micbias[i].mb_lvl -1) + << WM2200_MICB1_LVL_SHIFT; + + if (wm2200->pdata.micbias[i].discharge) + val |= WM2200_MICB1_DISCH; + + if (wm2200->pdata.micbias[i].fast_start) + val |= WM2200_MICB1_RATE; + + if (wm2200->pdata.micbias[i].bypass) + val |= WM2200_MICB1_MODE; + + regmap_update_bits(wm2200->regmap, + WM2200_MIC_BIAS_CTRL_1 + i, + WM2200_MICB1_LVL_MASK | + WM2200_MICB1_DISCH | + WM2200_MICB1_MODE | + WM2200_MICB1_RATE, val); + } + for (i = 0; i < ARRAY_SIZE(wm2200->pdata.in_mode); i++) { regmap_update_bits(wm2200->regmap, wm2200_mic_ctrl_reg[i], WM2200_IN1_MODE_MASK | -- 1.7.2.5

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