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[alsa-devel] [PATCH] ALSA: pcm - introduce soc_delay
by Vinod Koul 23 Jul '12

23 Jul '12

In many modern SoCs the audio DSP can buffer the PCM ring buffer data. Today we have no means to represent this buffering and ALSA wrongly detects an overrun when hw_ptr reaches app_ptr value, though DSP may still have some buffered data. This patch tries to add a new field "soc_delay" to represent buffering done in DSPs. This value is also used for the xrun calculations in ALSA. Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> -- Once we are okay with this approach, I will send a follow up patch which adds this notion in ASoC and uses this to compute cpu_dai delay. The codec_dai delay along with FIFO delay from cpu_dai should be added and reresented by today's notion of delay. --- include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 14 +++++++++++--- sound/core/pcm_native.c | 6 +++--- 3 files changed, 15 insertions(+), 6 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index a55d5db..405deb7 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -281,6 +281,7 @@ struct snd_pcm_runtime { unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ unsigned long hw_ptr_buffer_jiffies; /* buffer time in jiffies */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ + snd_pcm_sframes_t soc_delay; /* extra delay; typically delay incurred in soc */ /* -- HW params -- */ snd_pcm_access_t access; /* access mode */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 8f312fa..4977012 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -292,7 +292,15 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return -EPIPE; } } else { - if (avail >= runtime->stop_threshold) { + snd_pcm_uframes_t actual_avail; + if (avail < runtime->soc_delay) + actual_avail = avail; + else + actual_avail = avail - runtime->soc_delay; + if (actual_avail >= runtime->stop_threshold) { + snd_printd(KERN_ERR "avail > stop_threshold!!\n"); + snd_printd(KERN_ERR "actual_avail %ld, avail %ld, soc_delay %ld!!\n", + actual_avail, avail, runtime->soc_delay); xrun(substream); return -EPIPE; } @@ -440,9 +448,9 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (runtime->hw.info & SNDRV_PCM_INFO_BATCH) goto no_jiffies_check; hdelta = delta; - if (hdelta < runtime->delay) + if (hdelta < (runtime->delay + runtime->soc_delay)) goto no_jiffies_check; - hdelta -= runtime->delay; + hdelta -= (runtime->delay + runtime->soc_delay); jdelta = curr_jiffies - runtime->hw_ptr_jiffies; if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) { delta = jdelta / diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 53b5ada..fc2d664 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -606,13 +606,13 @@ int snd_pcm_status(struct snd_pcm_substream *substream, if (runtime->status->state == SNDRV_PCM_STATE_RUNNING || runtime->status->state == SNDRV_PCM_STATE_DRAINING) { status->delay = runtime->buffer_size - status->avail; - status->delay += runtime->delay; + status->delay += runtime->delay + runtime->soc_delay; } else status->delay = 0; } else { status->avail = snd_pcm_capture_avail(runtime); if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) - status->delay = status->avail + runtime->delay; + status->delay = status->avail + runtime->delay + runtime->soc_delay; else status->delay = 0; } @@ -2442,7 +2442,7 @@ static int snd_pcm_delay(struct snd_pcm_substream *substream, n = snd_pcm_playback_hw_avail(runtime); else n = snd_pcm_capture_avail(runtime); - n += runtime->delay; + n += runtime->delay + runtime->soc_delay; break; case SNDRV_PCM_STATE_XRUN: err = -EPIPE; -- 1.7.0.4

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[alsa-devel] Can ASoC support SPDIF AC3 passthrough playback (non linear PCM) ?
by Sangsu Park 23 Jul '12

23 Jul '12

Hi, We're developing a platform where we have SPDIF output. And we want to SPDIF AC3 playback (non linear PCM). Our CPU is a kind of ARM architecture one and it has a SPDIF that can support AC3 passthrough playback. I've already know that ALSA can support SPDIF AC3 passthrough playback like follows, http://www.mythtv.org/wiki/Configuring_Digital_Sound_with_AC3_and_SPDIF But, I can't find that any ARM architecture CPU can be supportable for SPDIF AC3 passthrough playback. Are there anyone who have a SPDIF AC3 passthrough playback? or anyone who know the clue about SPDIF AC3 passthrought playback supportable on ASoC? Our SPDIF already can support on linear PCM playback. But we have no experience in SPDIF AC3 passthrough playback. Just changing register setup from linear PCM playback to non-linear PCM playback is enough? Or any reference code on ASoC for SPDIF passthrough playback? Any hint would be greatly appreciated. Thanks.

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[alsa-devel] [PATCH] ASoC: wm8994: Ensure there are enough BCLKs for four channels
by Mark Brown 23 Jul '12

23 Jul '12

Otherwise if someone tries to use all four channels on AIF1 with the device in master mode we won't be able to clock out all the data. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bb62f4b..235577a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2649,7 +2649,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bclk_rate = params_rate(params) * 2; + bclk_rate = params_rate(params) * 4; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: bclk_rate *= 16; -- 1.7.10.4

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Re: [alsa-devel] Using MMAP feature
by Clemens Ladisch 23 Jul '12

23 Jul '12

(please don't drop the mailing list) (please configure your mailer to wrap lines) alberto Villa wrote: > Il 23/07/2012 12:38, Clemens Ladisch ha scritto: >> alberto Villa wrote: >>> Hello, >>> I'm going to use the MMAP feature of ALSA. >> >> Why do you need to do this? > > I need to use MMAP because we are running on an embedded system and we > want to optimize its operations. What exactly do you want to optimize? In most cases, mmap is no faster than snd_pcm_write; using mmap make sense only if the samples are generated dynamically without storing them in a userspace buffer. > It was necessary to use snd_pcm_wait() before snd_pcm_begin(), may be > this the reason? If you never wait, the buffer will be filled eventually. Regards, Clemens

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[alsa-devel] ASoC updates for 3.6
by Mark Brown 23 Jul '12

23 Jul '12

The following changes since commit 409b78cc17a4a3d07a541037575da648ced99437: ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define (2012-07-18 22:09:40 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.6 for you to fetch changes up to 15d47763b354256053c390a9b7bc9f2b4f197711: Merge branch 'for-3.5' into for-3.6 (2012-07-23 10:45:07 +0100) Diffstat looks a bit odd with the HDA stuff but as far as I can tell this is due to it showing up some diffs due to commits you've added. ---------------------------------------------------------------- ASoC: Additional updates for 3.6 A few more fixes for 3.6, some of which are relatively important - they've all been in -next for at least some time. - DAPM fixes for the recent locking changes. - Fix for _PRE and _POST widgets (which have been broken for a few releases now). - A couple of minor driver updates. ---------------------------------------------------------------- Benoît Thébaudeau (1): ASoC: dapm: Fix dapm_set_path_status() connect Liam Girdwood (2): ASoC: dapm: Fix locking during codec shutdown ASoC: dapm: Make sure all dapm contexts are updated Mark Brown (2): ASoC: dapm: Fix _PRE and _POST events for DAPM performance improvements Merge branch 'for-3.5' into for-3.6 Peter Ujfalusi (1): ASoC: omap-mcpdm: Add missing MODULE_ALIAS Robert Jarzmik (1): ASoC: mioa701: convert to snd_soc_register_card API sound/pci/hda/Kconfig | 13 ------------- sound/pci/hda/hda_auto_parser.c | 1 - sound/pci/hda/hda_auto_parser.h | 10 ++++++++++ sound/pci/hda/hda_codec.c | 4 ++++ sound/pci/hda/patch_conexant.c | 6 +++++- sound/pci/hda/patch_realtek.c | 15 +++++++++------ sound/pci/hda/patch_sigmatel.c | 7 ++++++- sound/soc/codecs/tlv320aic3x.c | 4 +--- sound/soc/codecs/tlv320aic3x.h | 1 + sound/soc/codecs/wm2200.c | 1 + sound/soc/codecs/wm8994.c | 3 --- sound/soc/omap/omap-mcpdm.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 33 ++++++++++++--------------------- sound/soc/soc-dapm.c | 28 ++++++++++++++++++++-------- sound/usb/mixer_maps.c | 8 ++++++++ sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 16 files changed, 108 insertions(+), 57 deletions(-)

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[alsa-devel] Using MMAP feature
by alberto Villa 23 Jul '12

23 Jul '12

Hello, I'm going to use the MMAP feature of ALSA. I understood I have to use snd_pcm_mmap_begin and snd_pcm_mmap_commit functions, but I can't understand the meaning of some parameters in the snd_pcm_mmap_begin function: when I call this function in order to have access to the capture stream on a call like this: err = snd_pcm_mmap_begin(capture, &c_area, &c_offset, &c_frames); it keeps to return c_frames = 0: is it right ? what does this mean? is the buffer full so that I can read samples or the available contiguous space to be read is 0? Thank you, Alberto -- Dr. Villa Alberto Sw Engineer SeleTech srl via Collodi, 8 20052 Monza (MI) tel: +39 039 5962000 fax: +39 039 9716905 email: a.villa(a)seletech.com web: www.seletech.com www.seletech.eu

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[alsa-devel] Using UCM with PulseAudio
by Arun Raghavan 23 Jul '12

23 Jul '12

Hello, I've been working on porting PulseAudio to Android on the OMAP4-based Galaxy Nexus, and have recently looking at the policy-bits. The basic porting has been greatly simplified with Wei Feng's work to renew PulseAudio UCM integration and Liam's help with fixing up some of the UCM config for the Galaxy Nexus. I am, however, facing some trouble mapping how the hardware is used to how UCM presents it (or, perhaps, the UCM-PA mapping does). Some simplified background: the devices of interest on the OMAP4 SoC are the main hifi PCM which can be routed to various outputs, the modem PCM which is not used for actual output but to enable use of the modem during calls, and the tones PCM which is intended to be used for playing ringtones, aiui. The first problem is mutual exclusivity of verbs. From what I can understand, verbs are intended to be mutually exclusive -- if you have a HiFi verb and a VoiceCall verb, only one may be used at a time. We have mapped verbs to card profiles, which offer the same guarantee. However, on Android (which is a fair example of the kind of audio policy we might want), the HiFi verb PCMs maybe used while the VoiceCall PCMs are open. This is done, for example, to play an end-of-call tone from the CPU while the modem PCMs are still held open. Is there some way to do this with UCM? The second problem is having separate PCMs for modifiers. In the OMAP4 profile, ringtone playback is exposed via a PlayTone modifier which corresponds to a separate PCM from regular HiFi playback. In the UCM-PA mapping we decided on, modifiers were implemented as device intended roles on a sink, so that when a stream with that role came in, we could enable the modifier, and disable it when such a stream ends. However, this doesn't account for switching the PCM on which playback is occurring. Should we be creating a separate sink for such modifiers (with lower priority, so they're not routed to unless there's a stream with the required role coming in)? Or should we be reopening the PCM for this? Finally, a question also related to modifiers -- is it expected that there will never be a case where a stream that requires no modifier is being played while a stream that does require a modifier also exists? If not, what kind of policy should we have for enabling the modifier or not? Cheers, Arun

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[alsa-devel] VIA VT2021 Support
by Paul Freeman 23 Jul '12

23 Jul '12

Hi, I recently acquired a Intel Z77 based motherboard, a GigaByte Z77-D3H which uses a VIA VT2021 chip.. having had a quick look/grep though the alsa source I didn't find any mention of it and Google has little say about it either with regards to linux. anyone aware of/working on support for this? - some of the outputs are working but not things like front heaphone out etc

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[alsa-devel] [PATCH] ASoC: dapm: Fix _PRE and _POST events for DAPM performance improvements
by Mark Brown 23 Jul '12

23 Jul '12

Ever since the DAPM performance improvements we've been marking all widgets as not dirty after each DAPM run. Since _PRE and _POST events aren't part of the DAPM graph this has rendered them non-functional, they will never be marked dirty again and thus will never be run again. Fix this by skipping them when marking widgets as not dirty. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 35c5f81..4d2c0f9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1597,7 +1597,15 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } list_for_each_entry(w, &card->widgets, list) { - list_del_init(&w->dirty); + switch (w->id) { + case snd_soc_dapm_pre: + case snd_soc_dapm_post: + /* These widgets always need to be powered */ + break; + default: + list_del_init(&w->dirty); + break; + } if (w->power) { d = w->dapm; -- 1.7.10.4

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[alsa-devel] [PATCH 0/2] RFC: support for audio wall clock
by Pierre-Louis Bossart 21 Jul '12

21 Jul '12

V2 of patches to show how the audio wall clock (eg. the HDAudio one) can be used to provide precise audio timestamps and track the drift between audio time and system time. On my laptop I am able to track a 7ppm delta. I will explain in more details how this can be used during LPC in San Diego, but with the summer coming I'd like to get feedback as there are changes to the core/alsa-lib that haven't been modified in years. TODO: - 64-bit alignment, verification of pcm_compat.c changes - handling of digital inputs, need to disable wall-clock timestamps in that case (feedback from Clemens) - check differences in protocol - change timecounter to avoid accumulation of rounding errors in cycle->ns conversions - check overflows - rework the timecounter init on trigger Pierre-Louis Bossart (2): ALSA: core: add hooks for audio timestamps read from WALLCLOCK ALSA: hda: support for wallclock timestamps include/sound/asound.h | 7 ++++- include/sound/pcm.h | 2 + sound/core/pcm_compat.c | 13 +++++++++- sound/core/pcm_lib.c | 15 ++++++++++- sound/core/pcm_native.c | 2 + sound/pci/hda/hda_intel.c | 54 ++++++++++++++++++++++++++++++++++++++++++++- 6 files changed, 86 insertions(+), 7 deletions(-) -- 1.7.6.5

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[alsa-devel] [PATCH][Trivial][resend] Documentation: Add newline at end-of-file to files lacking one
by Jesper Juhl 20 Jul '12

20 Jul '12

This patch simply adds a newline character at end-of-file to those files in Documentation/ that currently lack one. This is done for a few different reasons: A) It's rather annoying when you do "cat some_file.txt" that your prompt/cursor ends up at the end of the last line of output rather than on a new line. B) Some tools that process files line-by-line may get confused by the lack of a newline on the last line. C) The "\ No newline at end of file" line in diffs annoys me for some reason. So, let's just add the missing newline once and for all. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- Documentation/ABI/stable/vdso | 2 +- Documentation/ABI/testing/sysfs-block-zram | 2 +- Documentation/ABI/testing/sysfs-bus-usb-devices-usbsevseg | 2 +- Documentation/ABI/testing/sysfs-class-backlight-driver-adp8870 | 2 +- Documentation/arm/Samsung-S3C24XX/H1940.txt | 2 +- Documentation/arm/Samsung-S3C24XX/SMDK2440.txt | 2 +- Documentation/sound/alsa/hdspm.txt | 2 +- Documentation/video4linux/cpia2_overview.txt | 2 +- Documentation/video4linux/stv680.txt | 2 +- 9 files changed, 9 insertions(+), 9 deletions(-) diff --git a/Documentation/ABI/stable/vdso b/Documentation/ABI/stable/vdso index 8a1cbb5..7cdfc28 100644 --- a/Documentation/ABI/stable/vdso +++ b/Documentation/ABI/stable/vdso @@ -24,4 +24,4 @@ though. (As of this writing, this ABI documentation as been confirmed for x86_64. The maintainers of the other vDSO-using architectures should confirm - that it is correct for their architecture.) \ No newline at end of file + that it is correct for their architecture.) diff --git a/Documentation/ABI/testing/sysfs-block-zram b/Documentation/ABI/testing/sysfs-block-zram index c8b3b48..ec93fe3 100644 --- a/Documentation/ABI/testing/sysfs-block-zram +++ b/Documentation/ABI/testing/sysfs-block-zram @@ -96,4 +96,4 @@ Description: overhead, allocated for this disk. So, allocator space efficiency can be calculated using compr_data_size and this statistic. - Unit: bytes \ No newline at end of file + Unit: bytes diff --git a/Documentation/ABI/testing/sysfs-bus-usb-devices-usbsevseg b/Documentation/ABI/testing/sysfs-bus-usb-devices-usbsevseg index cb830df..70d00df 100644 --- a/Documentation/ABI/testing/sysfs-bus-usb-devices-usbsevseg +++ b/Documentation/ABI/testing/sysfs-bus-usb-devices-usbsevseg @@ -40,4 +40,4 @@ Description: Controls the decimal places on the device. the value of 10 ** n. Assume this field has the value k and has 1 or more decimal places set, to set the mth place (where m is not already set), - change this fields value to k + 10 ** m. \ No newline at end of file + change this fields value to k + 10 ** m. diff --git a/Documentation/ABI/testing/sysfs-class-backlight-driver-adp8870 b/Documentation/ABI/testing/sysfs-class-backlight-driver-adp8870 index 4a9c545..33e6488 100644 --- a/Documentation/ABI/testing/sysfs-class-backlight-driver-adp8870 +++ b/Documentation/ABI/testing/sysfs-class-backlight-driver-adp8870 @@ -53,4 +53,4 @@ Description: Documentation/ABI/stable/sysfs-class-backlight. It can be enabled by writing the value stored in /sys/class/backlight/<backlight>/max_brightness to - /sys/class/backlight/<backlight>/brightness. \ No newline at end of file + /sys/class/backlight/<backlight>/brightness. diff --git a/Documentation/arm/Samsung-S3C24XX/H1940.txt b/Documentation/arm/Samsung-S3C24XX/H1940.txt index f4a7b22..b738859 100644 --- a/Documentation/arm/Samsung-S3C24XX/H1940.txt +++ b/Documentation/arm/Samsung-S3C24XX/H1940.txt @@ -37,4 +37,4 @@ Maintainers Thanks to the many others who have also provided support. -(c) 2005 Ben Dooks \ No newline at end of file +(c) 2005 Ben Dooks diff --git a/Documentation/arm/Samsung-S3C24XX/SMDK2440.txt b/Documentation/arm/Samsung-S3C24XX/SMDK2440.txt index 32e1eae..429390b 100644 --- a/Documentation/arm/Samsung-S3C24XX/SMDK2440.txt +++ b/Documentation/arm/Samsung-S3C24XX/SMDK2440.txt @@ -53,4 +53,4 @@ Maintainers and to Simtec Electronics for allowing me time to work on this. -(c) 2004 Ben Dooks \ No newline at end of file +(c) 2004 Ben Dooks diff --git a/Documentation/sound/alsa/hdspm.txt b/Documentation/sound/alsa/hdspm.txt index 7a67ff7..7ba3194 100644 --- a/Documentation/sound/alsa/hdspm.txt +++ b/Documentation/sound/alsa/hdspm.txt @@ -359,4 +359,4 @@ Calling Parameter: enable_monitor int array (min = 1, max = 8), "Enable Analog Out on Channel 63/64 by default." - note: here the analog output is enabled (but not routed). \ No newline at end of file + note: here the analog output is enabled (but not routed). diff --git a/Documentation/video4linux/cpia2_overview.txt b/Documentation/video4linux/cpia2_overview.txt index a6e5366..ad6adbe 100644 --- a/Documentation/video4linux/cpia2_overview.txt +++ b/Documentation/video4linux/cpia2_overview.txt @@ -35,4 +35,4 @@ the camera. There are three modes for this. Block mode requests a number of contiguous registers. Random mode reads or writes random registers with a tuple structure containing address/value pairs. The repeat mode is only used by VP4 to load a firmware patch. It contains a starting address and -a sequence of bytes to be written into a gpio port. \ No newline at end of file +a sequence of bytes to be written into a gpio port. diff --git a/Documentation/video4linux/stv680.txt b/Documentation/video4linux/stv680.txt index 4f8946f..e3de336 100644 --- a/Documentation/video4linux/stv680.txt +++ b/Documentation/video4linux/stv680.txt @@ -50,4 +50,4 @@ The latest info on this driver can be found at: http://personal.clt.bellsouth.net/~kjsisson or at http://stv0680-usb.sourceforge.net -Any questions to me can be send to: kjsisson(a)bellsouth.net \ No newline at end of file +Any questions to me can be send to: kjsisson(a)bellsouth.net -- 1.7.11.2 -- 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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[alsa-devel] [PATCH 1/3] hda-emu: Add logging prefix option
by David Henningsson 20 Jul '12

20 Jul '12

When working with automated runs of hda-emu, it will be easier to catch warnings and errors if they are not color coded. With the -F option, each line is instead prefixed with "Warning: ", "Info: " etc, for easy parsing by automated tools. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-emu.c | 6 +++++- hda-log.c | 32 ++++++++++++++++++++++++++++++++ include/hda-log.h | 1 + 3 files changed, 38 insertions(+), 1 deletion(-) (There is no [PATCH 0/3] mail, because the patches are mostly unrelated.) diff --git a/hda-emu.c b/hda-emu.c index 87a198e..8bf1c81 100644 --- a/hda-emu.c +++ b/hda-emu.c @@ -934,6 +934,7 @@ static void usage(void) fprintf(stderr, " -q don't echo but only to log file\n"); fprintf(stderr, " -C print messages in color (default)\n"); fprintf(stderr, " -M no color print\n"); + fprintf(stderr, " -F print prefixes to messages\n"); fprintf(stderr, " -a issues SIGTRAP at codec errors\n"); fprintf(stderr, " -P pincfg initialize pin-configuration from sysfs entry\n"); fprintf(stderr, " -j NID turn on the initial jack-state of the given pin\n"); @@ -982,7 +983,7 @@ int main(int argc, char **argv) int num_active_jacks = 0; unsigned int active_jacks[16]; - while ((c = getopt(argc, argv, "al:i:p:m:do:qCMP:j:")) != -1) { + while ((c = getopt(argc, argv, "al:i:p:m:do:qCMFP:j:")) != -1) { switch (c) { case 'a': hda_log_trap_on_error = 1; @@ -1012,6 +1013,9 @@ int main(int argc, char **argv) case 'C': log_flags |= HDA_LOG_FLAG_COLOR; break; + case 'F': + log_flags |= HDA_LOG_FLAG_PREFIX; + break; case 'M': log_flags &= ~HDA_LOG_FLAG_COLOR; break; diff --git a/hda-log.c b/hda-log.c index 1ad9bda..65f3044 100644 --- a/hda-log.c +++ b/hda-log.c @@ -35,6 +35,7 @@ static FILE *logfp; static int log_flags = HDA_LOG_FLAG_COLOR; int hda_log_trap_on_error; +static int log_previous_prefix = -1; static void set_color(int level) { @@ -61,15 +62,43 @@ static void reset_color(void) printf("\x1b[0m"); } +static void print_prefix(int level) +{ + static char *prefix_seq[] = { + [HDA_LOG_ERR] = "Error", + [HDA_LOG_WARN] = "Warning", + [HDA_LOG_KERN] = "Kernel", + [HDA_LOG_INFO] = "Info", + [HDA_LOG_VERB] = "Verb", + }; + + if (!(log_flags & HDA_LOG_FLAG_PREFIX)) + return; + if (level < 0) + level = 0; + else if (level > HDA_LOG_VERB) + level = HDA_LOG_VERB; + + if (log_previous_prefix == level) + return; + if (log_previous_prefix != -1) + fprintf(logfp, "\n"); + fprintf(logfp, "%s: ", prefix_seq[level]); + log_previous_prefix = level; +} + static void _hda_log(int level, const char *fmt, va_list ap) { va_list ap2; + int i; if (level > log_level) return; if (logfp == stdout) set_color(level); + print_prefix(level); + va_copy(ap2, ap); vfprintf(logfp, fmt, ap); if (!(log_flags & HDA_LOG_FLAG_NO_ECHO) && logfp != stdout) @@ -77,6 +106,9 @@ static void _hda_log(int level, const char *fmt, va_list ap) va_end(ap2); if (logfp == stdout) reset_color(); + if (((i = strlen(fmt)) > 0) && (fmt[i-1] == '\n')) + log_previous_prefix = -1; + if ((level == HDA_LOG_ERR || level == HDA_LOG_WARN) && hda_log_trap_on_error) raise(SIGTRAP); diff --git a/include/hda-log.h b/include/hda-log.h index 351a427..2e2137b 100644 --- a/include/hda-log.h +++ b/include/hda-log.h @@ -12,6 +12,7 @@ enum { /* flags */ #define HDA_LOG_FLAG_NO_ECHO (1 << 0) #define HDA_LOG_FLAG_COLOR (1 << 1) +#define HDA_LOG_FLAG_PREFIX (1 << 2) int hda_log_init(const char *file, unsigned int flags); void hda_log(int level, const char *fmt, ...); -- 1.7.9.5

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[alsa-devel] [PULL REQUEST] hda-emu: Alsa info for Canonical certified systems
by David Henningsson 20 Jul '12

20 Jul '12

The following changes since commit 6e0c8f66e0a657c30d9d965d887af054cfae3eaf: Add a dummy linux/kernel.h (2012-07-16 11:34:08 +0200) are available in the git repository at: git://kernel.ubuntu.com/diwic/hda-emu.git for-tiwai for you to fetch changes up to fb3b3a7654452bdba188166ef9815baa15042e6f: hda-emu: Add alsa info for Canonical certified systems (2012-07-20 11:47:46 +0200) ---------------------------------------------------------------- David Henningsson (1): hda-emu: Add alsa info for Canonical certified systems .../92hd71b7x-dell-inspiron-1545-ccert-200910-4252 | 956 +++++++++++ .../92hd71b7x-dell-inspiron-1545-ccert-200912-4896 | 961 +++++++++++ .../92hd71b7x-dell-inspiron-1545-ccert-200912-4897 | 968 +++++++++++ ...92hd73c1x5-dell-studio-xps-13-ccert-200910-4256 | 1107 +++++++++++++ ...hd73c1x5-dell-studio-xps-1340-ccert-200912-4895 | 1100 +++++++++++++ .../92hd75b3x5-hp-probook-6550b-ccert-201101-6954 | 1147 +++++++++++++ ...2hd81b1c5-dell-latitude-e6410-ccert-201001-5167 | 1387 ++++++++++++++++ ...hd81b1c5-dell-precision-m6500-ccert-201004-5601 | 1062 ++++++++++++ .../92hd81b1c5-dell-vostro-1720-ccert-200910-4255 | 789 +++++++++ .../92hd81b1x5-dell-inspiron-15r-ccert-201010-6645 | 949 +++++++++++ ...2hd81b1x5-dell-latitude-e4310-ccert-201103-7379 | 1071 ++++++++++++ .../92hd81b1x5-dell-vostro-3300-ccert-201001-4955 | 785 +++++++++ .../92hd81b1x5-dell-vostro-3300-ccert-201004-5576 | 781 +++++++++ .../92hd81b1x5-dell-vostro-3400-ccert-201004-5571 | 944 +++++++++++ .../92hd81b1x5-dell-vostro-3500-ccert-201001-4959 | 962 +++++++++++ ...1-3-dell-inspiron-15r--n5110--ccert-201012-6931 | 923 +++++++++++ ...2hd87b1-3-dell-inspiron-m5110-ccert-201105-8042 | 904 +++++++++++ ...2hd87b1-3-dell-inspiron-m5110-ccert-201105-8043 | 892 ++++++++++ .../92hd87b1-3-dell-vostro-1440-ccert-201105-8046 | 916 +++++++++++ .../92hd87b1-3-dell-vostro-3350-ccert-201101-6951 | 921 +++++++++++ .../92hd87b1-3-dell-vostro-3550-ccert-201012-6933 | 969 +++++++++++ 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[alsa-devel] gitcompile fails
by Brand, Mark (Mr) (Summerstrand Campus South) 20 Jul '12

20 Jul '12

Fresh (yesterday?) git clones of alsa-driver and alsa-kmirror produce a ./gitcompile failure for me: copying file alsa-kernel/pci/hda/hda_beep.c patching file hda_beep.c Hunk #2 succeeded at 96 (offset 1 line). Hunk #3 succeeded at 124 (offset 1 line). Hunk #4 succeeded at 142 (offset 1 line). Hunk #5 succeeded at 164 (offset 1 line). Hunk #6 FAILED at 282. 1 out of 6 hunks FAILED -- saving rejects to file hda_beep.c.rej make[2]: *** [hda_beep.c] Error 1 make[2]: Leaving directory `/home/mark/workspace/alsa-driver/pci/hda' make[1]: *** [prepare] Error 1 make[1]: Leaving directory `/home/mark/workspace/alsa-driver/pci' make: *** [dep] Error 1 My best guess is that there is a version mismatch between the alsa-kernel/pci/hda/hda_beep.c (linked to alsa-kmirror) and pci/hda/hda_beep.patch files, since the patch seems to refer to a non-existent line-number in my source. Any suggestions will be much appreciated, thank you. Mark Brand. ________________________________ NOTICE: Please note that this eMail, and the contents thereof, is subject to the standard NMMU eMail disclaimer which may be found at: http://www.nmmu.ac.za/disclaimer/email.htm

3 2

[alsa-devel] Compatibility among various audio parameters
by Harman Kalra 20 Jul '12

20 Jul '12

I have few doubts regarding compatibility between various audio parameters: 1. How can I determine sampling rate, audio bit depth, no of channels of a already recorded audio sample. 2. Suppose I play this same recorded audio sample with my application whose hardware parameters are different from the sample which I received. So in this case what would be the impact on the sound quality. 3. What would be the impact of using different (at capture application and playback application) sampling rate, audio bit depth, no of channels. 4. If possible kindly provide me a link which is related to my queries.

1 0

[alsa-devel] Selecting Data Access Type
by Harman Kalra 20 Jul '12

20 Jul '12

I have few doubts regarding Data Access types, which controls the way that the program will deliver/receive data from the interface in Alsa: 1. What are the advantages/disadvantages of interleaved over non interleaved or vice versa 2. What are the advantages/disadvantages of MMAP interleaved over RW interleaved or vice versa 3. Which access type is most appropriate for, camera recording Regards Harman kalra

1 0

[alsa-devel] [PATCH] ALSA: hda - add dock support for Thinkpad X230 Tablet
by David Henningsson 20 Jul '12

20 Jul '12

Also add a model/fixup string "lenovo-dock", so that other Thinkpad users will be able to test this fixup easily, to see if it enables dock I/O for them as well. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/1026953 Tested-by: John McCarron <john.mccarron(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 3 ++- sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++++++++++ 2 files changed, 29 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 2ea8cf2..7456360 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -21,13 +21,14 @@ ALC267/268 ========== inv-dmic Inverted internal mic workaround -ALC269 +ALC269/270/275/276/280/282 ====== laptop-amic Laptops with analog-mic input laptop-dmic Laptops with digital-mic input alc269-dmic Enable ALC269(VA) digital mic workaround alc271-dmic Enable ALC271X digital mic workaround inv-dmic Inverted internal mic workaround + lenovo-dock Enables docking station I/O for some Lenovos ALC662/663/272 ============== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 537c365..f141395 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5881,6 +5881,15 @@ static int alc269_resume(struct hda_codec *codec) } #endif /* CONFIG_PM */ +static void alc269_fixup_pincfg_no_hp_to_lineout(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == ALC_FIXUP_ACT_PRE_PROBE) + spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; +} + static void alc269_fixup_hweq(struct hda_codec *codec, const struct alc_fixup *fix, int action) { @@ -6007,6 +6016,8 @@ enum { ALC269VB_FIXUP_DMIC, ALC269_FIXUP_MIC2_MUTE_LED, ALC269_FIXUP_INV_DMIC, + ALC269_FIXUP_LENOVO_DOCK, + ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, }; static const struct alc_fixup alc269_fixups[] = { @@ -6135,6 +6146,20 @@ static const struct alc_fixup alc269_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC269_FIXUP_LENOVO_DOCK] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x19, 0x23a11040 }, /* dock mic */ + { 0x1b, 0x2121103f }, /* dock headphone */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT + }, + [ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_pincfg_no_hp_to_lineout, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6161,6 +6186,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), @@ -6222,6 +6248,7 @@ static const struct alc_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_STEREO_DMIC, .name = "alc269-dmic"}, {.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"}, {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {} }; -- 1.7.9.5

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[alsa-devel] [PATCH v3] ALSA: hda - Turn on PIN_OUT from hdmi playback prepare.
by Dylan Reid 20 Jul '12

20 Jul '12

Turn on the pin widget's PIN_OUT bit from playback prepare. The pin is enabled in open, but is disabled in hdmi_init_pin which is called during system resume. This causes a system suspend/resume during playback to mute HDMI/DP. Enabling the pin in prepare instead of open allows calling snd_pcm_prepare after a system resume to restore audio. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_hdmi.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0b4a1ea..641408d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -876,7 +876,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; - int pinctl; /* Validate hinfo */ pin_idx = hinfo_to_pin_index(spec, hinfo); @@ -912,11 +911,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hda_codec_write(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, mux_idx); - pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_codec_write(codec, per_pin->pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl | PIN_OUT); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); /* Initially set the converter's capabilities */ @@ -1153,11 +1147,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid; + int pinctl; hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); hdmi_setup_audio_infoframe(codec, pin_idx, substream); + pinctl = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT); + return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } -- 1.7.11.2.138.g2b53359

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Re: [alsa-devel] [regression] ALSA: hda - Handle open while transitioning to D3.
by Dylan Reid 20 Jul '12

20 Jul '12

On Thu, Jul 19, 2012 at 12:29 AM, Oleksij Rempel (fishor) <bug-track(a)fisher-privat.net> wrote: > On 19.07.2012 08:44, Dylan Reid wrote: >> >> On Wed, Jul 18, 2012 at 10:26 PM, Oleksij Rempel (fishor) >> <bug-track(a)fisher-privat.net> wrote: >>> >>> On 17.07.2012 07:26, Dylan Reid wrote: >>>> >>>> >>>> On Sat, Jul 14, 2012 at 6:36 AM, Oleksij Rempel (fishor) >>>> <bug-track(a)fisher-privat.net> wrote: >>>>> >>>>> >>>>> dmesg is in attachment >>>> >>>> >>>> Thanks Oleksij, >>>> >>>> I didn't get much from that though. What is your powersave timeout >>>> set to? >>> >>> >>> >>> I do not know, it should be some default value. >>> >>> >>>> If you have time could you try adding the following and send >>>> me the output, I have two guesses about where it is spending it's time >>>> (no guesses about why yet though). I've got a couple of systems with >>>> a 269, I'll try to get some tests run on those tomorrow. >>> >>> >>> >>> result is attached. >>> >> Great, thanks Oleksij, >> >> I tried this on an ACER AC700 which also has the ALC269VB and could >> not get anything like this to happen. So I'm sorry to keep bothering >> you for more information. I attached the alsa-info.sh output, if you >> see any interesting differences let me know. >> >> From the output it looks like something is opening the driver about >> three times a second, do you know what is doing that? It appears to >> be doing what it should be in this case, although it is a little hard >> to tell. If you could strace the process doing all the opens that >> might provide some good info. >> >> Thanks again for the help. >> Dylan > > > > Ok, i did some more testing compared the difference between battery and > non-battery mode. > > first of all on non-battery mode, this part of code is not executed. I have > no messages about "power_up cancel sync". > Then i found some difference in syslog. > > Here is what pulse audio say if i reproduce this issue: > [pulseaudio] module-alsa-card.c: Failed to open hctl/mixer for jack > detection Interesting, I'll try to get pulseaudio installed on my system to test tomorrow. Does it give any more info if you turn up the verbosity? (set log-level to debug in /etc/pulse/daemon.conf)

1 1

Re: [alsa-devel] Selecting Audio formats
by Clemens Ladisch 19 Jul '12

19 Jul '12

Harman Kalra wrote: > Can you please explain difference between signed and unsigned PCM > formats, Signed PCM formats have a sign, unsigned don't. S16 allows values in the range -32768..32767, U16, 0..65535. > in what cases does unsigned is used, In practice, never. It was historically used by some 8-bit cards. > which is more efficient (memory and sound clarity) I refer the honourable gentleman to the reply I gave in the last mail. >> If you tried to set MPEG format, ALSA would inform you that the device >> does not support this. > > But when I queried the hardware regarding the supported audio > formats using the api " snd_pcm_format_name", it responded with a list of > formats, including , A-law, u-law-, IMA_ADPCM, MPEG, GSM, so what does it > mean. It means that the sample format identified by the constant SND_PCM_FORMAT_MPEG is indeed named "MPEG". This has nothing to do with what formats some particular hardware supports. > What would be audio sample file size in case of A-law, will it be > same as U8. I refer the honourable gentleman to the reply I gave in the last mail. Regards, Clemens

1 0

[alsa-devel] Selecting Audio formats
by Harman Kalra 19 Jul '12

19 Jul '12

Hello Everybody I am a newbie to Linux Sound Programming, so few of my queries may sound foolish, but kindly address them. *I want to capture a video on my beagle board Xm rev C (TI DM3730). *So as per as the audio capturing is concerned, I am very much confused about the audio formats, like which format should I choose to get a better sound quality in the least possible memory requirement. Since so many PCM formats such as S8, U8, S16_LE, U16_LE, A-law, u-law-, IMA_ADPCM, MPEG, GSM, etc are there so which format should I opt for. Also 1. What is difference between signed and unsigned PCM format, how does it can affect the sound quality. 2. If I go for MPEG format, does it mean that audio compression would handled at my audio port level and my DSP processor would not come into picture. 3. In reference to my above query i.e. 2, if audio compression is handled at audio port level, so at the time of audio/video sync up, how would be my processor be able to time sync both audio and video, since audio is not being compressed by the DSP. 4. If I opt for A-law or u-law, what could be benefits/disadvantages of using these formats. Would the audio sample be in compressed form in this case, if yes, how would audio/video sync up will happen in this case. If possible kindly provide me a link where I can find differences between these various formats, pros and cons of each of them. Thanks & Regards Harman

2 1

[alsa-devel] SGTL5000: fix max Mic Volume in SGTL5000_CHIP_MIC_CTRL
by Simon Brenner 19 Jul '12

19 Jul '12

Hi all, I have noticed a wrong value for the "max" field of the macro SOC_SINGLE_TLV() in the Linux kernel file <kernel_root>/sound/soc/codecs/sgtl5000.c (approx. line 402, sgtl5000_snd_controls[]): SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, 0, 4, 0, mic_gain_tlv) The value "4" should be a "3" (see datasheet "Low Power Stereo Codec with Headphone Amp" by Freescale, page 40). Regards, -Simon.

1 0

[alsa-devel] [PATCH v2] ALSA: hda - Turn on PIN_OUT from hdmi playback prepare.
by Dylan Reid 19 Jul '12

19 Jul '12

Turn on the pin widget's PIN_OUT bit from playback prepare. The pin is enabled in open, but is disabled in hdmi_init_pin which is called during system resume. This causes a system suspend/resume during playback to mute HDMI/DP. Enabling the pin in prepare instead of open allows calling snd_pcm_prepare after a system resume to restore audio. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_hdmi.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0b4a1ea..764b975 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -914,9 +914,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_codec_write(codec, per_pin->pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl | PIN_OUT); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); /* Initially set the converter's capabilities */ @@ -1153,11 +1150,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid; + int pinctl; hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); hdmi_setup_audio_infoframe(codec, pin_idx, substream); + pinctl = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT); + return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } -- 1.7.11.2.138.g2b53359

2 2

[alsa-devel] [PATCH] ALSA: hda - Turn on PIN_OUT from hdmi playback prepare.
by Dylan Reid 19 Jul '12

19 Jul '12

Turn on the pin widget's PIN_OUT bit from playback prepare. The pin is enabled in open, but is disabled in hdmi_init_pin which is called during system resume. This causes a system suspend/resume during playback to mute HDMI/DP. Enabling the pin in prepare allows calling snd_pcm_prepare after a system resume to restore audio. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_hdmi.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0b4a1ea..c53f99c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1153,11 +1153,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid; + int pinctl; hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); hdmi_setup_audio_infoframe(codec, pin_idx, substream); + pinctl = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT); + return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } -- 1.7.11.2.138.g2b53359

3 3

Re: [alsa-devel] [regression] ALSA: hda - Handle open while transitioning to D3.
by Dylan Reid 19 Jul '12

19 Jul '12

On Wed, Jul 18, 2012 at 10:26 PM, Oleksij Rempel (fishor) <bug-track(a)fisher-privat.net> wrote: > On 17.07.2012 07:26, Dylan Reid wrote: >> >> On Sat, Jul 14, 2012 at 6:36 AM, Oleksij Rempel (fishor) >> <bug-track(a)fisher-privat.net> wrote: >>> >>> dmesg is in attachment >> >> Thanks Oleksij, >> >> I didn't get much from that though. What is your powersave timeout >> set to? > > > I do not know, it should be some default value. > > >> If you have time could you try adding the following and send >> me the output, I have two guesses about where it is spending it's time >> (no guesses about why yet though). I've got a couple of systems with >> a 269, I'll try to get some tests run on those tomorrow. > > > result is attached. > Great, thanks Oleksij, I tried this on an ACER AC700 which also has the ALC269VB and could not get anything like this to happen. So I'm sorry to keep bothering you for more information. I attached the alsa-info.sh output, if you see any interesting differences let me know. >From the output it looks like something is opening the driver about three times a second, do you know what is doing that? It appears to be doing what it should be in this case, although it is a little hard to tell. If you could strace the process doing all the opens that might provide some good info. Thanks again for the help. Dylan

1 0

[alsa-devel] ASoC updates for 3.6
by Mark Brown 19 Jul '12

19 Jul '12

The following changes since commit 6b16351acbd415e66ba16bf7d473ece1574cf0bc: Linux 3.5-rc4 (2012-06-24 12:53:04 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.6 for you to fetch changes up to 409b78cc17a4a3d07a541037575da648ced99437: ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define (2012-07-18 22:09:40 +0100) ---------------------------------------------------------------- ASoC: Updates for 3.6 This has been a pretty quiet release - very little activity in framework terms, mostly just a few new drivers and updates: - Added the ability to add and remove DAPM paths dynamically, mostly for reparenting on clock changes. - New machine drivers for Marvell Brownstone, ST-Ericsson Ux500 reference platform and ttc-dkp. - New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP, Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF - New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI Isabelle and Wolfson Microelectronics WM5102 and WM5110 ---------------------------------------------------------------- Adam Thomson (2): ASoC: core: Add single controls with specified range of values ASoC: codecs: Add DA732x codec driver Axel Lin (1): ASoC: ml26124: Convert to devm_regmap_init_i2c Benoît Thébaudeau (2): ASoC: dapm: Fix snd_soc_dapm_put_volsw() connect ASoC: dapm: Remove incomplete stereo code Brian Austin (2): ASoC: cs42l73: Convert to devm_regmap_init_i2c() ASoC: cs42l52: Convert to devm_regmap_init_i2c() Dan Carpenter (3): ASoC: isabelle: using an uninitialized variable ASoC: Ux500: unlock on an error path ASoC: STA529: fix an error message Fabio Estevam (1): ASoC: imx-mc13783: Add audmux settings for mx27pdk Hui Wang (1): ASoC: fsl: remove unneeded AUDMUX register setting from imx-sgtl5000 Jarkko Nikula (1): ASoC: tlv320aic3x: Change Class-D amplifier gain control name Jiri Prchal (5): ASoC: tlv320aic3x: add missing registers and bits ASoC: tlv320aic3x: extending registers cache ASoC: tlv320aic3x: add deemphasis switch ASoC: tlv320aic3x: add AGC settings ASoC: tlv320aic3x: add input clock selection Kuninori Morimoto (2): ASoC: fsi: use dmaengine helper functions ASoC: fsi: use PIO handler if DMA handler was invalid Lars-Peter Clausen (3): dmaengine: Add wrapper for device_tx_status callback ASoC: dmaengine-pcm: Rename and deprecate snd_dmaengine_pcm_pointer ASoC: dmaengine-pcm: Add support for querying stream position from DMA driver Laxman Dewangan (1): ASoC: tegra: use dmaengine based dma driver Liam Girdwood (2): ASoC: pcm: Clean up logging in soc_new_pcm() ASoC: dpcm: Allow FE to be opened without valid BE routes. M R Swami Reddy (2): ASoC: Support TI Isabelle Audio driver ASoC: MAINTAINERS: Add maintainer for TI Isabelle Audio driver MR Swami Reddy (1): ASoC: isabelle: Remove regmap_exit() Marek Belisko (1): ASoC: dapm: Fix compilation warning Mark Brown (61): ASoC: codecs: Refresh copyrights for Wolfson drivers ASoC: ac97: Remove empty remove() function ASoC: wm2000: Add register readability information ASoC: io: Use dev_get_regmap() if driver doesn't provide a regmap ASoC: wm8350: Convert to direct regmap API usage ASoC: dapm: Use devm_clk_get() ASoC: dapm: Bodge for lack of a widely available clk API ASoC: dapm: The clock API is even less consistent than thought Revert "ASoC: fsl_ssi: convert to use devm_clk_get" ASoC: lm59453: Unconstify dai_driver ASoC: max98095: Staticise non-exported functions and export jack detect ASoC: wm8903: Convert to devm_regmap_init_i2c() ASoC: wm8904: Convert to module_i2c_driver() ASoC: wm8904: Convert to devm_regmap_init_i2c() ASoC: wm8996: Remove write sequencer registers from the defaults table ASoC: wm8996: Convert to devm_regmap_init_i2c() ASoC: wm8903: Move pin configuration into I2C probe() function ASoC: wm8903: Make interrupt handler use regmap directly ASoC: wm8903: Move interrupt request to I2C probe ASoC: wm8903: Move register default changes to I2C probe Merge branch 'asoc-ab8500' into for-3.6 ASoC: wm5100: Remove stubs of ASoC-level register map code ASoC: da732x: Staticise non-exported symbol soc_codec_dev_da732x ASoC: isabelle: Staticise non-exported isabelle_dai ASoC: core: Add DOUBLE_R variants of the _RANGE controls ASoC: dapm: Try to add all routes even if one fails ASoC: io: Don't dereference regmap if we failed to get one ASoC: Add shared code for Wolfson Arizona class devices ASoC: wm5102: Initial driver Merge tag 'v3.5-rc4' into for-3.6 ASoC: wm8904: Rely entirely on the core for bias level management ASoC: wm8904: Move regulator acquisition and device identification to I2C ASoC: wm8904: Move register default setup into I2C probe() ASoC: wm8996: Move regulator notifier callbacks into I2C level ASoC: wm8996: Inline wm8996_reset() and optimise cache-only usage ASoC: wm8996: Move register default configuration to I2C probe ASoC: dwc: Bodge around continuing absence of clock API stubs ASoC: wm5102: Remove unused platform data header ASoC: dwc: Add missing __iomem annotations ASoC: spdif: Build separate RX and TX objects ASoC: wm8962: Log AIF configuration requested by hw_params() ASoC: core: Try to use regmap if the driver doesn't set up any I/O ASoC: wm8994: Don't suspend accessory detection ASoC: dwc: Staticise non-exported i2s_start() ASoC: arizona: Rename current rates tables to bclk_rates ASoC: arizona: Implement AIF clock configuration ASoC: arizona: Be more forgiving in BCLK selection ASoC: arizona: Enable ASYNCCLK domain for audio interfaces ASoC: wm5102: Allow routing through the ASRCs ASoC: wm1250-ev1: Flag all supported rates in the DAI ASoC: dapm: Mark widgets as dirty when a route is added ASoC: dapm: Allow routes to be deleted at runtime ASoC: arizona: Change DAPM routes for AIF clocks when we change them ASoC: arizona: Export dai_ops ASoC: arizona: Add IN4 to the mixer tables ASoC: wm5102: Fix cut'n'paste for digital volume registers ASoC: arizona: Support variable FLL VCO multipliers ASoC: wm5110: Add audio CODEC driver ASoC: wm8962: Redo early init of the part on resume ASoC: wm8994: Update micdet for irqdomain conversion ASoC: littlemill: Add userspace control of the WM1250 I/O Matthias Kaehlcke (1): ASoC: Free memory in the error paths of soc_of_parse_audio_routing() Ola Lilja (7): ALSA: pcm: Add debug-print helper function ASoC: core: Add widget SND_SOC_DAPM_CLOCK_SUPPLY ASoC: Ux500: Add platform-driver ASoC: codecs: Add AB8500 codec-driver mfd/ab8500: Move platform-data for ab8500-codec into mfd-driver ASoC: Ux500: Correct license strings ASoC: Ux500: Add machine-driver Qiao Zhou (1): ASoC: add ttc-dkb machine support Rajeev Kumar (3): ASoC: Add support for synopsys i2s controller as per ASoC framework. ASoC: Add support for SPEAr ASoC pcm layer. ASoC: STA529: Add support for STA529 Audio Codec Richard Zhao (1): ASoC: fsl_ssi: convert to use devm_clk_get Sachin Kamat (3): ASoC: cs42l52: Remove version.h header file inclusion ASoC: lm49453: Remove version.h header file inclusion ASoC: smdk_wm8994: Convert to use snd_soc_register_card() Scott Jiang (3): ASoC: add sport driver for bf6xx soc ASoC: add i2s dai driver for bf6xx soc ASoC: bfin: use dev_err to print error log instead of dev_dbg Simon Wilson (1): ASoC: twl6040: fix spelling mistake Stephen Warren (18): ASoC: wm8903: init GPIOs during I2C probe not codec probe ASoC: tegra+wm8903: move all GPIO setup into probe ASoC: tegra+wm8903: Use devm_gpio_request_one ASoC: tegra+wm8903: unconditionally free jack GPIOs in remove ASoC: tegra+alc5632: move all GPIO setup into probe ASoC: tegra+alc5632: unconditionally free jack GPIOs in remove ASoC: tegra+wm8903: simplify gpio tests in widget callbacks ASoC: tegra+wm8903: remove non-DT support for Seaboard ASoC: make snd_soc_dai_link more symmetrical ASoC: tegra: add .stream_name to CPU DAIs ASoC: tegra: statically define DAI link format ASoC: tegra: remove usage of rtd->codec ASoC: tegra: use DAI's not card's dev for dev_err ASoC: tegra: use regmap more directly ASoC: when initializing CPU DAI, don't duplicate any CODEC init ASoC: when removing a CPU DAI, clean up its DAPM context ASoC: factor out soc_remove_platform() ASoC: probe CODECs and platforms before DAIs and links Sylwester Nawrocki (2): ASoC: Convert S3C24XX I2S driver to gpiolib API ASoC: Convert S3C2412 I2S driver to gpiolib API Torben Hohn (1): ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define Uwe Kleine-König (1): ASoC: dmaengine_pcm: fix typo in comment Vipin Kumar (3): ASoC: SPEAr spdif_in: Add spdif IN support ASoC: spdif_receiver: Add support for spdif in Audio Codec ASoC: SPEAr spdif_out: Add spdif out support Zhangfei Gao (3): ASoC: mmp: add audio dma support ASoC: mmp: add sspa support ASoC: add mmp brownstone support MAINTAINERS | 2 + arch/arm/mach-ux500/board-mop500.c | 14 + include/linux/dmaengine.h | 6 + include/linux/mfd/abx500/ab8500-codec.h | 52 + include/linux/mfd/abx500/ab8500.h | 2 + include/linux/platform_data/mmp_audio.h | 22 + include/sound/designware_i2s.h | 69 + include/sound/dmaengine_pcm.h | 1 + include/sound/pcm.h | 11 + include/sound/soc-dapm.h | 12 + include/sound/soc.h | 77 +- include/sound/spear_dma.h | 35 + include/sound/spear_spdif.h | 29 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/blackfin/Kconfig | 21 +- sound/soc/blackfin/Makefile | 4 + sound/soc/blackfin/bf6xx-i2s.c | 234 +++ sound/soc/blackfin/bf6xx-sport.c | 422 ++++++ sound/soc/blackfin/bf6xx-sport.h | 82 + sound/soc/codecs/Kconfig | 31 + sound/soc/codecs/Makefile | 19 +- sound/soc/codecs/ab8500-codec.c | 2522 +++++++++++++++++++++++++++++++ sound/soc/codecs/ab8500-codec.h | 590 ++++++++ sound/soc/codecs/ac97.c | 6 - sound/soc/codecs/arizona.c | 937 ++++++++++++ sound/soc/codecs/arizona.h | 159 ++ sound/soc/codecs/cs42l52.c | 19 +- sound/soc/codecs/cs42l73.c | 20 +- sound/soc/codecs/da732x.c | 1627 ++++++++++++++++++++ sound/soc/codecs/da732x.h | 133 ++ sound/soc/codecs/da732x_reg.h | 654 ++++++++ sound/soc/codecs/isabelle.c | 1176 ++++++++++++++ sound/soc/codecs/isabelle.h | 143 ++ sound/soc/codecs/lm49453.c | 3 +- sound/soc/codecs/max98095.c | 5 +- sound/soc/codecs/ml26124.c | 5 +- sound/soc/codecs/spdif_receiver.c | 67 + sound/soc/codecs/sta529.c | 442 ++++++ sound/soc/codecs/tlv320aic3x.c | 40 +- sound/soc/codecs/tlv320aic3x.h | 27 +- sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/wm1250-ev1.c | 7 +- sound/soc/codecs/wm2000.c | 32 +- sound/soc/codecs/wm5100-tables.c | 2 +- sound/soc/codecs/wm5100.c | 11 +- sound/soc/codecs/wm5102.c | 903 +++++++++++ sound/soc/codecs/wm5102.h | 21 + sound/soc/codecs/wm5110.c | 950 ++++++++++++ sound/soc/codecs/wm5110.h | 21 + sound/soc/codecs/wm8350.c | 22 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8731.c | 1 + sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8903.c | 316 ++-- sound/soc/codecs/wm8904.c | 272 ++-- sound/soc/codecs/wm8960.c | 2 + sound/soc/codecs/wm8961.c | 2 + sound/soc/codecs/wm8962.c | 8 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 46 +- sound/soc/codecs/wm8996.c | 587 +++---- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/dwc/Kconfig | 9 + sound/soc/dwc/Makefile | 3 + sound/soc/dwc/designware_i2s.c | 455 ++++++ sound/soc/ep93xx/ep93xx-pcm.c | 2 +- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-audmux.h | 1 + sound/soc/fsl/imx-mc13783.c | 49 +- sound/soc/fsl/imx-pcm-dma.c | 2 +- sound/soc/fsl/imx-sgtl5000.c | 3 +- sound/soc/mxs/mxs-pcm.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/pxa/Kconfig | 42 + sound/soc/pxa/Makefile | 8 + sound/soc/pxa/brownstone.c | 174 +++ sound/soc/pxa/mmp-pcm.c | 297 ++++ sound/soc/pxa/mmp-sspa.c | 480 ++++++ sound/soc/pxa/mmp-sspa.h | 92 ++ sound/soc/pxa/ttc-dkb.c | 173 +++ sound/soc/samsung/littlemill.c | 7 + sound/soc/samsung/s3c2412-i2s.c | 10 +- sound/soc/samsung/s3c24xx-i2s.c | 10 +- sound/soc/samsung/smdk_wm8994.c | 36 +- sound/soc/sh/fsi.c | 48 +- sound/soc/soc-core.c | 328 +++- sound/soc/soc-dapm.c | 164 +- sound/soc/soc-dmaengine-pcm.c | 33 +- sound/soc/soc-io.c | 15 +- sound/soc/soc-pcm.c | 12 +- sound/soc/spear/spdif_in.c | 297 ++++ sound/soc/spear/spdif_in_regs.h | 60 + sound/soc/spear/spdif_out.c | 389 +++++ sound/soc/spear/spdif_out_regs.h | 79 + sound/soc/spear/spear_pcm.c | 214 +++ sound/soc/tegra/Kconfig | 3 +- sound/soc/tegra/tegra20_i2s.c | 94 +- sound/soc/tegra/tegra20_i2s.h | 1 - sound/soc/tegra/tegra20_spdif.c | 36 +- sound/soc/tegra/tegra20_spdif.h | 1 - sound/soc/tegra/tegra30_i2s.c | 85 +- sound/soc/tegra/tegra30_i2s.h | 1 - sound/soc/tegra/tegra_alc5632.c | 32 +- sound/soc/tegra/tegra_pcm.c | 115 ++ sound/soc/tegra/tegra_pcm.h | 2 + sound/soc/tegra/tegra_wm8753.c | 8 +- sound/soc/tegra/tegra_wm8903.c | 259 ++-- sound/soc/tegra/trimslice.c | 30 +- sound/soc/ux500/Kconfig | 18 + sound/soc/ux500/Makefile | 6 + sound/soc/ux500/mop500.c | 113 ++ sound/soc/ux500/mop500_ab8500.c | 431 ++++++ sound/soc/ux500/mop500_ab8500.h | 22 + sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/ux500/ux500_msp_i2s.c | 2 +- sound/soc/ux500/ux500_pcm.c | 318 ++++ sound/soc/ux500/ux500_pcm.h | 35 + 126 files changed, 16719 insertions(+), 1344 deletions(-) create mode 100644 include/linux/mfd/abx500/ab8500-codec.h create mode 100644 include/linux/platform_data/mmp_audio.h create mode 100644 include/sound/designware_i2s.h create mode 100644 include/sound/spear_dma.h create mode 100644 include/sound/spear_spdif.h create mode 100644 sound/soc/blackfin/bf6xx-i2s.c create mode 100644 sound/soc/blackfin/bf6xx-sport.c create mode 100644 sound/soc/blackfin/bf6xx-sport.h create mode 100644 sound/soc/codecs/ab8500-codec.c create mode 100644 sound/soc/codecs/ab8500-codec.h create mode 100644 sound/soc/codecs/arizona.c create mode 100644 sound/soc/codecs/arizona.h create mode 100644 sound/soc/codecs/da732x.c create mode 100644 sound/soc/codecs/da732x.h create mode 100644 sound/soc/codecs/da732x_reg.h create mode 100644 sound/soc/codecs/isabelle.c create mode 100644 sound/soc/codecs/isabelle.h create mode 100644 sound/soc/codecs/spdif_receiver.c create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/codecs/wm5102.c create mode 100644 sound/soc/codecs/wm5102.h create mode 100644 sound/soc/codecs/wm5110.c create mode 100644 sound/soc/codecs/wm5110.h create mode 100644 sound/soc/dwc/Kconfig create mode 100644 sound/soc/dwc/Makefile create mode 100644 sound/soc/dwc/designware_i2s.c create mode 100644 sound/soc/pxa/brownstone.c create mode 100644 sound/soc/pxa/mmp-pcm.c create mode 100644 sound/soc/pxa/mmp-sspa.c create mode 100644 sound/soc/pxa/mmp-sspa.h create mode 100644 sound/soc/pxa/ttc-dkb.c create mode 100644 sound/soc/spear/spdif_in.c create mode 100644 sound/soc/spear/spdif_in_regs.h create mode 100644 sound/soc/spear/spdif_out.c create mode 100644 sound/soc/spear/spdif_out_regs.h create mode 100644 sound/soc/spear/spear_pcm.c create mode 100644 sound/soc/ux500/mop500.c create mode 100644 sound/soc/ux500/mop500_ab8500.c create mode 100644 sound/soc/ux500/mop500_ab8500.h create mode 100644 sound/soc/ux500/ux500_pcm.c create mode 100644 sound/soc/ux500/ux500_pcm.h

2 1

[alsa-devel] [PATCH] ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define
by Torben Hohn 18 Jul '12

18 Jul '12

The MX31 Audmux also has 7 Ports. This patch adds the missing define, and makes the debugfs code iterate over that port too. Signed-off-by: Torben Hohn <torbenh(a)linutronix.de> --- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-audmux.h | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 0803274..e7c800e 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -156,7 +156,7 @@ static void __init audmux_debugfs_init(void) return; } - for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) { + for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) { snprintf(buf, sizeof(buf), "ssi%d", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h index 04ebbab..b8ff44b 100644 --- a/sound/soc/fsl/imx-audmux.h +++ b/sound/soc/fsl/imx-audmux.h @@ -14,6 +14,7 @@ #define MX31_AUDMUX_PORT4_SSI_PINS_4 3 #define MX31_AUDMUX_PORT5_SSI_PINS_5 4 #define MX31_AUDMUX_PORT6_SSI_PINS_6 5 +#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 #define MX51_AUDMUX_PORT1_SSI0 0 #define MX51_AUDMUX_PORT2_SSI1 1 -- 1.7.10

3 2

[alsa-devel] [PATCH] ASoC: littlemill: Add userspace control of the WM1250 I/O
by Mark Brown 18 Jul '12

18 Jul '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/littlemill.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index c82c646..ee52c8a0 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -211,6 +211,11 @@ static int bbclk_ev(struct snd_soc_dapm_widget *w, return 0; } +static const struct snd_kcontrol_new controls[] = { + SOC_DAPM_PIN_SWITCH("WM1250 Input"), + SOC_DAPM_PIN_SWITCH("WM1250 Output"), +}; + static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), @@ -282,6 +287,8 @@ static struct snd_soc_card littlemill = { .set_bias_level = littlemill_set_bias_level, .set_bias_level_post = littlemill_set_bias_level_post, + .controls = controls, + .num_controls = ARRAY_SIZE(controls), .dapm_widgets = widgets, .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_paths, -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Update micdet for irqdomain conversion
by Mark Brown 18 Jul '12

18 Jul '12

The conversion of the core driver to irqdomains means that we don't need and irq_base to have working interrupts so use wm8994_request_irq() to deal with looking up the interrupt number for the micdet IRQ. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a4aafdb..235577a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3692,9 +3692,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) if (wm8994->pdata && wm8994->pdata->micdet_irq) wm8994->micdet_irq = wm8994->pdata->micdet_irq; - else if (wm8994->pdata && wm8994->pdata->irq_base) - wm8994->micdet_irq = wm8994->pdata->irq_base + - WM8994_IRQ_MIC1_DET; pm_runtime_enable(codec->dev); pm_runtime_idle(codec->dev); @@ -3833,6 +3830,10 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) dev_warn(codec->dev, "Failed to request Mic detect IRQ: %d\n", ret); + } else { + wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_MIC1_DET, + wm8958_mic_irq, "Mic detect", + wm8994); } } -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ALSA: hda - make sure alc268 does not OOPS on codec parse
by David Henningsson 18 Jul '12

18 Jul '12

A recent commit made patch_alc268 call snd_hda_pick_fixup with NULL quirk pointer. Make sure we do not reference that NULL pointer. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index f7520b9..647218d 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -727,7 +727,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, models++; } } - if (id < 0) { + if (id < 0 && quirk) { q = snd_pci_quirk_lookup(codec->bus->pci, quirk); if (q) { id = q->value; @@ -736,7 +736,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, #endif } } - if (id < 0) { + if (id < 0 && quirk) { for (q = quirk; q->subvendor; q++) { unsigned int vendorid = q->subdevice | (q->subvendor << 16); -- 1.7.9.5

2 1

[alsa-devel] [RFC PATCH] introduce snd-cmi8328: C-Media CMI8328 driver
by Ondrej Zary 18 Jul '12

18 Jul '12

Hello, this patch introduces snd-cmi8328 driver for C-Media CMI8328-based sound cards, such as AudioExcel AV500. It supports PCM playback and capture (full-duplex) through wss_lib, gameport, OPL3 and MPU401. The AV500 card has onboard Dream wavetable synth connected to the MPU401 port and Aux 1 input internally which works too. The CDROM interface is not supported (as the drivers for these CDROMs were removed from the kernel some time ago). snd_legacy_find_free_ioport() function is present in a couple of ISA sound drivers so perhaps it should be moved to sound/initval.h diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 52064cf..a38d964 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -117,6 +117,18 @@ config SND_AZT2320 To compile this driver as a module, choose M here: the module will be called snd-azt2320. +config SND_CMI8328 + tristate "C-Media CMI8328" + select SND_WSS_LIB + select SND_OPL3_LIB + select SND_MPU401_UART + help + Say Y here to include support for soundcards based on the + C-Media CMI8328 chip. + + To compile this driver as a module, choose M here: the module + will be called snd-cmi8328. + config SND_CMI8330 tristate "C-Media CMI8330" select SND_WSS_LIB diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 8d781e4..9a15f14 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -6,6 +6,7 @@ snd-adlib-objs := adlib.o snd-als100-objs := als100.o snd-azt2320-objs := azt2320.o +snd-cmi8328-objs := cmi8328.o snd-cmi8330-objs := cmi8330.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o @@ -16,6 +17,7 @@ snd-sscape-objs := sscape.o obj-$(CONFIG_SND_ADLIB) += snd-adlib.o obj-$(CONFIG_SND_ALS100) += snd-als100.o obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o +obj-$(CONFIG_SND_CMI8328) += snd-cmi8328.o obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c new file mode 100644 index 0000000..2691116 --- /dev/null +++ b/sound/isa/cmi8328.c @@ -0,0 +1,482 @@ +/* + * Driver for C-Media CMI8328-based soundcards, such as AudioExcel AV500 + * Copyright (c) 2012 Ondrej Zary + * + * AudioExcel AV500 card consists of: + * - CMI8328 - main chip (SB Pro emulation, gameport, OPL3, MPU401, CD-ROM) + * - CS4231A - WSS codec + * - Dream SAM9233+GMS950400+RAM+ROM: Wavetable MIDI, connected to MPU401 + */ + +#include <linux/init.h> +#include <linux/isa.h> +#include <linux/module.h> +#include <linux/gameport.h> +#include <asm/dma.h> +#include <sound/core.h> +#include <sound/wss.h> +#include <sound/opl3.h> +#include <sound/mpu401.h> +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include <sound/initval.h> + +MODULE_AUTHOR("Ondrej Zary <linux(a)rainbow-software.org>"); +MODULE_DESCRIPTION("C-Media CMI8328"); +MODULE_LICENSE("GPL"); + +#if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE) +#define SUPPORT_JOYSTICK 1 +#endif + +/* I/O port is configured by jumpers on the card to one of these */ +static int cmi8328_ports[] = { 0x530, 0xe80, 0xf40, 0x604 }; +#define CMI8328_MAX ARRAY_SIZE(cmi8328_ports) + +static int index[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = -1}; +static char *id[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = NULL}; +static long port[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_PORT}; +static int irq[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_IRQ}; +static int dma1[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_DMA}; +static int dma2[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_DMA}; +static long mpuport[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_PORT}; +static int mpuirq[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_IRQ}; +#ifdef SUPPORT_JOYSTICK +static bool gameport[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = true}; +#endif + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for CMI8328 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for CMI8328 soundcard."); + +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for CMI8328 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for CMI8328 driver."); +module_param_array(dma1, int, NULL, 0444); +MODULE_PARM_DESC(dma1, "DMA1 for CMI8328 driver."); +module_param_array(dma2, int, NULL, 0444); +MODULE_PARM_DESC(dma2, "DMA2 for CMI8328 driver."); + +module_param_array(mpuport, long, NULL, 0444); +MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8328 driver."); +module_param_array(mpuirq, int, NULL, 0444); +MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8328 MPU-401 port."); +#ifdef SUPPORT_JOYSTICK +module_param_array(gameport, bool, NULL, 0444); +MODULE_PARM_DESC(gameport, "Enable gameport."); +#endif + +struct snd_cmi8328 { + u16 port; + u8 cfg[3]; + u8 wss_cfg; + struct snd_card *card; + struct snd_wss *wss; +#ifdef SUPPORT_JOYSTICK + struct gameport *gameport; +#endif +}; + +static int __devinit snd_legacy_find_free_ioport(int *port_table, int size) +{ + while (*port_table != -1) { + if (request_region(*port_table, size, "ALSA test")) { + release_region(*port_table, size); + return *port_table; + } + port_table++; + } + return -1; +} + +/* CMI8328 configuration registers */ +#define CFG1 0x61 +#define CFG1_SB_DISABLE (1 << 0) +#define CFG1_GAMEPORT (1 << 1) +/* + * bit 0: SB: 0=enabled, 1=disabled + * bit 1: gameport: 0=disabled, 1=enabled + * bits 2-4: SB IRQ: 001=3, 010=5, 011=7, 100=9, 101=10, 110=11 + * bits 5-6: SB DMA: 00=disabled (when SB disabled), 01=DMA0, 10=DMA1, 11=DMA3 + * bit 7: SB port: 0=0x220, 1=0x240 + */ +#define CFG2 0x62 +#define CFG2_MPU_ENABLE (1 << 2) +/* + * bits 0-1: CD-ROM mode: 00=disabled, 01=Panasonic, 10=Sony/Mitsumi/Wearnes, + 11=IDE + * bit 2: MPU401: 0=disabled, 1=enabled + * bits 3-4: MPU401 IRQ: 00=3, 01=5, 10=7, 11=9, + * bits 5-7: MPU401 port: 000=0x300, 001=0x310, 010=0x320, 011=0x330, 100=0x332, + 101=0x334, 110=0x336 + */ +#define CFG3 0x63 +/* + * bits 0-2: CD-ROM IRQ: 000=disabled, 001=3, 010=5, 011=7, 100=9, 101=10, + 110=11 + * bits 3-4: CD-ROM DMA: 00=disabled, 01=DMA0, 10=DMA1, 11=DMA3 + * bits 5-7: CD-ROM port: 000=0x300, 001=0x310, 010=0x320, 011=0x330, 100=0x340, + 101=0x350, 110=0x360, 111=0x370 + */ + +static u8 snd_cmi8328_cfg_read(u16 port, u8 reg) +{ + outb(0x43, port + 3); + outb(0x21, port + 3); + outb(reg, port + 3); + return inb(port); +} + +static void snd_cmi8328_cfg_write(u16 port, u8 reg, u8 val) +{ + outb(0x43, port + 3); + outb(0x21, port + 3); + outb(reg, port + 3); + outb(val, port + 3); /* yes, value goes to the same port as index */ +} + +static void snd_cmi8328_cfg_save(u16 port, u8 cfg[]) +{ + cfg[0] = snd_cmi8328_cfg_read(port, CFG1); + cfg[1] = snd_cmi8328_cfg_read(port, CFG2); + cfg[2] = snd_cmi8328_cfg_read(port, CFG3); +} + +static void snd_cmi8328_cfg_restore(u16 port, u8 cfg[]) +{ + snd_cmi8328_cfg_write(port, CFG1, cfg[0]); + snd_cmi8328_cfg_write(port, CFG2, cfg[1]); + snd_cmi8328_cfg_write(port, CFG3, cfg[2]); +} + +static int __devinit snd_cmi8328_mixer(struct snd_wss *chip) +{ + struct snd_card *card; + struct snd_ctl_elem_id id1, id2; + int err; + + card = chip->card; + + memset(&id1, 0, sizeof(id1)); + memset(&id2, 0, sizeof(id2)); + id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + /* rename AUX0 switch to CD */ + strcpy(id1.name, "Aux Playback Switch"); + strcpy(id2.name, "CD Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + /* rename AUX0 volume to CD */ + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "CD Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + /* rename AUX1 switch to Synth */ + strcpy(id1.name, "Aux Playback Switch"); + id1.index = 1; + strcpy(id2.name, "Synth Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + /* rename AUX1 volume to Synth */ + strcpy(id1.name, "Aux Playback Volume"); + id1.index = 1; + strcpy(id2.name, "Synth Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + + return 0; +} + +/* find index of an item in "-1"-ended array */ +int array_find(int array[], int item) +{ + int i; + + for (i = 0; array[i] != -1; i++) + if (array[i] == item) + return i; + + return -1; +} + +static int __devinit snd_cmi8328_probe(struct device *pdev, unsigned int ndev) +{ + struct snd_card *card; + struct snd_opl3 *opl3; + struct snd_cmi8328 *cmi; +#ifdef SUPPORT_JOYSTICK + struct resource *res; +#endif + int err, pos; + static int mpu_ports[] = { 0x330, 0x300, 0x310, 0x320, 0x332, 0x334, + 0x336, -1 }; + static u8 mpu_port_bits[] = { 3, 0, 1, 2, 4, 5, 6 }; + static int mpu_irqs[] = { 9, 7, 5, 3, -1 }; + static u8 mpu_irq_bits[] = { 3, 2, 1, 0 }; + static int irqs[] = { 9, 10, 11, 7, -1 }; + static u8 irq_bits[] = { 2, 3, 4, 1 }; + static int dma1s[] = { 3, 1, 0, -1 }; + static u8 dma_bits[] = { 3, 2, 1 }; + static int dma2s[][2] = { {1, -1}, {0, -1}, {-1, -1}, {0, -1} }; + u16 port = cmi8328_ports[ndev]; + u8 val; + + /* 0xff is invalid configuration (but settable - hope it isn't set) */ + if (snd_cmi8328_cfg_read(port, CFG1) == 0xff) + return -ENODEV; + /* the SB disable bit must NEVER EVER be cleared or the WSS dies */ + snd_cmi8328_cfg_write(port, CFG1, CFG1_SB_DISABLE); + if (snd_cmi8328_cfg_read(port, CFG1) != CFG1_SB_DISABLE) + return -ENODEV; + /* disable everything first */ + snd_cmi8328_cfg_write(port, CFG2, 0); /* disable CDROM and MPU401 */ + snd_cmi8328_cfg_write(port, CFG3, 0); /* disable CDROM IRQ and DMA */ + + if (irq[ndev] == SNDRV_AUTO_IRQ) { + irq[ndev] = snd_legacy_find_free_irq(irqs); + if (irq[ndev] < 0) { + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + return -EBUSY; + } + } + if (dma1[ndev] == SNDRV_AUTO_DMA) { + dma1[ndev] = snd_legacy_find_free_dma(dma1s); + if (dma1[ndev] < 0) { + snd_printk(KERN_ERR "unable to find a free DMA1\n"); + return -EBUSY; + } + } + if (dma2[ndev] == SNDRV_AUTO_DMA) { + dma2[ndev] = snd_legacy_find_free_dma(dma2s[dma1[ndev] % 4]); + if (dma2[ndev] < 0) { + snd_printk(KERN_WARNING "unable to find a free DMA2, full-duplex will not work\n"); + dma2[ndev] = -1; + } + } + /* configure WSS IRQ... */ + pos = array_find(irqs, irq[ndev]); + if (pos < 0) { + snd_printk(KERN_ERR "invalid IRQ %d\n", irq[ndev]); + return -EINVAL; + } + val = irq_bits[pos] << 3; + /* ...and DMA... */ + pos = array_find(dma1s, dma1[ndev]); + if (pos < 0) { + snd_printk(KERN_ERR "invalid DMA1 %d\n", dma1[ndev]); + return -EINVAL; + } + val |= dma_bits[pos]; + /* ...and DMA2 */ + if (dma2[ndev] >= 0 && dma1[ndev] != dma2[ndev]) { + pos = array_find(dma2s[dma1[ndev]], dma2[ndev]); + if (pos < 0) { + snd_printk(KERN_ERR "invalid DMA2 %d\n", dma2[ndev]); + return -EINVAL; + } + val |= 0x04; /* enable separate capture DMA */ + } + outb(val, port); + + err = snd_card_create(index[ndev], id[ndev], THIS_MODULE, + sizeof(struct snd_cmi8328), &card); + if (err < 0) + return err; + cmi = card->private_data; + cmi->card = card; + cmi->port = port; + cmi->wss_cfg = val; + snd_card_set_dev(card, pdev); + + err = snd_wss_create(card, port + 4, -1, irq[ndev], dma1[ndev], + dma2[ndev], WSS_HW_DETECT, 0, &cmi->wss); + if (err < 0) + goto error; + + err = snd_wss_pcm(cmi->wss, 0, NULL); + if (err < 0) + goto error; + + err = snd_wss_mixer(cmi->wss); + if (err < 0) + goto error; + err = snd_cmi8328_mixer(cmi->wss); + if (err < 0) + goto error; + + if (snd_wss_timer(cmi->wss, 0, NULL) < 0) + snd_printk(KERN_WARNING "error initializing WSS timer\n"); + + if (mpuport[ndev] == SNDRV_AUTO_PORT) { + mpuport[ndev] = snd_legacy_find_free_ioport(mpu_ports, 2); + if (mpuport[ndev] < 0) + snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); + } + if (mpuirq[ndev] == SNDRV_AUTO_IRQ) { + mpuirq[ndev] = snd_legacy_find_free_irq(mpu_irqs); + if (mpuirq[ndev] < 0) + snd_printk(KERN_ERR "unable to find a free MPU401 IRQ\n"); + } + /* enable and configure MPU401 */ + if (mpuport[ndev] > 0 && mpuirq[ndev] > 0) { + val = CFG2_MPU_ENABLE; + pos = array_find(mpu_ports, mpuport[ndev]); + if (pos < 0) + snd_printk(KERN_WARNING "invalid MPU401 port 0x%lx\n", + mpuport[ndev]); + else { + val |= mpu_port_bits[pos] << 5; + pos = array_find(mpu_irqs, mpuirq[ndev]); + if (pos < 0) + snd_printk(KERN_WARNING "invalid MPU401 IRQ %d\n", + mpuirq[ndev]); + else { + val |= mpu_irq_bits[pos] << 3; + snd_cmi8328_cfg_write(port, CFG2, val); + if (snd_mpu401_uart_new(card, 0, + MPU401_HW_MPU401, mpuport[ndev], + 0, mpuirq[ndev], NULL) < 0) + snd_printk(KERN_ERR "error initializing MPU401\n"); + } + } + } + /* OPL3 is hardwired to 0x388 and cannot be disabled */ + if (snd_opl3_create(card, 0x388, 0x38a, OPL3_HW_AUTO, 0, &opl3) < 0) + snd_printk(KERN_ERR "error initializing OPL3\n"); + else + if (snd_opl3_hwdep_new(opl3, 0, 1, NULL) < 0) + snd_printk(KERN_WARNING "error initializing OPL3 hwdep\n"); + + strcpy(card->driver, "CMI8328"); + strcpy(card->shortname, "C-Media CMI8328"); + sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d,%d", + card->shortname, cmi->wss->port, irq[ndev], dma1[ndev], + (dma2[ndev] >= 0) ? dma2[ndev] : dma1[ndev]); + + dev_set_drvdata(pdev, card); + err = snd_card_register(card); + if (err < 0) + goto error; +#ifdef SUPPORT_JOYSTICK + if (!gameport[ndev]) + return 0; + /* gameport is hardwired to 0x200 */ + res = request_region(0x200, 8, "CMI8328 gameport"); + if (!res) + snd_printk(KERN_WARNING "unable to allocate gameport I/O port\n"); + else { + struct gameport *gp = cmi->gameport = gameport_allocate_port(); + if (!cmi->gameport) + release_and_free_resource(res); + else { + gameport_set_name(gp, "CMI8328 Gameport"); + gameport_set_phys(gp, "%s/gameport0", dev_name(pdev)); + gameport_set_dev_parent(gp, pdev); + gp->io = 0x200; + gameport_set_port_data(gp, res); + /* Enable gameport */ + snd_cmi8328_cfg_write(port, CFG1, + CFG1_SB_DISABLE | CFG1_GAMEPORT); + gameport_register_port(gp); + } + } +#endif + return 0; +error: + snd_card_free(card); + + return err; +} + +static int __devexit snd_cmi8328_remove(struct device *pdev, unsigned int dev) +{ + struct snd_card *card = dev_get_drvdata(pdev); +#ifdef SUPPORT_JOYSTICK + struct snd_cmi8328 *cmi = card->private_data; + if (cmi->gameport) { + struct resource *res = gameport_get_port_data(cmi->gameport); + gameport_unregister_port(cmi->gameport); + release_and_free_resource(res); + } +#endif + /* disable everything */ + snd_cmi8328_cfg_write(cmi->port, CFG1, CFG1_SB_DISABLE); + snd_cmi8328_cfg_write(cmi->port, CFG2, 0); + snd_cmi8328_cfg_write(cmi->port, CFG3, 0); + snd_card_free(card); + dev_set_drvdata(pdev, NULL); + return 0; +} + +#ifdef CONFIG_PM +static int snd_cmi8328_suspend(struct device *pdev, unsigned int n, + pm_message_t state) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_cmi8328 *cmi; + + if (!card) /* ignore absent devices */ + return 0; + cmi = card->private_data; + snd_cmi8328_cfg_save(cmi->port, cmi->cfg); + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(cmi->wss->pcm); + cmi->wss->suspend(cmi->wss); + + return 0; +} + +static int snd_cmi8328_resume(struct device *pdev, unsigned int n) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_cmi8328 *cmi; + + if (!card) /* ignore absent devices */ + return 0; + cmi = card->private_data; + snd_cmi8328_cfg_restore(cmi->port, cmi->cfg); + outb(cmi->wss_cfg, cmi->port); + cmi->wss->resume(cmi->wss); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} +#endif + +static struct isa_driver snd_cmi8328_driver = { + .probe = snd_cmi8328_probe, + .remove = __devexit_p(snd_cmi8328_remove), +#ifdef CONFIG_PM + .suspend = snd_cmi8328_suspend, + .resume = snd_cmi8328_resume, +#endif + .driver = { + .name = "cmi8328" + }, +}; + +static int __init alsa_card_cmi8328_init(void) +{ + return isa_register_driver(&snd_cmi8328_driver, CMI8328_MAX); +} + +static void __exit alsa_card_cmi8328_exit(void) +{ + isa_unregister_driver(&snd_cmi8328_driver); +} + +module_init(alsa_card_cmi8328_init) +module_exit(alsa_card_cmi8328_exit) -- Ondrej Zary

2 2

[alsa-devel] HDA_PINCFG_NO_HP_FIXUP question
by David Henningsson 18 Jul '12

18 Jul '12

Hi, I'm trying to fix a bug where the docking station is not working. I started by adding pins for them (because that was not present), like this: [ALC269_FIXUP_LENOVO_DOCK] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { { 0x19, 0x23a11040 }, /* dock mic */ { 0x1b, 0x2121103f }, /* dock headphone */ { } }, }, However, this caused the real (non-dock) headphone out to "fixed up" to a line out, which is wrong. I'm a little unsure about the best way to resolve this, I can think of a few different options: 1) Also override the non-dock headphone out to have sequence number 0xf 2) Set parse_flags to HDA_PINCFG_NO_HP_FIXUP in patch_alc269 3) Try to improve the fix up algorithm to not fix up this particular case, but since I don't know when and where this was needed in the first place, I'm not sure how to do that without throwing the baby out with the bath water...? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

2 1

[alsa-devel] [alsa-utils] [PATCH 1/3] aplay: Fix signal.h include
by Arun Raghavan 18 Jul '12

18 Jul '12

The correct header to include is <signal.h> not <sys/signal.h>. --- aplay/aplay.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 8462484..932497d 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -42,10 +42,10 @@ #include <alsa/asoundlib.h> #include <assert.h> #include <termios.h> +#include <signal.h> #include <sys/poll.h> #include <sys/uio.h> #include <sys/time.h> -#include <sys/signal.h> #include <sys/stat.h> #include <sys/types.h> #include <endian.h> -- 1.7.8.6

2 4

[alsa-devel] [PATCH] ALSA: hda - Add support for Realtek ALC282
by David Henningsson 18 Jul '12

18 Jul '12

This codec has a separate dmic path (separate dmic only ADC), and thus it looks mostly like ALC275. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/1025377 Tested-by: Ray Chen <ray.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) One of the machines we are about enable has this chip installed. Ray Chen has verified that this patch makes the internal speaker and microphone work. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 04e9243..ec14224 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7016,6 +7016,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 }, { .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 }, { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 }, + { .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, -- 1.7.9.5

2 1

[alsa-devel] how to unload alsa kernel driver
by Liu, George 18 Jul '12

18 Jul '12

I'm new to kernel development and is trying out the dummy alsa driver. I can compile and install snd_dummy driver on my Ubuntu, but how do I remove it? It seems that after installation, snd_pcm and snd automatically depends on snd_dummy, so rmmod will not work. Alsa force-unload snd-dummy also not work. Can somebody help? I don't want to restart my machine every time. Thanks George

1 0

[alsa-devel] ALSA server upgrade
by Jaroslav Kysela 17 Jul '12

17 Jul '12

Hello, I will do the complete ALSA server upgrade in next days. It means that all services may be affected for some time (the devel mailing list, web server, ftp server, GIT server). Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

1 1

Re: [alsa-devel] [PATCH V2] Add missing modules aliases to get sound working on omap devices
by Mark Brown 17 Jul '12

17 Jul '12

On Tue, Jul 17, 2012 at 03:31:11PM +0200, Guillaume Gardet wrote: > >Please resend in the format from SubmittingPatches rather than as an > >attachment. > Ok. Please find the patch below. > > This patch add missing modules aliases to get sound working on omap devices. I'm not sure you've looked at all of SubmittingPatches...

1 0

[alsa-devel] [alsa-lib] Assorted patches for building on Android
by Arun Raghavan 17 Jul '12

17 Jul '12

Hello, This patch series is required for building alsa-lib on Android (I'm using this for my current PulseAudio-on-Android work). None of these changes introduce anything Android-specific, though. These could likely also be useful for other, non-GNU userspace implementations. I'll eventually clean up and send in the build-system changes that I have, but this will likely need some more dialogue, since the CyanogenMod folks have a completely different approach from what we've taken (and I believe their solution, while simpler, is less complete). Regards, Arun

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Re: [alsa-devel] 3.4.4: Oops in snd_hda_codec_realtek (alc_auto_create_multi_out_ctls)
by Takashi Iwai 17 Jul '12

17 Jul '12

At Mon, 9 Jul 2012 13:40:09 -0600, Robert Hancock wrote: > > I've got a sort of industrial portable PC that uses a Supermicro C2SBX > motherboard. Running an RHEL6 kernel (2.6.32-ish) it works fine, but > if I run a 3.4.4 kernel (using the kernel-ml builds provided by ELRepo > for RHEL6) it blows up on boot inside snd_hda_codec_realtek. Below is > the dmesg output from bootup up to the oops, after I blacklisted > snd_hda_intel from auto-loading and then manually modprobed it. Is > this a known problem? I don't know of this issue, so need to check more. Could you load snd-hda-intel module with probe_only=1 option, and run alsa-info.sh with --no-upload option and give the output? Then I can try the emulator to see what's wrong. Also, of course, testing a newer kernel like 3.5-rc7 would be helpful, too. thanks, Takashi

1 0

Re: [alsa-devel] [sound:for-next 37/38] WARNING: Section mismatches on snd_opti9xx_resume()/snd_opti9xx_configure()
by Takashi Iwai 17 Jul '12

17 Jul '12

At Tue, 17 Jul 2012 18:12:38 +0800, Fengguang Wu wrote: > > Hi Ondrej, > > There are new compile warnings show up in > > tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next > head: 0c7293b6ea0b363c6715c6f0b2894fa63b8e9053 > commit: 5dd250728aa4cad49cfa18eb8ed11ba470ce382a [37/38] ALSA: snd-opti9xx: Implement suspend/resume > config: i386-allmodconfig (attached as .config) > > All warnings: > > > WARNING: sound/isa/opti9xx/snd-opti92x-ad1848.o(.text+0x1b2): Section mismatch in reference from the function snd_opti9xx_resume() to the function .devinit.text:snd_opti9xx_configure.isra.2() > The function snd_opti9xx_resume() references > the function __devinit snd_opti9xx_configure.isra.2(). > This is often because snd_opti9xx_resume lacks a __devinit > annotation or the annotation of snd_opti9xx_configure.isra.2 is wrong. > > WARNING: sound/isa/opti9xx/snd-opti92x-cs4231.o(.text+0xb6): Section mismatch in reference from the function snd_opti9xx_resume() to the function .devinit.text:snd_opti9xx_configure() > The function snd_opti9xx_resume() references > the function __devinit snd_opti9xx_configure(). > This is often because snd_opti9xx_resume lacks a __devinit > annotation or the annotation of snd_opti9xx_configure is wrong. > > WARNING: sound/isa/opti9xx/snd-opti93x.o(.text+0x1ea): Section mismatch in reference from the function snd_opti9xx_resume() to the function .devinit.text:snd_opti9xx_configure() > The function snd_opti9xx_resume() references > the function __devinit snd_opti9xx_configure(). > This is often because snd_opti9xx_resume lacks a __devinit > annotation or the annotation of snd_opti9xx_configure is wrong. Fixed now with the patch below. Thanks. Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: opti9xx: Fix section mismatch by PM support In the previous commit, snd_opti9xx_configure() is called from the resume handler but it's still marked as __devinit. Fix it. Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/opti9xx/opti92x-ad1848.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index d7ce012..f8fbe22 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -348,7 +348,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) -static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, +static int snd_opti9xx_configure(struct snd_opti9xx *chip, long port, int irq, int dma1, int dma2, long mpu_port, int mpu_irq) -- 1.7.10.4

1 0

[alsa-devel] [PATCH] snd-opti9xx: Implement suspend/resume
by Ondrej Zary 17 Jul '12

17 Jul '12

Implement suspend/resume support for Opti 92x and 93x chips. Tested with Opti 929A+AD1848 and Opti 931. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -136,8 +136,8 @@ struct snd_opti9xx { #ifdef OPTi93X unsigned long mc_indir_index; struct resource *res_mc_indir; - struct snd_wss *codec; #endif /* OPTi93X */ + struct snd_wss *codec; unsigned long pwd_reg; spinlock_t lock; @@ -870,9 +870,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) &codec); if (error < 0) return error; -#ifdef OPTi93X chip->codec = codec; -#endif error = snd_wss_pcm(codec, 0, &pcm); if (error < 0) return error; @@ -1053,11 +1051,55 @@ static int __devexit snd_opti9xx_isa_remove(struct device *devptr, return 0; } +#ifdef CONFIG_PM +static int snd_opti9xx_suspend(struct snd_card *card) +{ + struct snd_opti9xx *chip = card->private_data; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + chip->codec->suspend(chip->codec); + return 0; +} + +static int snd_opti9xx_resume(struct snd_card *card) +{ + struct snd_opti9xx *chip = card->private_data; + int error, xdma2; +#if defined(CS4231) || defined(OPTi93X) + xdma2 = dma2; +#else + xdma2 = -1; +#endif + + error = snd_opti9xx_configure(chip, port, irq, dma1, xdma2, + mpu_port, mpu_irq); + if (error) + return error; + chip->codec->resume(chip->codec); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} + +static int snd_opti9xx_isa_suspend(struct device *dev, unsigned int n, + pm_message_t state) +{ + return snd_opti9xx_suspend(dev_get_drvdata(dev)); +} + +static int snd_opti9xx_isa_resume(struct device *dev, unsigned int n) +{ + return snd_opti9xx_resume(dev_get_drvdata(dev)); +} +#endif + static struct isa_driver snd_opti9xx_driver = { .match = snd_opti9xx_isa_match, .probe = snd_opti9xx_isa_probe, .remove = __devexit_p(snd_opti9xx_isa_remove), - /* FIXME: suspend/resume */ +#ifdef CONFIG_PM + .suspend = snd_opti9xx_isa_suspend, + .resume = snd_opti9xx_isa_resume, +#endif .driver = { .name = DEV_NAME }, @@ -1123,12 +1165,29 @@ static void __devexit snd_opti9xx_pnp_remove(struct pnp_card_link * pcard) snd_opti9xx_pnp_is_probed = 0; } +#ifdef CONFIG_PM +static int snd_opti9xx_pnp_suspend(struct pnp_card_link *pcard, + pm_message_t state) +{ + return snd_opti9xx_suspend(pnp_get_card_drvdata(pcard)); +} + +static int snd_opti9xx_pnp_resume(struct pnp_card_link *pcard) +{ + return snd_opti9xx_resume(pnp_get_card_drvdata(pcard)); +} +#endif + static struct pnp_card_driver opti9xx_pnpc_driver = { .flags = PNP_DRIVER_RES_DISABLE, .name = "opti9xx", .id_table = snd_opti9xx_pnpids, .probe = snd_opti9xx_pnp_probe, .remove = __devexit_p(snd_opti9xx_pnp_remove), +#ifdef CONFIG_PM + .suspend = snd_opti9xx_pnp_suspend, + .resume = snd_opti9xx_pnp_resume, +#endif }; #endif -- Ondrej Zary

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Re: [alsa-devel] [PATCH] ALSA: hda - Add new GPU codec ID to snd-hda
by Takashi Iwai 17 Jul '12

17 Jul '12

At Mon, 16 Jul 2012 17:10:04 -0700, Aaron Plattner wrote: > > Vendor ID 0x10de0051 is used by a yet-to-be-named GPU chip. > > Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> > Acked-by: Andy Ritger <aritger(a)nvidia.com> > Reviewed-by: Daniel Dadap <ddadap(a)nvidia.com> Applied now. Thanks. Takashi > --- > sound/pci/hda/patch_hdmi.c | 2 ++ > 1 file changed, 2 insertions(+) > > diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c > index ad319d4..4efe741 100644 > --- a/sound/pci/hda/patch_hdmi.c > +++ b/sound/pci/hda/patch_hdmi.c > @@ -1902,6 +1902,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { > { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_generic_hdmi }, > { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_generic_hdmi }, > { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_generic_hdmi }, > +{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_generic_hdmi }, > { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, > { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, > { .id = 0x80860054, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi }, > @@ -1948,6 +1949,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0041"); > MODULE_ALIAS("snd-hda-codec-id:10de0042"); > MODULE_ALIAS("snd-hda-codec-id:10de0043"); > MODULE_ALIAS("snd-hda-codec-id:10de0044"); > +MODULE_ALIAS("snd-hda-codec-id:10de0051"); > MODULE_ALIAS("snd-hda-codec-id:10de0067"); > MODULE_ALIAS("snd-hda-codec-id:10de8001"); > MODULE_ALIAS("snd-hda-codec-id:17e80047"); > -- > 1.7.9.5 >

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Re: [alsa-devel] [regression] ALSA: hda - Handle open while transitioning to D3.
by Dylan Reid 17 Jul '12

17 Jul '12

On Sat, Jul 14, 2012 at 6:36 AM, Oleksij Rempel (fishor) <bug-track(a)fisher-privat.net> wrote: > dmesg is in attachment Thanks Oleksij, I didn't get much from that though. What is your powersave timeout set to? If you have time could you try adding the following and send me the output, I have two guesses about where it is spending it's time (no guesses about why yet though). I've got a couple of systems with a 269, I'll try to get some tests run on those tomorrow. diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 20580bf..fd8ce9e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4383,9 +4383,11 @@ static void hda_power_work(struct work_struct *work) } spin_unlock(&codec->power_lock); + snd_printd("power_work start %p\n", codec);; hda_call_codec_suspend(codec); if (bus->ops.pm_notify) bus->ops.pm_notify(bus); + snd_printd("power_work done %p\n", codec);; } static void hda_keep_power_on(struct hda_codec *codec) @@ -4426,7 +4428,9 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) } spin_unlock(&codec->power_lock); + snd_printd("power_up cancel sync start %p\n", codec); cancel_delayed_work_sync(&codec->power_work); + snd_printd("power_up cancel sync done %p\n", codec); spin_lock(&codec->power_lock); /* If the power down delayed work was cancelled above before starting, > > > On 14.07.2012 11:21, Dylan Reid wrote: >> >> On Jul 13, 2012 11:01 PM, "Oleksij Rempel (fishor)" >> <bug-track(a)fisher-privat.net <mailto:bug-track@fisher-privat.net>> wrote: >> > >> > Hallo Dylan, Takashi, >> > >> > I have regression after patch b4a91cf05c3 "ALSA: hda - Handle open >> while transitioning to D3." >> > >> > This regression i can reproduce only if my laptop works on batteries. >> After this patch it needs almost 1 minute more to start. The login >> screen usually plays start sound, but after this patch it wait until >> sound card is ready. >> > >> > My hardware: asus zenbook ux31e. >> > >> > Codec: Realtek ALC269VB >> > >> > 00:1b.0 0403: 8086:1c20 (rev 05) >> > Subsystem: 1043:1427 >> > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- >> ParErr- Stepping- SERR- FastB2B- DisINTx+ >> > Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >> >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- >> > Latency: 0, Cache Line Size: 64 bytes >> > Interrupt: pin A routed to IRQ 50 >> > Region 0: Memory at dfe00000 (64-bit, non-prefetchable) >> [size=16K] >> > Capabilities: <access denied> >> > Kernel driver in use: snd_hda_intel >> > Kernel modules: snd-hda-intel >> > >> >> Could you enable config_snd_debug and send the output during startup? >> I'm actually using the same codec on one of our systems. >> >> Thanks for testing. >> >> Dylan >> > >

1 0

[alsa-devel] [PATCH] ALSA: hda - Add DeviceID for Haswell HDA
by Wang Xingchao 16 Jul '12

16 Jul '12

this patch add proper id for Haswell HDA Controller. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_intel.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2b6392b..f530c40 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, CPT}," "{Intel, PPT}," "{Intel, LPT}," + "{Intel, HPT}," "{Intel, PBG}," "{Intel, SCH}," "{ATI, SB450}," @@ -3253,6 +3254,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE}, + /* Haswell */ + { PCI_DEVICE(0x8086, 0x0c0c), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE}, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- 1.7.9.5

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[alsa-devel] Kernel 3.2, and "speaker" controlling headphone path
by David Henningsson 16 Jul '12

16 Jul '12

It seems like on one machine the "Speaker Playback Volume" was part of the headphone path, in combination with PulseAudio this is leading to muted headphones. One of my colleagues wrote a workaround for that specific machine - I'm providing links in case you're interested: The patch: https://launchpadlibrarian.net/109499922/restrict_bass.patch Alsa-info: https://launchpadlibrarian.net/109251353/bad-kernel-yes-headphones.txt Bug link: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/994685 -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

2 2

[alsa-devel] When will Realtek High-Defintion audio codec ALC282 be supported?
by Lin, Mengdong 16 Jul '12

16 Jul '12

Hi, I'm going to have a platform with Realtek HDA ALC282 codec. But the driver has no support for ALC282 now. Does anybody know when ALC282 will be supported? I saw that another Realtek ALC280 is supported by just adding the codec id and still using 'patch_alc269'. I wonder if ALC282 can also be supported in this way if it's similar enough to other ALC2XX codecs. Could anyone share some idea about this? Thanks Mengdong

3 5

[alsa-devel] [PATCH] wss_lib: Fix resume on Yamaha OPL3-SAx
by Ondrej Zary 16 Jul '12

16 Jul '12

Yamaha OPL3-SAx chips don't resume properly when playback is running - garbage is played after resume. Restoring the CS4231_PLAYBK_FORMAT register last fixes the problem. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1656,6 +1656,10 @@ static void snd_wss_resume(struct snd_wss *chip) break; } } + /* Yamaha needs this to resume properly */ + if (chip->hardware == WSS_HW_OPL3SA2) + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, + chip->image[CS4231_PLAYBK_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); #if 1 snd_wss_mce_down(chip); -- Ondrej Zary

2 1

[alsa-devel] [PATCH] wss_lib: fix suspend/resume
by Ondrej Zary 16 Jul '12

16 Jul '12

By setting SNDRV_PCM_INFO_RESUME, wss_lib claims that it can restore the card state fully on resume. But in fact, it can't as DMA is not restored so any playback/capture running during suspend will fail to continue after resume. Remove SNDRV_PCM_INFO_RESUME flag from pcm info field to fix the problem. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1456,7 +1456,6 @@ static struct snd_pcm_hardware snd_wss_playback = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_SYNC_START), .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), -- Ondrej Zary

2 1

Re: [alsa-devel] [PATCH 1/1] hda_intel: activate COMBO mode for Intel client chipsets
by Wang Xingchao 16 Jul '12

16 Jul '12

Hi Seth, 2012/6/15 Seth Heasley <seth.heasley(a)intel.com>: > This patch activates the COMBO position_fix for recent Intel client chipsets. COMBO mode is the recommended setting for Intel chipsets and eliminates HD audio warnings in dmesg. This patch has been tested on Lynx Point, Panther Point, and Cougar Pont. Haswell HDMI audio works now on Sharkbay mobile platform, so i would like to test whether this patch is useful for Haswell. Would you tell me how to finish the test to prove it works better? thanks --xingchao > > Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> > --- > sound/pci/hda/hda_intel.c | 11 ++++++++--- > 1 files changed, 8 insertions(+), 3 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index 2b6392b..466a88c 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -535,6 +535,7 @@ enum { > #define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */ > #define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ > #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ > +#define AZX_DCAPS_POSFIX_COMBO (1 << 24) /* Use COMBO as default */ > > /* quirks for ATI SB / AMD Hudson */ > #define AZX_DCAPS_PRESET_ATI_SB \ > @@ -2728,6 +2729,10 @@ static int __devinit check_position_fix(struct azx *chip, int fix) > snd_printd(SFX "Using LPIB position fix\n"); > return POS_FIX_LPIB; > } > + if (chip->driver_caps & AZX_DCAPS_POSFIX_COMBO) { > + snd_printd(SFX "Using COMBO position fix\n"); > + return POS_FIX_COMBO; > + } > return POS_FIX_AUTO; > } > > @@ -3240,7 +3245,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* CPT */ > { PCI_DEVICE(0x8086, 0x1c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE }, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* PBG */ > { PCI_DEVICE(0x8086, 0x1d20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > @@ -3248,11 +3253,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* Panther Point */ > { PCI_DEVICE(0x8086, 0x1e20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE}, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* Lynx Point */ > { PCI_DEVICE(0x8086, 0x8c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > - AZX_DCAPS_BUFSIZE}, > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* SCH */ > { PCI_DEVICE(0x8086, 0x811b), > .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | > -- > 1.7.4.4 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/

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[alsa-devel] Status of USB audio class 2 drivers in ALSA
by Anand Joseph 15 Jul '12

15 Jul '12

I am working on building a multi-channel USB audio device using the PIC32MX2 series with dual WM8731 audio codecs. The goal is a low cost 4 channel USB audio device. While this might not be the best combination for multi-channel audio, I do not have a better solution as the XMOS series is not easily available in India. I would like to know the status of USB audio class 2 support in ALSA and is there anyway I can contribute. I did go through the USB specifications for audio class 2 devices, but an idea of the implementation in ALSA would help in writing the firmware. Regards, Anand

2 1

Re: [alsa-devel] [RESEND PATCH 1/2] ASoC: Convert S3C24XX I2S driver to gpiolib API
by Mark Brown 15 Jul '12

15 Jul '12

On Fri, Jul 13, 2012 at 07:22:44PM +0200, Sylwester Nawrocki wrote: > The s3c2410_gpio* calls are obsolete and have been scheduled for > removal since several kernel releases. Remove them and use common > gpiolib API. > This patch is a prerequisite for removal of the obsolete S3C24XX > SoC GPIO definitions. Applied both, thanks.

1 0

Re: [alsa-devel] [regression] ALSA: hda - Handle open while transitioning to D3.
by Dylan Reid 14 Jul '12

14 Jul '12

On Jul 13, 2012 11:01 PM, "Oleksij Rempel (fishor)" < bug-track(a)fisher-privat.net> wrote: > > Hallo Dylan, Takashi, > > I have regression after patch b4a91cf05c3 "ALSA: hda - Handle open while transitioning to D3." > > This regression i can reproduce only if my laptop works on batteries. After this patch it needs almost 1 minute more to start. The login screen usually plays start sound, but after this patch it wait until sound card is ready. > > My hardware: asus zenbook ux31e. > > Codec: Realtek ALC269VB > > 00:1b.0 0403: 8086:1c20 (rev 05) > Subsystem: 1043:1427 > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ > Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- > Latency: 0, Cache Line Size: 64 bytes > Interrupt: pin A routed to IRQ 50 > Region 0: Memory at dfe00000 (64-bit, non-prefetchable) [size=16K] > Capabilities: <access denied> > Kernel driver in use: snd_hda_intel > Kernel modules: snd-hda-intel > Could you enable config_snd_debug and send the output during startup? I'm actually using the same codec on one of our systems. Thanks for testing. Dylan

1 0

[alsa-devel] [PATCH - CA0132 HDA Codec 2/2] Change spin_lock()/unlock() to spin_lock_irqsave()/restore() in azx_pcm_trigger(). This prevents deadlock when an interrupt occurs, caused by chip->reg_lock contention.
by ian_minett＠creativelabs.com 14 Jul '12

14 Jul '12

From: Ian Minett <ian_minett(a)creativelabs.com> Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 796472d..0ccffde 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -551,6 +551,7 @@ enum { (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\ AZX_DCAPS_ALIGN_BUFSIZE) +/* quirks for Creative CTHDA */ #define AZX_DCAPS_PRESET_CTHDA \ (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY) @@ -1930,6 +1931,7 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_pcm_substream *s; int rstart = 0, start, nsync = 0, sbits = 0; int nwait, timeout; + unsigned long flags; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -1956,7 +1958,7 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) snd_pcm_trigger_done(s, substream); } - spin_lock(&chip->reg_lock); + spin_lock_irqsave(&chip->reg_lock, flags); if (nsync > 1) { /* first, set SYNC bits of corresponding streams */ if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) @@ -1980,7 +1982,7 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } azx_dev->running = start; } - spin_unlock(&chip->reg_lock); + spin_unlock_irqrestore(&chip->reg_lock, flags); if (start) { if (nsync == 1) return 0; -- 1.7.4.1

1 0

[alsa-devel] segmentation fault for snd_pcm_open after sudden removal for USB Camera (with USB Mic)
by Kuldeep Ghan 13 Jul '12

13 Jul '12

Hi All, I am seeing segmentation fault in alsa-lib while recording using USB Camera. Following is the test setup 1. Moto Razr (Android ICS, alsa-lib 1.0.24.1) is connected to Smart Dock 2. USB Camera (Philips SPC 1330NC) is connected to Razr through dock While Video recording from USB Camera, suddenly Power Cable of Dock or USB camera is removed, effectively resulting in USB camera removal from Razr. The recording continues with still view, but when I stop recording session, the snd_pcm_open to Play stop tone results in the crash as shown below. Although we are using internal device mic for capture, the USB camera has USB microphone which gets enumerated as card2, device 0. I did some debugging with help of prints in alsa-lib but was not really helpful to point to root cause. Logs with my added prints for snd_pcm_open successful(*successful_open.txt*) and segmentation fault(*segfault_open.txt*) case are attached. I am not very familiar with alsa-lib internals. Is this a known issue? Segmentation fault goes away with upgrade of alsa-lib to 1.0.25. But currently Kernel is 1.0.24, and we can't upgrade the kernel at this time, so is there any problem running lib 25 wither kernel 24? If so, do you have any suggestion of what fixes from .25 I should try back-porting?" 07-13 14:42:14.517 160 312 D AudioHardwareALSA: openOutputStream called for devices: 0x00000002 07-13 14:42:14.517 160 312 D Omap4ALSA: open called for devices 00000002 in mode 0... 07-13 14:42:14.524 160 312 F libc : Fatal signal 11 (SIGSEGV) at 0x00008140 (code=1) 07-13 14:42:14.532 6173 6173 I DEBUG : *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** *** 07-13 14:42:14.532 6173 6173 I DEBUG : Build fingerprint: 'ATT/edison_att_us/edison:4.0.4/6.7.2_245/3202:userdebug/test-keys' 07-13 14:42:14.532 6173 6173 I DEBUG : pid: 160, tid: 312 >>> /system/bin/mediaserver <<< 07-13 14:42:14.532 6173 6173 I DEBUG : signal 11 (SIGSEGV), code 1 (SEGV_MAPERR), fault addr 00008140 07-13 14:42:14.532 6173 6173 I DEBUG : tls: 41623f00 07-13 14:42:14.532 6173 6173 I DEBUG : 07-13 14:42:14.532 6173 6173 I DEBUG : r0 0050e7fd r1 00000004 r2 00000011 r3 00008130 07-13 14:42:14.532 6173 6173 I DEBUG : r4 b0009f58 r5 41623928 r6 4162390c r7 000081fc 07-13 14:42:14.532 6173 6173 I DEBUG : r8 0000841c r9 41623928 10 003e1dd0 fp 4162398c 07-13 14:42:14.532 6173 6173 I DEBUG : ip 00755f6c sp 416238d0 lr b00039f3 pc b00039f6 cpsr 48000030 07-13 14:42:14.532 6173 6173 I DEBUG : d0 636e75665f646e67 d1 6c75616665642e65 07-13 14:42:14.532 6173 6173 I DEBUG : d2 612f61736c612f74 d3 666e6f632e617365 07-13 14:42:14.532 6173 6173 I DEBUG : d4 0000000000000000 d5 0000100000000000 07-13 14:42:14.532 6173 6173 I DEBUG : d6 0000100045800000 d7 0000100000000000 07-13 14:42:14.532 6173 6173 I DEBUG : d8 0000000000000000 d9 0000000000000000 07-13 14:42:14.532 6173 6173 I DEBUG : d10 0000000000000000 d11 0000000000000000 07-13 14:42:14.532 6173 6173 I DEBUG : d12 0000000000000000 d13 0000000000000000 07-13 14:42:14.532 6173 6173 I DEBUG : d14 0000000000000000 d15 0000000000000000 07-13 14:42:14.532 6173 6173 I DEBUG : d16 0000000006ba1000 d17 3fdfffffe8287c00 07-13 14:42:14.532 6173 6173 I DEBUG : d18 3fe0000000000000 d19 3fe000000bebc200 07-13 14:42:14.532 6173 6173 I DEBUG : d20 0000000000000000 d21 0000000000000000 07-13 14:42:14.532 6173 6173 I DEBUG : d22 0000000000000000 d23 0000000000000000 07-13 14:42:14.540 6173 6173 I DEBUG : d24 0000000000000000 d25 0000000000000000 07-13 14:42:14.540 6173 6173 I DEBUG : d26 0000000000000000 d27 0000000000000000 07-13 14:42:14.540 6173 6173 I DEBUG : d28 0000000000000000 d29 0000000000000000 07-13 14:42:14.540 6173 6173 I DEBUG : d30 0000000000000000 d31 0000000000000000 07-13 14:42:14.540 6173 6173 I DEBUG : scr 80000010 07-13 14:42:14.540 6173 6173 I DEBUG : 07-13 14:42:15.126 6173 6173 I DEBUG : #00 pc b00039f6 /system/bin/linker 07-13 14:42:15.126 6173 6173 I DEBUG : #01 lr b00039f3 /system/bin/linker 07-13 14:42:15.126 6173 6173 I DEBUG : 07-13 14:42:15.126 6173 6173 I DEBUG : code around pc: 07-13 14:42:15.126 6173 6173 I DEBUG : b00039d4 00005b8e 47f0e92d f8d04604 469170b0 .[..-..G.F...p.F 07-13 14:42:15.126 6173 6173 I DEBUG : b00039e4 80acf8d0 f8d44608 f7fd10b4 f8d4ef4e .....F......N... 07-13 14:42:15.134 6173 6173 I DEBUG : b00039f4 f85330bc e0125021 187e0129 46495878 .0S.!P..).~.xXIF 07-13 14:42:15.134 6173 6173 I DEBUG : b0003a04 f7fd4440 b930ece8 09037b30 2a011e5a @D....0.0{..Z..* 07-13 14:42:15.134 6173 6173 I DEBUG : b0003a14 89f1d801 f8d4b931 f85660c0 2d005025 ....1....`V.%P.- 07-13 14:42:15.134 6173 6173 I DEBUG : 07-13 14:42:15.134 6173 6173 I DEBUG : code around lr: 07-13 14:42:15.134 6173 6173 I DEBUG : b00039d0 00011708 00005b8e 47f0e92d f8d04604 .....[..-..G.F.. 07-13 14:42:15.134 6173 6173 I DEBUG : b00039e0 469170b0 80acf8d0 f8d44608 f7fd10b4 .p.F.....F...... 07-13 14:42:15.134 6173 6173 I DEBUG : b00039f0 f8d4ef4e f85330bc e0125021 187e0129 N....0S.!P..).~. 07-13 14:42:15.134 6173 6173 I DEBUG : b0003a00 46495878 f7fd4440 b930ece8 09037b30 xXIF@D....0.0{.. 07-13 14:42:15.134 6173 6173 I DEBUG : b0003a10 2a011e5a 89f1d801 f8d4b931 f85660c0 Z..*....1....`V. 07-13 14:42:15.134 6173 6173 I DEBUG : 07-13 14:42:15.134 6173 6173 I DEBUG : memory map around addr 00008140: 07-13 14:42:15.134 6173 6173 I DEBUG : (no map below) 07-13 14:42:15.134 6173 6173 I DEBUG : (no map for address) 07-13 14:42:15.134 6173 6173 I DEBUG : 003dc000-0047f000 [heap] 07-13 14:42:15.134 6173 6173 I DEBUG : 07-13 14:42:15.134 6173 6173 I DEBUG : stack: 07-13 14:42:15.134 6173 6173 I DEBUG : 41623890 00000000 07-13 14:42:15.134 6173 6173 I DEBUG : 41623894 00000000 07-13 14:42:15.134 6173 6173 I DEBUG : 41623898 00000000 07-13 14:42:15.134 6173 6173 I DEBUG : 4162389c 00000000 07-13 14:42:15.134 6173 6173 I DEBUG : 416238a0 400ec588 07-13 14:42:15.134 6173 6173 I DEBUG : 416238a4 ffffffff 07-13 14:42:15.134 6173 6173 I DEBUG : 416238a8 410d3eb8 /system/lib/libasound.so 07-13 14:42:15.134 6173 6173 I DEBUG : 416238ac 416238d4 07-13 14:42:15.134 6173 6173 I DEBUG : 416238b0 003e1dd0 [heap] 07-13 14:42:15.134 6173 6173 I DEBUG : 416238b4 416238d0 07-13 14:42:15.134 6173 6173 I DEBUG : 416238b8 416238cc 07-13 14:42:15.134 6173 6173 I DEBUG : 416238bc 4162392c 07-13 14:42:15.134 6173 6173 I DEBUG : 416238c0 416238fc 07-13 14:42:15.134 6173 6173 I DEBUG : 416238c4 055f67d1 07-13 14:42:15.134 6173 6173 I DEBUG : 416238c8 df0027ad 07-13 14:42:15.134 6173 6173 I DEBUG : 416238cc 00000000 07-13 14:42:15.134 6173 6173 I DEBUG : #00 416238d0 b0009f58 /system/bin/linker 07-13 14:42:15.134 6173 6173 I DEBUG : 416238d4 41623928 07-13 14:42:15.134 6173 6173 I DEBUG : 416238d8 4162390c 07-13 14:42:15.134 6173 6173 I DEBUG : 416238dc 055f67d1 07-13 14:42:15.142 6173 6173 I DEBUG : 416238e0 41623928 07-13 14:42:15.142 6173 6173 I DEBUG : 416238e4 00476150 [heap] 07-13 14:42:15.142 6173 6173 I DEBUG : 416238e8 003e1dd0 [heap] 07-13 14:42:15.142 6173 6173 I DEBUG : 416238ec b0003d65 /system/bin/linker 07-13 14:42:15.142 6173 6173 I DEBUG : 416238f0 ffffffff 07-13 14:42:15.142 6173 6173 I DEBUG : 416238f4 4108ee80 /system/lib/libasound.so 07-13 14:42:15.142 6173 6173 I DEBUG : 416238f8 41623928 07-13 14:42:15.142 6173 6173 I DEBUG : 416238fc ffffffff 07-13 14:42:15.142 6173 6173 I DEBUG : 41623900 41623928 07-13 14:42:15.142 6173 6173 I DEBUG : 41623904 b0005817 /system/bin/linker 07-13 14:42:15.142 6173 6173 I DEBUG : 41623908 ffffffff 07-13 14:42:15.142 6173 6173 I DEBUG : 4162390c 41623928 07-13 14:42:15.142 6173 6173 I DEBUG : 41623910 410e19e4 /system/lib/libasound.so 07-13 14:42:15.142 6173 6173 I DEBUG : 41623914 ffffff84 Thanks, Kuldeep

1 0

[alsa-devel] Tascam US1800.
by allan walker 13 Jul '12

13 Jul '12

Hello, please forgive me if i am contravening mailing list protocol, this is my first time posting outside of forums. my problem is discribed here http://www.linuxmusicians.com/viewtopic.php?f=6&t=8792 basically i would like to contribute my time to helping to get this device working under alsa. my info from the scrip is here http://www.alsa-project.org/db/?f=d444ad0ee58b3994ee1c91e5ed4415bb2416f486p… indorm me if there is anything at all that i can do to aid you in gaining driver support for this hardware. it has the most inputs for the price, and im sure its something that many people would be happy to see supported. best regards, Allan Walker

2 1

Re: [alsa-devel] [PATCH 1/2] sound:soc:soc-dmap: Fix compilation warning.
by Mark Brown 13 Jul '12

13 Jul '12

On Thu, Jul 12, 2012 at 11:00:16PM +0200, Marek Belisko wrote: > Fix following: > sound/soc/soc-dapm.c: In function ‘dapm_clock_event’: > sound/soc/soc-dapm.c:1021:1: warning: control reaches end of non-void function [-Wreturn-type] Applied,:but:please:do:make:an:effort:to:use:subject:lines:appropriate:to the:subsystem. As well as using a totally different style there's a spelling issue, I very nearly deleted the patch unread because of these things.

1 0

Re: [alsa-devel] [PATCH 1/2] ASoC: Convert S3C24XX I2S driver to gpiolib API
by Mark Brown 13 Jul '12

13 Jul '12

On Thu, Jul 12, 2012 at 08:09:09PM +0200, Sylwester Nawrocki wrote: > Thanks, would that be Takashi Iwai and Jaroslav Kysela then ? I just > want to get it right this time... > Or perhaps could you take care of them ? As with everything else look in MAINTAINERS, so me for ASoC.

1 0

Re: [alsa-devel] [PATCH V2] Add missing modules aliases to get sound working on omap devices
by Mark Brown 13 Jul '12

13 Jul '12

On Thu, Jul 12, 2012 at 03:08:16PM +0200, Guillaume Gardet wrote: > This patch add missing modules aliases to get sound working on omap devices. > > Tested on Beagleboard xM rev. B. > > This patch is against 3.5-rc6 vanilla. > > Signed-off-by: Guillaume GARDET <guillaume.gardet(a)free.fr> > Signed-off-by: Mans Rullgard <mans.rullgard(a)linaro.org> > > > From 18b1ba8becc3dd256bdaad2d825f46b551debda3 Mon Sep 17 00:00:00 2001 > From: Guillaume GARDET <guillaume.gardet(a)oliseo.fr> > Date: Tue, 10 Jul 2012 13:47:16 +0200 Please resend in the format from SubmittingPatches rather than as an attachment.

1 0

[alsa-devel] Plug plugin not working on Raspberry Pi.
by Krzysztof Adamski 13 Jul '12

13 Jul '12

Hi, for a couple of days I'm trying to figure out problems with ALSA on Raspberry Pi. It turned out that for some reason, plug plugin is not working as expected. Id does work correctly when I'm using PCM configuration that can be handled directly by HW but any other configuration (like unsuported format, more channes etc) fails with "Invalid argument" on snd_pcm_hw_params. It's not my code that fails because of this by it's both mplayer and VLC. Both snd_pcm_hw_params_set_format and snd_pcm_hw_params_test_format (and other similar functions) does not return error. Only when snd_pcm_hw_params is run, there is one. I'm using Debian Wheezy, with ALSA 1.0.25 with all default ALSA configuration. ALSA drivers for RPi are not in mainline kernel but they can be viewed here: https://github.com/raspberrypi/linux/tree/rpi-patches/sound/arm (it's bcm2835). Card capabilities seems to be described correctly: static struct snd_pcm_hardware snd_bcm2835_playback_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000, .rate_min = 8000, .rate_max = 48000, .channels_min = 1, .channels_max = 2, .buffer_bytes_max = (4 * 8 - 1) * 1024, /* Needs to be less than audioplay buffer size */ .period_bytes_min = 1 * 1024, .period_bytes_max = (4 * 8 - 1) * 1024, .periods_min = 1, .periods_max = 4 * 8 - 1, }; So, I would like someone to help me investigate if it's a userspace or driver BUG and how to fix this. I will be happy to provide any information that is needed for debug, test patches, recompile with some debugin enabled etc.

1 0

[alsa-devel] [PATCH 1/2] ALSA: snd-usb: tighten EP_FLAG_RUNNING checks
by Daniel Mack 13 Jul '12

13 Jul '12

In endpoint.c, bail out earlier in case the stream is stopped. Also clear the flag from deactivate_urbs() in case the chip's shutdown flag is set. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/endpoint.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e690690..a75bdf4 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -350,7 +350,8 @@ static void snd_complete_urb(struct urb *urb) urb->status == -ENODEV || /* device removed */ urb->status == -ECONNRESET || /* unlinked */ urb->status == -ESHUTDOWN || /* device disabled */ - ep->chip->shutdown)) /* device disconnected */ + ep->chip->shutdown) || /* device disconnected */ + !test_bit(EP_FLAG_RUNNING, &ep->flags)) goto exit_clear; if (usb_pipeout(ep->pipe)) { @@ -522,13 +523,13 @@ static int deactivate_urbs(struct snd_usb_endpoint *ep, int force, int can_sleep unsigned int i; int async; + clear_bit(EP_FLAG_RUNNING, &ep->flags); + if (!force && ep->chip->shutdown) /* to be sure... */ return -EBADFD; async = !can_sleep && ep->chip->async_unlink; - clear_bit(EP_FLAG_RUNNING, &ep->flags); - INIT_LIST_HEAD(&ep->ready_playback_urbs); ep->next_packet_read_pos = 0; ep->next_packet_write_pos = 0; -- 1.7.10.4

3 8

Re: [alsa-devel] [PATCH 1/2] ALSA: snd-usb: tighten EP_FLAG_RUNNING checks
by Daniel Mack 12 Jul '12

12 Jul '12

On 12.07.2012 21:19, Joseph Salisbury wrote: > On 07/12/2012 11:27 AM, Daniel Mack wrote: >> On 12.07.2012 17:20, Clemens Ladisch wrote: >>> Daniel Mack wrote: >>>> On 12.07.2012 16:29, Clemens Ladisch wrote: >>>>> Daniel Mack wrote: >>>>>> In endpoint.c, bail out earlier in case the stream is stopped. >>>>>> ... >>>>>> @@ -350,7 +350,8 @@ static void snd_complete_urb(struct urb *urb) >>>>>> urb->status == -ENODEV || /* device removed */ >>>>>> urb->status == -ECONNRESET || /* unlinked */ >>>>>> urb->status == -ESHUTDOWN || /* device disabled */ >>>>>> - ep->chip->shutdown)) /* device disconnected */ >>>>>> + ep->chip->shutdown) || /* device disconnected */ >>>>>> + !test_bit(EP_FLAG_RUNNING, &ep->flags)) >>>>>> goto exit_clear; >>>>> >>>>> Is this really needed? >>>>> The URBs will be unlinked at the same time. >>>> >>>> This just brings the code in sync with what we had before. If URBs are >>>> just unlinked but not killed, they will return with data payload, and in >>>> case of implicit feedback streams, the retire code could issue new >>>> output packets. >>> >>> And is that bad? >>> >>> The EP_FLAG_RUNNING bit is cleared immediately before the URBs are >>> unlinked, and some URB could have been completed regularly immediately >>> before that. So if there is any case where the additional test_bit() >>> call prevents submitting a new playback packet, there is another similar >>> case where you've just got such a playback packet anyway. >>> >>> In other words: if that test_bit() call were necessary to protect >>> against something bad, it wouldn't be sufficient. >>> >>> The driver ensures that (implicit feedback) playback URBs are unlinked >>> only after the corresponding capture URBs have been completely killed, >>> doesn't it? >> >> You're right. Especially because sending out packets in implicit >> feedback mode happens from queue_pending_output_urbs() which has an >> extra chip. >> >> So we can drop the whole patch then. Thanks for the review! >> >> >> Daniel > > Hi Daniel, > > I tested without your first patch. I'm not sure why, but dropping the > first patch(ALSA: snd-usb: tighten EP_FLAG_RUNNING checks) causes a long > delay, about 30 seconds, during bootup of an ubuntu system. > > During the delay, I am able to enter my password, but the system seems > to be doing something in the background, slowing the login process. > > I confirmed that adding the patch back causes the delay to go away. That really sounds odd and totally unrelated. Is there any message in dmesg that looks suspicious? Otherwise, I would ask you to test again. Could be it was caused by something that runs on every n'th boot or so. Daniel

1 0

Re: [alsa-devel] [PATCH 1/2] ASoC: Convert S3C24XX I2S driver to gpiolib API
by Mark Brown 12 Jul '12

12 Jul '12

On Thu, Jul 12, 2012 at 03:56:36PM +0200, Sylwester Nawrocki wrote: > These are the only two patches (out of 15) that are now needed to remove > some obsolete s3c2410_gpio_* API. 1/2 was tested on Micro2440 board > and 2/2 is an identical change which wasn't tested on a real H/W since > I don't have any s3c2412 based board. I'm a bit lost through which tree > should these be merged. Could someone please pick these patches up ? If you send them to the relevant maintainers that'd probably help...

1 0

Re: [alsa-devel] Regression in sound/usb/
by Daniel Mack 12 Jul '12

12 Jul '12

On 12.07.2012 11:18, Philipp Dreimann wrote: > On 12 July 2012 01:10, Daniel Mack <zonque(a)gmail.com> wrote: >> On 11.07.2012 09:30, Philipp Dreimann wrote: >> >>> The camera is now working for a short time: >>> [ 56.421399] usb 2-1.2: new high-speed USB device number 4 using ehci_hcd >>> [ 57.224234] uvcvideo: Found UVC 1.00 device <unnamed> (046d:0821) >>> [ 57.237221] input: UVC Camera (046d:0821) as >>> /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.2/2-1.2:1.2/input/input15 >>> Camera and microphone are working. Some time later we're back to the old issue: >> >> So I found the reason for this breakage now. It's about orphaned packets >> that are sent over the wire after alt setting 0 has been selected. The >> chipset doesn't like these at and bails out with protocol errors. Fixing >> this involves some rework of the logic. The patch is unfortunately a >> little big, but I see no option than getting it merged for 3.6. > > The issue seems to be fixed for me. Let's hope that it'll make it in > the next v3.5-rc. Thanks for testing. I just sent two patches. Combined, they are identical to what you tested. Anyone else wants to add a Tested-by? Thanks, Daniel

1 0

[PATCH v2] ALSA: snd-usb: move calls to usb_set_interface
by Daniel Mack 12 Jul '12

12 Jul '12

The rework of the snd-usb endpoint logic moved the calls to snd_usb_set_interface() into the snd_usb_endpoint implemenation. This changed the order in which these calls are issued to the device, and thereby caused regressions for some webcams. Fix this by moving the calls back to pcm.c for now to make it work again and use snd_usb_endpoint_activate() to really tear down all remaining URBs in the flight, consequently fixing another regression caused by USB packets on the wire after altsetting 0 has been selected. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: Philipp Dreimann <philipp(a)dreimann.net> Reported-by: Joseph Salisbury <joseph.salisbury(a)canonical.com> --- sound/usb/endpoint.c | 73 +++++--------------------------------------------- sound/usb/pcm.c | 57 ++++++++++++++++++++++++--------------- 2 files changed, 41 insertions(+), 89 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a75bdf4..4bc5778 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -415,7 +415,7 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, { struct list_head *p; struct snd_usb_endpoint *ep; - int ret, is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK; + int is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK; mutex_lock(&chip->mutex); @@ -435,16 +435,6 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, type == SND_USB_ENDPOINT_TYPE_DATA ? "data" : "sync", ep_num); - /* select the alt setting once so the endpoints become valid */ - ret = usb_set_interface(chip->dev, alts->desc.bInterfaceNumber, - alts->desc.bAlternateSetting); - if (ret < 0) { - snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n", - __func__, ret); - ep = NULL; - goto __exit_unlock; - } - ep = kzalloc(sizeof(*ep), GFP_KERNEL); if (!ep) goto __exit_unlock; @@ -832,9 +822,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (++ep->use_count != 1) return 0; - if (snd_BUG_ON(!test_bit(EP_FLAG_ACTIVATED, &ep->flags))) - return -EINVAL; - /* just to be sure */ deactivate_urbs(ep, 0, 1); wait_clear_urbs(ep); @@ -912,9 +899,6 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, if (snd_BUG_ON(ep->use_count == 0)) return; - if (snd_BUG_ON(!test_bit(EP_FLAG_ACTIVATED, &ep->flags))) - return; - if (--ep->use_count == 0) { deactivate_urbs(ep, force, can_sleep); ep->data_subs = NULL; @@ -928,42 +912,6 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, } /** - * snd_usb_endpoint_activate: activate an snd_usb_endpoint - * - * @ep: the endpoint to activate - * - * If the endpoint is not currently in use, this functions will select the - * correct alternate interface setting for the interface of this endpoint. - * - * In case of any active users, this functions does nothing. - * - * Returns an error if usb_set_interface() failed, 0 in all other - * cases. - */ -int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep) -{ - if (ep->use_count != 0) - return 0; - - if (!ep->chip->shutdown && - !test_and_set_bit(EP_FLAG_ACTIVATED, &ep->flags)) { - int ret; - - ret = usb_set_interface(ep->chip->dev, ep->iface, ep->alt_idx); - if (ret < 0) { - snd_printk(KERN_ERR "%s() usb_set_interface() failed, ret = %d\n", - __func__, ret); - clear_bit(EP_FLAG_ACTIVATED, &ep->flags); - return ret; - } - - return 0; - } - - return -EBUSY; -} - -/** * snd_usb_endpoint_deactivate: deactivate an snd_usb_endpoint * * @ep: the endpoint to deactivate @@ -981,24 +929,15 @@ int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) if (!ep) return -EINVAL; + deactivate_urbs(ep, 1, 1); + wait_clear_urbs(ep); + if (ep->use_count != 0) return 0; - if (!ep->chip->shutdown && - test_and_clear_bit(EP_FLAG_ACTIVATED, &ep->flags)) { - int ret; - - ret = usb_set_interface(ep->chip->dev, ep->iface, 0); - if (ret < 0) { - snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n", - __func__, ret); - return ret; - } + clear_bit(EP_FLAG_ACTIVATED, &ep->flags); - return 0; - } - - return -EBUSY; + return 0; } /** diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f0ede13..a1298f3 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -261,19 +261,6 @@ static void stop_endpoints(struct snd_usb_substream *subs, force, can_sleep, wait); } -static int activate_endpoints(struct snd_usb_substream *subs) -{ - if (subs->sync_endpoint) { - int ret; - - ret = snd_usb_endpoint_activate(subs->sync_endpoint); - if (ret < 0) - return ret; - } - - return snd_usb_endpoint_activate(subs->data_endpoint); -} - static int deactivate_endpoints(struct snd_usb_substream *subs) { int reta, retb; @@ -314,6 +301,33 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) if (fmt == subs->cur_audiofmt) return 0; + /* close the old interface */ + if (subs->interface >= 0 && subs->interface != fmt->iface) { + err = usb_set_interface(subs->dev, subs->interface, 0); + if (err < 0) { + snd_printk(KERN_ERR "%d:%d:%d: return to setting 0 failed (%d)\n", + dev->devnum, fmt->iface, fmt->altsetting, err); + return -EIO; + } + subs->interface = -1; + subs->altset_idx = 0; + } + + /* set interface */ + if (subs->interface != fmt->iface || + subs->altset_idx != fmt->altset_idx) { + err = usb_set_interface(dev, fmt->iface, fmt->altsetting); + if (err < 0) { + snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed (%d)\n", + dev->devnum, fmt->iface, fmt->altsetting, err); + return -EIO; + } + snd_printdd(KERN_INFO "setting usb interface %d:%d\n", + fmt->iface, fmt->altsetting); + subs->interface = fmt->iface; + subs->altset_idx = fmt->altset_idx; + } + subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, alts, fmt->endpoint, subs->direction, SND_USB_ENDPOINT_TYPE_DATA); @@ -460,12 +474,6 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, mutex_lock(&subs->stream->chip->shutdown_mutex); /* format changed */ stop_endpoints(subs, 0, 0, 0); - deactivate_endpoints(subs); - - ret = activate_endpoints(subs); - if (ret < 0) - goto unlock; - ret = snd_usb_endpoint_set_params(subs->data_endpoint, hw_params, fmt, subs->sync_endpoint); if (ret < 0) @@ -500,6 +508,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) subs->period_bytes = 0; mutex_lock(&subs->stream->chip->shutdown_mutex); stop_endpoints(subs, 0, 1, 1); + deactivate_endpoints(subs); mutex_unlock(&subs->stream->chip->shutdown_mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); } @@ -938,16 +947,20 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) { - int ret; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; stop_endpoints(subs, 0, 0, 0); - ret = deactivate_endpoints(subs); + + if (!as->chip->shutdown && subs->interface >= 0) { + usb_set_interface(subs->dev, subs->interface, 0); + subs->interface = -1; + } + subs->pcm_substream = NULL; snd_usb_autosuspend(subs->stream->chip); - return ret; + return 0; } /* Since a URB can handle only a single linear buffer, we must use double -- 1.7.10.4 --------------020204080001050308060908--

1 0

Re: [alsa-devel] Regression in sound/usb/
by Daniel Mack 12 Jul '12

12 Jul '12

On 11.07.2012 09:30, Philipp Dreimann wrote: > The camera is now working for a short time: > [ 56.421399] usb 2-1.2: new high-speed USB device number 4 using ehci_hcd > [ 57.224234] uvcvideo: Found UVC 1.00 device <unnamed> (046d:0821) > [ 57.237221] input: UVC Camera (046d:0821) as > /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.2/2-1.2:1.2/input/input15 > Camera and microphone are working. Some time later we're back to the old issue: So I found the reason for this breakage now. It's about orphaned packets that are sent over the wire after alt setting 0 has been selected. The chipset doesn't like these at and bails out with protocol errors. Fixing this involves some rework of the logic. The patch is unfortunately a little big, but I see no option than getting it merged for 3.6. Everyone, please test - and sorry for the trouble caused. Thanks, Daniel

1 1

[alsa-devel] [PATCH] ASoC: wm8962: Redo early init of the part on resume
by Mark Brown 11 Jul '12

11 Jul '12

Ensure robust startup of the part by going through the reset procedure prior to resyncing the full register cache, avoiding potential intermittent faults in some designs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5781581..6999f74 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3727,6 +3727,9 @@ static int wm8962_runtime_resume(struct device *dev) } regcache_cache_only(wm8962->regmap, false); + + wm8962_reset(wm8962); + regcache_sync(wm8962->regmap); regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, -- 1.7.10

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Re: [alsa-devel] [PATCH] ASoC: Free memory in the error paths of soc_of_parse_audio_routing()
by Mark Brown 11 Jul '12

11 Jul '12

On Wed, Jul 11, 2012 at 05:36:34PM +0200, Matthias Kaehlcke wrote: > Release the memory of the routing table before leaving the function upon errors > in the device tree Applied, thanks.

1 0

Re: [alsa-devel] Regression in sound/usb/
by Daniel Mack 11 Jul '12

11 Jul '12

On 11.07.2012 17:26, Joseph Salisbury wrote: > On 07/11/2012 04:31 AM, Daniel Mack wrote: >> On 11.07.2012 09:30, Philipp Dreimann wrote: >>> On 10 July 2012 23:14, Daniel Mack <zonque(a)gmail.com> wrote: >>>> On 05.07.2012 17:36, Takashi Iwai wrote: >>>>> At Thu, 5 Jul 2012 17:32:30 +0200, >>>>> Philipp Dreimann wrote: >>>>>> >>>>>> On 5 July 2012 16:41, Takashi Iwai <tiwai(a)suse.de> wrote: >>>>>>>> Try the patch below. If it still doesn't work, give the kernel >>>>>>>> message again and lsusb -v output, too. >>>>>>> >>>>>>> Did it make any difference? >>>>>> >>>>>> No, it unfortunately did not. >>>>>> >>>>>> lsusb is attached and the parts of the kernel log are in the mail: >>>>>> >>>>>> [ 41.243211] usb 2-1.2: new high-speed USB device number 3 using ehci_hcd >>>>>> [ 41.605666] uvcvideo: Found UVC 1.00 device <unnamed> (046d:0821) >>>>>> [ 42.002275] input: UVC Camera (046d:0821) as >>>>>> /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.2/2-1.2:1.2/input/input12 >>>>>> [ 42.085880] usbcore: registered new interface driver snd-usb-audio >>>>>> >>>>>> [ 52.790139] uvcvideo: Failed to query (GET_DEF) UVC control 2 on >>>>>> unit 2: -71 (exp. 2). >>>>>> [ 52.794363] uvcvideo: Failed to query (GET_DEF) UVC control 2 on >>>>>> unit 2: -71 (exp. 2). >>>>>> [ 52.798375] uvcvideo: Failed to query (GET_DEF) UVC control 3 on >>>>>> unit 2: -71 (exp. 2). >>>>>> [ 52.802499] uvcvideo: Failed to query (GET_DEF) UVC control 7 on >>>>>> unit 2: -71 (exp. 2). >>>>>> [ 52.806617] uvcvideo: Failed to query (GET_DEF) UVC control 11 on >>>>>> unit 2: -71 (exp. 1). >>>>>> [ 52.810734] uvcvideo: Failed to query (GET_DEF) UVC control 4 on >>>>>> unit 2: -71 (exp. 2). >>>>>> [ 52.814855] uvcvideo: Failed to query (GET_DEF) UVC control 5 on >>>>>> unit 2: -71 (exp. 1). >>>>>> [ 52.818976] uvcvideo: Failed to query (GET_CUR) UVC control 11 on >>>>>> unit 2: -71 (exp. 1). >>>>>> [ 52.823096] uvcvideo: Failed to query (GET_DEF) UVC control 8 on >>>>>> unit 2: -71 (exp. 2). >>>>>> [ 52.827215] uvcvideo: Failed to query (GET_DEF) UVC control 1 on >>>>>> unit 2: -71 (exp. 2). >>>>>> [ 52.831331] uvcvideo: Failed to set UVC probe control : -71 (exp. 26). >>>>>> [ 52.835459] uvcvideo: Failed to set UVC probe control : -71 (exp. 26). >>>>>> The last line repeats a few times, then the Failed to query block >>>>>> repeats again, and so on. But only if I try to use the webcam. >>>>>> >>>>>> [ 82.363599] 3:1:3: cannot set freq 32000 to ep 0x86 >>>>>> Is showing up without using the webcam. >>>>> >>>>> Hm, it looks better than before (it was '3:-1:3' or such), but there >>>>> is still something strange there. >>>> >>>> Sorry for the long delay on this. I finally found some time to look into >>>> this and it seems that I got the calls to usb_set_interface wrong. >>>> Attached is a patch to fix this - could everybody who is affect please >>>> give that a quick try? More explanation is in the commit log. >>> >>> The camera is now working for a short time: >>> [ 56.421399] usb 2-1.2: new high-speed USB device number 4 using ehci_hcd >>> [ 57.224234] uvcvideo: Found UVC 1.00 device <unnamed> (046d:0821) >>> [ 57.237221] input: UVC Camera (046d:0821) as >>> /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.2/2-1.2:1.2/input/input15 >>> Camera and microphone are working. Some time later we're back to the old issue: >> >> Ok, I see. Drop this, I'm working on a new one. I will find some time >> later today to look into this. Thanks for your patience! >> >> >> Daniel > > I just sent an email to Philipp that I've been unable to reproduce the > failure on my hardware. However, a user testing your patch has also run > into the same failure as Philipp: > > https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1018020/comments/38 > > That users webcam will work up to 15 minutes, than fail. I'll pick up an USB hw analyzer later today and then try with a Logitech camera. Will keep you posted.

1 0

[alsa-devel] [patch] ASoC: STA529: fix an error message
by Dan Carpenter 11 Jul '12

11 Jul '12

GCC complains that "ret" is uninitialized here. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index a9f34c7..0c225cd 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -397,8 +397,9 @@ static __devinit int sta529_i2c_probe(struct i2c_client *i2c, sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap); if (IS_ERR(sta529->regmap)) { + ret = PTR_ERR(sta529->regmap); dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); - return PTR_ERR(sta529->regmap); + return ret; } i2c_set_clientdata(i2c, sta529);

3 2

Re: [alsa-devel] Regression in sound/usb/
by Daniel Mack 11 Jul '12

11 Jul '12

On 11.07.2012 09:30, Philipp Dreimann wrote: > On 10 July 2012 23:14, Daniel Mack <zonque(a)gmail.com> wrote: >> On 05.07.2012 17:36, Takashi Iwai wrote: >>> At Thu, 5 Jul 2012 17:32:30 +0200, >>> Philipp Dreimann wrote: >>>> >>>> On 5 July 2012 16:41, Takashi Iwai <tiwai(a)suse.de> wrote: >>>>>> Try the patch below. If it still doesn't work, give the kernel >>>>>> message again and lsusb -v output, too. >>>>> >>>>> Did it make any difference? >>>> >>>> No, it unfortunately did not. >>>> >>>> lsusb is attached and the parts of the kernel log are in the mail: >>>> >>>> [ 41.243211] usb 2-1.2: new high-speed USB device number 3 using ehci_hcd >>>> [ 41.605666] uvcvideo: Found UVC 1.00 device <unnamed> (046d:0821) >>>> [ 42.002275] input: UVC Camera (046d:0821) as >>>> /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.2/2-1.2:1.2/input/input12 >>>> [ 42.085880] usbcore: registered new interface driver snd-usb-audio >>>> >>>> [ 52.790139] uvcvideo: Failed to query (GET_DEF) UVC control 2 on >>>> unit 2: -71 (exp. 2). >>>> [ 52.794363] uvcvideo: Failed to query (GET_DEF) UVC control 2 on >>>> unit 2: -71 (exp. 2). >>>> [ 52.798375] uvcvideo: Failed to query (GET_DEF) UVC control 3 on >>>> unit 2: -71 (exp. 2). >>>> [ 52.802499] uvcvideo: Failed to query (GET_DEF) UVC control 7 on >>>> unit 2: -71 (exp. 2). >>>> [ 52.806617] uvcvideo: Failed to query (GET_DEF) UVC control 11 on >>>> unit 2: -71 (exp. 1). >>>> [ 52.810734] uvcvideo: Failed to query (GET_DEF) UVC control 4 on >>>> unit 2: -71 (exp. 2). >>>> [ 52.814855] uvcvideo: Failed to query (GET_DEF) UVC control 5 on >>>> unit 2: -71 (exp. 1). >>>> [ 52.818976] uvcvideo: Failed to query (GET_CUR) UVC control 11 on >>>> unit 2: -71 (exp. 1). >>>> [ 52.823096] uvcvideo: Failed to query (GET_DEF) UVC control 8 on >>>> unit 2: -71 (exp. 2). >>>> [ 52.827215] uvcvideo: Failed to query (GET_DEF) UVC control 1 on >>>> unit 2: -71 (exp. 2). >>>> [ 52.831331] uvcvideo: Failed to set UVC probe control : -71 (exp. 26). >>>> [ 52.835459] uvcvideo: Failed to set UVC probe control : -71 (exp. 26). >>>> The last line repeats a few times, then the Failed to query block >>>> repeats again, and so on. But only if I try to use the webcam. >>>> >>>> [ 82.363599] 3:1:3: cannot set freq 32000 to ep 0x86 >>>> Is showing up without using the webcam. >>> >>> Hm, it looks better than before (it was '3:-1:3' or such), but there >>> is still something strange there. >> >> Sorry for the long delay on this. I finally found some time to look into >> this and it seems that I got the calls to usb_set_interface wrong. >> Attached is a patch to fix this - could everybody who is affect please >> give that a quick try? More explanation is in the commit log. > > The camera is now working for a short time: > [ 56.421399] usb 2-1.2: new high-speed USB device number 4 using ehci_hcd > [ 57.224234] uvcvideo: Found UVC 1.00 device <unnamed> (046d:0821) > [ 57.237221] input: UVC Camera (046d:0821) as > /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.2/2-1.2:1.2/input/input15 > Camera and microphone are working. Some time later we're back to the old issue: Ok, I see. Drop this, I'm working on a new one. I will find some time later today to look into this. Thanks for your patience! Daniel

1 0

[alsa-devel] Full duplex ALSA-plugin
by Christian Gruber 11 Jul '12

11 Jul '12

I was looking for information on writing an echo cancellation ALSA-plugin. So I studied the ALSA-plugin docs and also found several interesting threads in this mailing list. But what I found out, was quiet disillusioning. If I understood the ALSA-plugin concept right, there is no possibility to write a plugin working on two streams, a playback and a record stream, simultaneously. One can only write a filter plugin, which works either on a playback stream or a record stream but not both. And there is no possibility to open a full duplex stream (consisting of playback and record) with snd_pcm_open(). Is there a reason, why full duplex streams are not implemented yet? Or is it planned in the future to implement this feature? Regards, Christian

2 2

Re: [alsa-devel] Regression in sound/usb/
by Takashi Iwai 10 Jul '12

10 Jul '12

At Thu, 5 Jul 2012 17:32:30 +0200, Philipp Dreimann wrote: > > On 5 July 2012 16:41, Takashi Iwai <tiwai(a)suse.de> wrote: > >> Try the patch below. If it still doesn't work, give the kernel > >> message again and lsusb -v output, too. > > > > Did it make any difference? > > No, it unfortunately did not. > > lsusb is attached and the parts of the kernel log are in the mail: > > [ 41.243211] usb 2-1.2: new high-speed USB device number 3 using ehci_hcd > [ 41.605666] uvcvideo: Found UVC 1.00 device <unnamed> (046d:0821) > [ 42.002275] input: UVC Camera (046d:0821) as > /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.2/2-1.2:1.2/input/input12 > [ 42.085880] usbcore: registered new interface driver snd-usb-audio > > [ 52.790139] uvcvideo: Failed to query (GET_DEF) UVC control 2 on > unit 2: -71 (exp. 2). > [ 52.794363] uvcvideo: Failed to query (GET_DEF) UVC control 2 on > unit 2: -71 (exp. 2). > [ 52.798375] uvcvideo: Failed to query (GET_DEF) UVC control 3 on > unit 2: -71 (exp. 2). > [ 52.802499] uvcvideo: Failed to query (GET_DEF) UVC control 7 on > unit 2: -71 (exp. 2). > [ 52.806617] uvcvideo: Failed to query (GET_DEF) UVC control 11 on > unit 2: -71 (exp. 1). > [ 52.810734] uvcvideo: Failed to query (GET_DEF) UVC control 4 on > unit 2: -71 (exp. 2). > [ 52.814855] uvcvideo: Failed to query (GET_DEF) UVC control 5 on > unit 2: -71 (exp. 1). > [ 52.818976] uvcvideo: Failed to query (GET_CUR) UVC control 11 on > unit 2: -71 (exp. 1). > [ 52.823096] uvcvideo: Failed to query (GET_DEF) UVC control 8 on > unit 2: -71 (exp. 2). > [ 52.827215] uvcvideo: Failed to query (GET_DEF) UVC control 1 on > unit 2: -71 (exp. 2). > [ 52.831331] uvcvideo: Failed to set UVC probe control : -71 (exp. 26). > [ 52.835459] uvcvideo: Failed to set UVC probe control : -71 (exp. 26). > The last line repeats a few times, then the Failed to query block > repeats again, and so on. But only if I try to use the webcam. > > [ 82.363599] 3:1:3: cannot set freq 32000 to ep 0x86 > Is showing up without using the webcam. Hm, it looks better than before (it was '3:-1:3' or such), but there is still something strange there. thanks, Takashi

2 1

[PATCH] ALSA: snd-usb: fix calls to usb_set_interface
by Daniel Mack 10 Jul '12

10 Jul '12

Don't call usb_set_interface() on snd_usb_endpoint creation and also drop it from snd_usb_endpoint_deactivate(). Rather unselect the the active interface before choosing a new alt setting upon the next stream start. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/endpoint.c | 45 ++++++++++++++------------------------------- 1 file changed, 14 insertions(+), 31 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e690690..2226769 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -414,7 +414,7 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, { struct list_head *p; struct snd_usb_endpoint *ep; - int ret, is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK; + int is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK; mutex_lock(&chip->mutex); @@ -434,16 +434,6 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, type == SND_USB_ENDPOINT_TYPE_DATA ? "data" : "sync", ep_num); - /* select the alt setting once so the endpoints become valid */ - ret = usb_set_interface(chip->dev, alts->desc.bInterfaceNumber, - alts->desc.bAlternateSetting); - if (ret < 0) { - snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n", - __func__, ret); - ep = NULL; - goto __exit_unlock; - } - ep = kzalloc(sizeof(*ep), GFP_KERNEL); if (!ep) goto __exit_unlock; @@ -936,8 +926,8 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, * * In case of any active users, this functions does nothing. * - * Returns an error if usb_set_interface() failed, 0 in all other - * cases. + * Returns an error if any of the usb_set_interface() calls failed, 0 in + * all other cases. */ int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep) { @@ -948,6 +938,14 @@ int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep) !test_and_set_bit(EP_FLAG_ACTIVATED, &ep->flags)) { int ret; + ret = usb_set_interface(ep->chip->dev, ep->iface, 0); + if (ret < 0) { + snd_printk(KERN_ERR "%s() usb_set_interface() failed, ret = %d\n", + __func__, ret); + clear_bit(EP_FLAG_ACTIVATED, &ep->flags); + return ret; + } + ret = usb_set_interface(ep->chip->dev, ep->iface, ep->alt_idx); if (ret < 0) { snd_printk(KERN_ERR "%s() usb_set_interface() failed, ret = %d\n", @@ -967,13 +965,8 @@ int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep) * * @ep: the endpoint to deactivate * - * If the endpoint is not currently in use, this functions will select the - * alternate interface setting 0 for the interface of this endpoint. - * - * In case of any active users, this functions does nothing. - * - * Returns an error if usb_set_interface() failed, 0 in all other - * cases. + * Returns 0 on success, -EINVAL if ep was NULL and -EBUSY if the endpoint + * is still in use. */ int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) { @@ -984,18 +977,8 @@ int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep) return 0; if (!ep->chip->shutdown && - test_and_clear_bit(EP_FLAG_ACTIVATED, &ep->flags)) { - int ret; - - ret = usb_set_interface(ep->chip->dev, ep->iface, 0); - if (ret < 0) { - snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n", - __func__, ret); - return ret; - } - + test_and_clear_bit(EP_FLAG_ACTIVATED, &ep->flags)) return 0; - } return -EBUSY; } -- 1.7.10.4 --------------080102090909090401070304--

1 0

[alsa-devel] usage of snd_seq_set_client_event_filter
by Kaspar Bumke 10 Jul '12

10 Jul '12

Hi all, I was trying to use snd_seq_set_client_event_filter in my application but I can't seem to get it to work correctly. I have attached a small test program which can be compiled with "g++ -o test test.cpp -lasound". My alsa-lib is at 1.0.25. It should simple print the note number for any noteOn events. I was under the impression that the way I have used it only noteOn events would get through but it seems that only the first noteOn event I send it gets through. I have been sending events using VMPK. If I remove snd_seq_set_client_event_filter everything seems to work (except the events aren't filtered of course). What am I doing wrong? All the Best, Kaspar

2 3

[alsa-devel] [PATCH] ASoC: arizona: Support variable FLL VCO multipliers
by Mark Brown 10 Jul '12

10 Jul '12

Some Arizona chips have a higher frequency for the FLL VCO, support this in the common code. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 6 +++--- sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5102.c | 3 +++ 3 files changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index f3680c3..5c9caca 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -734,9 +734,9 @@ static int arizona_calc_fll(struct arizona_fll *fll, /* Apply the division for our remaining calculations */ Fref /= div; - /* Fvco should be 90-100MHz; don't check the upper bound */ + /* Fvco should be over the targt; don't check the upper bound */ div = 1; - while (Fout * div < 90000000) { + while (Fout * div < 90000000 * fll->vco_mult) { div++; if (div > 7) { arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", @@ -744,7 +744,7 @@ static int arizona_calc_fll(struct arizona_fll *fll, return -EINVAL; } } - target = Fout * div; + target = Fout * div / fll->vco_mult; cfg->outdiv = div; arizona_fll_dbg(fll, "Fvco=%dHz\n", target); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b894b64..59caca8 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -141,6 +141,7 @@ struct arizona_fll { struct arizona *arizona; int id; unsigned int base; + unsigned int vco_mult; struct completion lock; struct completion ok; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 7a6a11a..6537f16 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -853,6 +853,9 @@ static int __devinit wm5102_probe(struct platform_device *pdev) wm5102->core.arizona = arizona; + for (i = 0; i < ARRAY_SIZE(wm5102->fll); i++) + wm5102->fll[i].vco_mult = 1; + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, &wm5102->fll[0]); -- 1.7.10

1 0

[alsa-devel] MX28EVK mainline Patchs-3.5-rc5 sgtl5000 record not working
by Andreas Gretler 10 Jul '12

10 Jul '12

Hi, i've tested mainline 3.5-Patchs-3.5-rc5 for MX28EVK and it boot up and work fine. But I'm not able to record something. Playback works. I also test it against the old 2.6.35.3 Kernel and there record works. arecord -l show me this: **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: HiFi Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: HiFi Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 I try record with arecord -d 2 -D hw:0,1 -f dat test.wav , but it crash the Linux. No ErrMsg printed and the board stop working. I have no idea what goes wrong. I put some printk message in mxs-saif.c and the last message i can see before the board crash is in mxs_saif_trigger() is before clk_enable is called. /* * If the saif's master is not himself, we also need to enable * itself clk for its internal basic logic to work. */ if (saif != master_saif) { clk_enable(saif->clk); __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_SET_ADDR); } In alsamixer i can not toggle the capture source from Mic-In to Line-in and also can't change the Headphone from DAC to Line-IN. I'm not sure this is the right place to ask? I will also post it to linux-arm-kernel(a)lists.infradead.org. Can anyone confirm the Problem? Or do/understand I something wrong with record and there is no Problem? Andreas Gretler

1 0

[alsa-devel] [PATCH 1/2] ASoC: tegra+alc5632: Swap MICBIAS1 and MIC1 dapm widgets
by Marc Dietrich 10 Jul '12

10 Jul '12

From: Andrey Danin <danindrey(a)mail.ru> MICBIAS1 is an internal widget. If MICBIAS1 is connected to the board driver instead of the MIC1 ASoC core, it enables widgets even if the capture stream isn't active. Cc: Leon Romanovsky <leon(a)leon.nu> Signed-off-by: Andrey Danin <danindrey(a)mail.ru> Signed-off-by: Marc Dietrich <marvin24(a)gmx.de> --- sound/soc/codecs/alc5632.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 7dd0242..8609a86 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -551,8 +551,10 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"Left LineIn", NULL, "LINEINL"}, {"Right LineIn", NULL, "LINEINR"}, {"Phone", NULL, "PHONEP"}, - {"MIC1 Pre Amp", NULL, "MIC1"}, - {"MIC2 Pre Amp", NULL, "MIC2"}, + {"MICBIAS1", NULL, "MIC1"}, + {"MIC1 Pre Amp", NULL, "MICBIAS1"}, + {"MICBIAS2", NULL, "MIC2"}, + {"MIC2 Pre Amp", NULL, "MICBIAS2"}, {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, -- 1.7.9.5

3 4

[alsa-devel] [PATCH] ASoC: tlv320aic3x: add input clock selection
by Jiri Prchal 10 Jul '12

10 Jul '12

This patch adds input selection of main codec clock - from what pin. Both registers set same value since codec uses clock divider or pll at one time. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x.c | 6 ++++++ sound/soc/codecs/tlv320aic3x.h | 8 ++++++++ 2 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 11e195f..5bc5543 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1030,6 +1030,12 @@ static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + /* set clock on MCLK or GPIO2 or BCLK */ + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK, + clk_id << PLLCLK_IN_SHIFT); + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, CLKDIV_IN_MASK, + clk_id << CLKDIV_IN_SHIFT); + aic3x->sysclk = freq; return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 586c5d5..ed4a788 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -195,6 +195,14 @@ #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 +#define PLLCLK_IN_MASK 0x30 +#define PLLCLK_IN_SHIFT 4 +#define CLKDIV_IN_MASK 0xc0 +#define CLKDIV_IN_SHIFT 6 +/* clock in source */ +#define CLKIN_MCLK 0 +#define CLKIN_GPIO2 1 +#define CLKIN_BCLK 2 /* Software reset register bits */ #define SOFT_RESET 0x80 -- 1.7.7

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic3x: add AGC settings
by Jiri Prchal 10 Jul '12

10 Jul '12

This patch adds AGC target level and times settings for TLV320AIC3x. Enums uses small arrays of two channels left and right since it uses different registers. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x.c | 25 +++++++++++++++++++++++++ 1 files changed, 25 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 0b17e66..11e195f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -256,6 +256,25 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; +static const char *aic3x_agc_level[] = + { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; +static const struct soc_enum aic3x_agc_level_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), +}; + +static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; +static const struct soc_enum aic3x_agc_attack_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), +}; + +static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; +static const struct soc_enum aic3x_agc_decay_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), +}; + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -380,6 +399,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + SOC_ENUM("Left AGC Target level", aic3x_agc_level_enum[0]), + SOC_ENUM("Right AGC Target level", aic3x_agc_level_enum[1]), + SOC_ENUM("Left AGC Attack time", aic3x_agc_attack_enum[0]), + SOC_ENUM("Right AGC Attack time", aic3x_agc_attack_enum[1]), + SOC_ENUM("Left AGC Decay time", aic3x_agc_decay_enum[0]), + SOC_ENUM("Right AGC Decay time", aic3x_agc_decay_enum[1]), /* De-emphasis */ SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), -- 1.7.7

2 1

[alsa-devel] [PATCH] powerpc: select PPC_CLOCK unconditionally for FSL_SOC
by Shawn Guo 10 Jul '12

10 Jul '12

Freescale PowerPC SoCs share a number of IP blocks with Freescale ARM/IMX SoCs, FlexCAN, SSI, FEC, eSDHC, USB, etc. There are some effort consolidating those drivers to make them work for both architectures. One outstanding difference between two architectures is ARM/IMX will turn off module clocks during platform initialization for power saving and expects drivers manage clocks using clk API, while PowerPC mostly does not do that, and thus does not always build in clk API. Listing all those driver Kconfig options in "select PPC_CLOCK if" seems not scalable for long term maintenance, and could easily introduce Kconfig recursive dependency. This patch chooses to select PPC_CLOCK unconditionally for FSL_SOC to always build clk API for PowerPC in. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- arch/powerpc/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/arch/powerpc/Kconfig b/arch/powerpc/Kconfig index feab3ba..63fa7fb 100644 --- a/arch/powerpc/Kconfig +++ b/arch/powerpc/Kconfig @@ -661,7 +661,7 @@ config SBUS config FSL_SOC bool select HAVE_CAN_FLEXCAN if NET && CAN - select PPC_CLOCK if CAN_FLEXCAN + select PPC_CLOCK config FSL_PCI bool -- 1.7.5.4

2 4

[alsa-devel] [PATCH 4/4] ASoC: tlv320aic3x: add AGC settings
by Jiri Prchal 10 Jul '12

10 Jul '12

This patch adds AGC target level and times settings for TLV320AIC3x. Enums uses small arrays of two channels left and right since it uses different registers. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x.c | 25 +++++++++++++++++++++++++ 1 files changed, 25 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 0b17e66..11e195f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -256,6 +256,25 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; +static const char *aic3x_agc_level[] = + { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; +static const struct soc_enum aic3x_agc_level_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), +}; + +static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; +static const struct soc_enum aic3x_agc_attack_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), +}; + +static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; +static const struct soc_enum aic3x_agc_decay_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), +}; + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -380,6 +399,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + SOC_ENUM("Left AGC Target level", aic3x_agc_level_enum[0]), + SOC_ENUM("Right AGC Target level", aic3x_agc_level_enum[1]), + SOC_ENUM("Left AGC Attack time", aic3x_agc_attack_enum[0]), + SOC_ENUM("Right AGC Attack time", aic3x_agc_attack_enum[1]), + SOC_ENUM("Left AGC Decay time", aic3x_agc_decay_enum[0]), + SOC_ENUM("Right AGC Decay time", aic3x_agc_decay_enum[1]), /* De-emphasis */ SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), -- 1.7.7

3 3

[alsa-devel] [PATCH 5/5] ASoC: tlv320aic3x: add input clock selection
by Jiri Prchal 10 Jul '12

10 Jul '12

This patch adds input selection of main codec clock - from what pin. Both registers set same value since codec uses clock divider or pll at one time. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x.c | 6 ++++++ sound/soc/codecs/tlv320aic3x.h | 8 ++++++++ 2 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 11e195f..5bc5543 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1030,6 +1030,12 @@ static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + /* set clock on MCLK or GPIO2 or BCLK */ + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK, + clk_id << PLLCLK_IN_SHIFT); + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, CLKDIV_IN_MASK, + clk_id << CLKDIV_IN_SHIFT); + aic3x->sysclk = freq; return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 586c5d5..ed4a788 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -195,6 +195,14 @@ #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 +#define PLLCLK_IN_MASK 0x30 +#define PLLCLK_IN_SHIFT 4 +#define CLKDIV_IN_MASK 0xc0 +#define CLKDIV_IN_SHIFT 6 +/* clock in source */ +#define CLKIN_MCLK 0 +#define CLKIN_GPIO2 1 +#define CLKIN_BCLK 2 /* Software reset register bits */ #define SOFT_RESET 0x80 -- 1.7.7

1 0

[alsa-devel] [PATCH 3/3] ASoC: tlv320aic3x: add deemphasis switch
by Jiri Prchal 09 Jul '12

09 Jul '12

This patch adds missing deemphasis switch. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index bd03f4d..0b17e66 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -381,6 +381,9 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + /* De-emphasis */ + SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), + /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 119, 0, adc_tlv), -- 1.7.7

2 1

[alsa-devel] [PATCH] ASoC: wm5102: Fix cut'n'paste for digital volume registers
by Mark Brown 09 Jul '12

09 Jul '12

The analogue PGA shifts were used; this makes no practical difference as the values are the same. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 3827fa2..7a6a11a 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -68,13 +68,13 @@ SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_PGA_VOL_SHIFT, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_PGA_VOL_SHIFT, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_PGA_VOL_SHIFT, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), -- 1.7.10

1 0

[alsa-devel] [PATCH] ASoC: arizona: Add IN4 to the mixer tables
by Mark Brown 09 Jul '12

09 Jul '12

Some devices have four input structures rather than three. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 4 ++++ sound/soc/codecs/arizona.h | 2 +- 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 901b53e..0970117 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -79,6 +79,8 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "IN2R", "IN3L", "IN3R", + "IN4L", + "IN4R", "AIF1RX1", "AIF1RX2", "AIF1RX3", @@ -138,6 +140,8 @@ int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { 0x13, 0x14, 0x15, + 0x16, + 0x17, 0x20, /* AIF1RX1 */ 0x21, 0x22, diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 896711e..b894b64 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -59,7 +59,7 @@ struct arizona_priv { struct arizona_dai_priv dai[ARIZONA_MAX_DAI]; }; -#define ARIZONA_NUM_MIXER_INPUTS 55 +#define ARIZONA_NUM_MIXER_INPUTS 57 extern const unsigned int arizona_mixer_tlv[]; extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; -- 1.7.10

1 0

Re: [alsa-devel] [PATCH] ASoC: core: Use dedicated substream ptr for ASoC rtd.
by Liam Girdwood 09 Jul '12

09 Jul '12

On Mon, 2012-07-09 at 10:55 +0100, Liam Girdwood wrote: > The ALSA substream private data should really be used by individual > drivers and > not really by the subsystem to store subsystem data. Create a > dedicated > rtd pointer in the PCM substream for ASoC operations. > > The component driver were converted using this sed :- > > sed -i 's/substream->private_data/substream->rtd/' $(git ls-files > sound/soc) > > Signed-off-by: Liam Girdwood <lrg(a)ti.com> > --- Cancel this one, I've left something from another patch in it. Liam

1 0

[alsa-devel] 3.5-rc6 doesn't run powerpc SoundByLayout
by Elimar Riesebieter 08 Jul '12

08 Jul '12

Hi all, trying 3.5.0-rc6 doesn't give me sound from my snd_aoa_codec_onyx. ii libasound2:powerpc 1.0.25-3 ii libasound2-plugins:powerpc 1.0.25-2 ii alsa-utils 1.0.25-3 alsamixer tells: This sound device does not have any controls. $ aplay -l **** List of PLAYBACK Hardware Devices **** $ aplay -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default:CARD=SoundByLayout Default Audio Device sysdefault:CARD=SoundByLayout Default Audio Device Kernel 3.4.4 runs just fine! Does one had similar experience? Elimar -- Learned men are the cisterns of knowledge, not the fountainheads ;-)

1 0

[alsa-devel] [PATCH] ASoC: omap-mcpdm: Add missing MODULE_ALIAS
by Peter Ujfalusi 06 Jul '12

06 Jul '12

The MODULE_ALIAS() was missing from the driver. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-mcpdm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index b5e23af..f7babb3 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -501,6 +501,7 @@ static struct platform_driver asoc_mcpdm_driver = { module_platform_driver(asoc_mcpdm_driver); +MODULE_ALIAS("platform:omap-mcpdm"); MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez(a)ti.com>"); MODULE_DESCRIPTION("OMAP PDM SoC Interface"); MODULE_LICENSE("GPL"); -- 1.7.8.6

2 1

[alsa-devel] [PATCH] ASoC: dpcm: Allow FE to be opened without valid BE routes.
by Liam Girdwood 06 Jul '12

06 Jul '12

Some userspace will open a PCM device and then configure mixers for routing before triggering. This patch allows userspace to do this sequence. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7063b8f..ef22d0b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1955,10 +1955,8 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) fe->dpcm[stream].runtime = fe_substream->runtime; if (dpcm_path_get(fe, stream, &list) <= 0) { - dev_warn(fe->dev, "asoc: %s no valid %s route\n", + dev_dbg(fe->dev, "asoc: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); - mutex_unlock(&fe->card->mutex); - return -EINVAL; } /* calculate valid and active FE <-> BE dpcms */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: twl6040: fix spelling mistake
by Liam Girdwood 06 Jul '12

06 Jul '12

From: Simon Wilson <simonwilson(a)google.com> Fix spelling mistake in "High-Performance" option of twl6040 power mode. Signed-off-by: Simon Wilson <simonwilson(a)google.com> --- sound/soc/codecs/twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index a36e9fc..0ff1e70 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -553,7 +553,7 @@ static const struct snd_kcontrol_new vibrar_mux_controls = /* Headset power mode */ static const char *twl6040_power_mode_texts[] = { - "Low-Power", "High-Perfomance", + "Low-Power", "High-Performance", }; static const struct soc_enum twl6040_power_mode_enum = -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: dapm: Fix locking during codec shutdown
by Liam Girdwood 06 Jul '12

06 Jul '12

Codec shutdown performs a DAPM power sequence that might cause conflicts and/or race conditions if another stream power event is running simultaneously. Use card's dapm mutex to protect any potential race condition between them. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0f078de..b485960 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3578,10 +3578,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_free); static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) { + struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; LIST_HEAD(down_list); int powerdown = 0; + mutex_lock(&card->dapm_mutex); + list_for_each_entry(w, &dapm->card->widgets, list) { if (w->dapm != dapm) continue; @@ -3604,6 +3607,8 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); } + + mutex_unlock(&card->dapm_mutex); } /* -- 1.7.9.5

2 6

[alsa-devel] [PATCH] ASoC: dapm: Make sure all dapm contexts are updated
by Liam Girdwood 06 Jul '12

06 Jul '12

Make sure we set the bias level for all DAPM contexts when changing level. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 912330b..0f078de 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -291,9 +291,9 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, if (dapm->codec->driver->set_bias_level) ret = dapm->codec->driver->set_bias_level(dapm->codec, level); - else - dapm->bias_level = level; - } + } else + dapm->bias_level = level; + if (ret != 0) goto out; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: pcm: Clean up logging in soc_new_pcm()
by Liam Girdwood 06 Jul '12

06 Jul '12

Use dev_ style logging throughout soc_new_pcm() Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 48fd15b..7063b8f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2003,7 +2003,6 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; @@ -2042,7 +2041,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture, &pcm); } if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + dev_err(rtd->card->dev, "can't create pcm for %s\n", + rtd->dai_link->name); return ret; } dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num, new_name); @@ -2099,14 +2099,14 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); if (ret < 0) { - pr_err("asoc: platform pcm constructor failed\n"); + dev_err(platform->dev, "pcm constructor failed\n"); return ret; } } pcm->private_free = platform->driver->pcm_free; out: - printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, + dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define
by Torben Hohn 06 Jul '12

06 Jul '12

The MX31 Audmux also has 7 Ports. This patch adds the missing define, and makes the debugfs code iterate over that port too. Signed-off-by: Torben Hohn <torbenh(a)linutronix.de> --- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-audmux.h | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 0803274..e7c800e 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -156,7 +156,7 @@ static void __init audmux_debugfs_init(void) return; } - for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) { + for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) { snprintf(buf, sizeof(buf), "ssi%d", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h index 04ebbab..b8ff44b 100644 --- a/sound/soc/fsl/imx-audmux.h +++ b/sound/soc/fsl/imx-audmux.h @@ -14,6 +14,7 @@ #define MX31_AUDMUX_PORT4_SSI_PINS_4 3 #define MX31_AUDMUX_PORT5_SSI_PINS_5 4 #define MX31_AUDMUX_PORT6_SSI_PINS_6 5 +#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 #define MX51_AUDMUX_PORT1_SSI0 0 #define MX51_AUDMUX_PORT2_SSI1 1 -- 1.7.10

1 0

[alsa-devel] [PATCH 1/3] ASoC: dapm: Mark widgets as dirty when a route is added
by Mark Brown 06 Jul '12

06 Jul '12

If we add a new route at runtime then we'll need to recheck the connections to the affected widgets. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ed1860c..beac5e6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2250,6 +2250,10 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, path->connect = 0; return 0; } + + dapm_mark_dirty(wsource, "Route added"); + dapm_mark_dirty(wsink, "Route added"); + return 0; err: -- 1.7.10

2 3

[alsa-devel] then there was "overrun" occue every time once trigger the audio recording
by Zhen Fu 06 Jul '12

06 Jul '12

Hi Clemens, Your answer is very useful to help me. Thank you very much. > case AUDIO_STREAM_CMDID_DATARXTXREQ: > len1 = get_zsp_buf_avail(&prtd->zsp_buf); > len2 = get_fw_avail(p_zsp_req, substream->stream); > len3 = get_buf_avail(&prtd->zsp_buf, substream->stream); > tsize = ((len1 < len2) ? len1 : len2); > if ((tsize == len1) && (len1 == len3) && (tsize >= \ > 2 * prtd->zsp_buf.zsp_period_bytes)) { > tsize -= prtd->zsp_buf.zsp_period_bytes; > } > This doesn't look as if tsize is always the same as the period size. > And what do the get_fw_avail and get_buf_avail functions do? The ZSP have own buffer to save data, so get_fw_avail is acquired zsp buf avial. And ZSP buffer must ensure the zsp_buf have sequenced, so do above process. -- Best Regards, Zhen Fu

1 0

[alsa-devel] then there was "overrun" occue every time once trigger the audio recording
by Zhen Fu 06 Jul '12

06 Jul '12

Hi developer, When I use arecord command, then there was "overrun" occue every time once trigger the audio recording. I have no clue of alsa driver when and how handler xrun. In other words, how to understand function snd_pcm_update_hw_ptr0 in pcm_lib.c. Please enlighten me on this issue. Thanks a lot in advance. -- Best Regards, Zhen Fu

2 3

Re: [alsa-devel] [PATCH] Add missing modules aliases to get sound working on omap devices
by Peter Ujfalusi 06 Jul '12

06 Jul '12

On 07/04/2012 12:16 PM, Guillaume Gardet wrote: > This patch add missing modules aliases to get sound working on omap devices. > > Tested on Beagleboard xM rev. B. > > This patch is against 3.5-rc5 vanilla. > > Signed-off-by: Guillaume GARDET <guillaume.gardet(a)free.fr> > > add-missing-modules-aliases-to-fix-audio-on-omap-devices.patch Can you send the patch generated by git format-patch? Since this is sound related patch, please send it to alsa-devel. Please also CC the maintainers for the patches to be sure they are reaching the right audience. > --- ./sound/soc/omap/omap-pcm.c.orig 2012-07-04 10:30:13.611641186 +0200 > +++ ./sound/soc/omap/omap-pcm.c 2012-07-04 10:30:38.894034111 +0200 > @@ -441,3 +441,4 @@ module_platform_driver(omap_pcm_driver); > MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)bitmer.com>"); > MODULE_DESCRIPTION("OMAP PCM DMA module"); > MODULE_LICENSE("GPL"); > +MODULE_ALIAS("platform:omap-pcm-audio"); > --- ./sound/soc/omap/omap-mcbsp.c.orig 2012-07-04 10:29:14.240066817 +0200 > +++ ./sound/soc/omap/omap-mcbsp.c 2012-07-04 12:03:28.020756978 +0200 > @@ -820,3 +820,4 @@ module_platform_driver(asoc_mcbsp_driver > MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)bitmer.com>"); > MODULE_DESCRIPTION("OMAP I2S SoC Interface"); > MODULE_LICENSE("GPL"); > +MODULE_ALIAS("platform:omap-mcbsp-dai"); This should be: +MODULE_ALIAS("platform:omap-mcbsp"); -- Péter

1 0

[alsa-devel] [PATCH 1/2] ASoC: arizona: Enable ASYNCCLK domain for audio interfaces
by Mark Brown 05 Jul '12

05 Jul '12

If an audio interface is configured to use ASYNCCLK then update the asynchronous sample rate rather than one of our primary sample rates. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 30 ++++++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8e066eb..d0bcca9 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -507,6 +507,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; int base = dai->driver->base; const int *rates; int i; @@ -530,10 +532,6 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* - * We will need to be more flexible than this in future, - * currently we use a single sample rate for the chip. - */ for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++) if (arizona_sr_vals[i] == params_rate(params)) break; @@ -552,8 +550,28 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, wl = snd_pcm_format_width(params_format(params)); frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; - snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, - ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + /* + * We will need to be more flexible than this in future, + * currently we use a single sample rate for SYSCLK. + */ + switch (dai_priv->clk) { + case ARIZONA_CLK_SYSCLK: + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, + ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 0); + break; + case ARIZONA_CLK_ASYNCCLK: + snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, + ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 8); + break; + default: + arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); + return -EINVAL; + } + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE, -- 1.7.10

1 1

Re: [alsa-devel] Regression in sound/usb/
by Daniel Mack 05 Jul '12

05 Jul '12

On 05.07.2012 17:32, Philipp Dreimann wrote: > On 5 July 2012 16:41, Takashi Iwai <tiwai(a)suse.de> wrote: >>> Try the patch below. If it still doesn't work, give the kernel >>> message again and lsusb -v output, too. >> >> Did it make any difference? > > No, it unfortunately did not. > > lsusb is attached and the parts of the kernel log are in the mail: > > [ 41.243211] usb 2-1.2: new high-speed USB device number 3 using ehci_hcd > [ 41.605666] uvcvideo: Found UVC 1.00 device <unnamed> (046d:0821) > [ 42.002275] input: UVC Camera (046d:0821) as > /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.2/2-1.2:1.2/input/input12 > [ 42.085880] usbcore: registered new interface driver snd-usb-audio > > [ 52.790139] uvcvideo: Failed to query (GET_DEF) UVC control 2 on > unit 2: -71 (exp. 2). > [ 52.794363] uvcvideo: Failed to query (GET_DEF) UVC control 2 on > unit 2: -71 (exp. 2). > [ 52.798375] uvcvideo: Failed to query (GET_DEF) UVC control 3 on > unit 2: -71 (exp. 2). > [ 52.802499] uvcvideo: Failed to query (GET_DEF) UVC control 7 on > unit 2: -71 (exp. 2). > [ 52.806617] uvcvideo: Failed to query (GET_DEF) UVC control 11 on > unit 2: -71 (exp. 1). > [ 52.810734] uvcvideo: Failed to query (GET_DEF) UVC control 4 on > unit 2: -71 (exp. 2). > [ 52.814855] uvcvideo: Failed to query (GET_DEF) UVC control 5 on > unit 2: -71 (exp. 1). > [ 52.818976] uvcvideo: Failed to query (GET_CUR) UVC control 11 on > unit 2: -71 (exp. 1). > [ 52.823096] uvcvideo: Failed to query (GET_DEF) UVC control 8 on > unit 2: -71 (exp. 2). > [ 52.827215] uvcvideo: Failed to query (GET_DEF) UVC control 1 on > unit 2: -71 (exp. 2). > [ 52.831331] uvcvideo: Failed to set UVC probe control : -71 (exp. 26). > [ 52.835459] uvcvideo: Failed to set UVC probe control : -71 (exp. 26). > The last line repeats a few times, then the Failed to query block > repeats again, and so on. But only if I try to use the webcam. > > [ 82.363599] 3:1:3: cannot set freq 32000 to ep 0x86 > Is showing up without using the webcam. > What really puzzles me is that the uvcvideo code fails as well, and even before the sound driver bails out. If you wouldn't have bisected the problem, I would clearly blame some other change for that. It seems something confuses the device quite badly so requests to it return -EPROTO. From a quick grep, this error can only be originated from low-level EHCI functions. I need to ponder on that a little ... Daniel

1 0

Re: [alsa-devel] Regression in sound/usb/
by Takashi Iwai 05 Jul '12

05 Jul '12

At Tue, 3 Jul 2012 10:40:51 +0200, Philipp Dreimann wrote: > > On 3 July 2012 10:27, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 3 Jul 2012 10:20:13 +0200, > > Philipp Dreimann wrote: > >> > >> On 3 July 2012 10:00, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > At Tue, 3 Jul 2012 09:41:09 +0200, > >> > Philipp Dreimann wrote: > >> >> > >> >> On 2 July 2012 14:10, Daniel Mack <zonque(a)gmail.com> wrote: > >> >> > On 01.07.2012 13:40, Philipp Dreimann wrote: > >> >> >> Hello, > >> >> >> > >> >> >> my usb webcam, which happens to have a microphone too, stopped working > >> >> >> using usb2 since I tried kernel 3.5-rc2. (It works using usb3.) > >> >> > > >> >> > Please try 3.5-rc5. A number of regressions were fixed just recently. > >> >> > > >> >> > Let me know if that version still causes problems. > >> >> > >> >> It still does. > >> > > >> > Try to pass ignore_ctl_error=1 to snd-usb-audio module. > >> > If this works and no other problem happens, we can add a similar quirk > >> > like some Logitech webcams. > >> > >> This does not seem to help. > > > > And webcam itself is working with our without snd-usb-audio module? > Yes, if I blacklist the module, the webcam works fine. > > > Which device at all? > logitech c910 > > > Too little information to analyze. > You guys tell me what you need to know. Try the patch below. If it still doesn't work, give the kernel message again and lsusb -v output, too. thanks, Takashi --- diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 54607f8..f0ede13 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -387,7 +387,7 @@ add_sync_ep: subs->data_endpoint->sync_master = subs->sync_endpoint; } - if ((err = snd_usb_init_pitch(subs->stream->chip, subs->interface, alts, fmt)) < 0) + if ((err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt)) < 0) return err; subs->cur_audiofmt = fmt; @@ -450,7 +450,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct usb_interface *iface; iface = usb_ifnum_to_if(subs->dev, fmt->iface); alts = &iface->altsetting[fmt->altset_idx]; - ret = snd_usb_init_sample_rate(subs->stream->chip, subs->interface, alts, fmt, rate); + ret = snd_usb_init_sample_rate(subs->stream->chip, fmt->iface, alts, fmt, rate); if (ret < 0) return ret; subs->cur_rate = rate;

1 1

[alsa-devel] [PATCH] ASoC: wm1250-ev1: Flag all supported rates in the DAI
by Mark Brown 05 Jul '12

05 Jul '12

Not previously noticed due to normal usage being with CODEC<->CODEC links. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm1250-ev1.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index e0b51e9..951d7b4 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -121,20 +121,23 @@ static const struct snd_soc_dai_ops wm1250_ev1_ops = { .hw_params = wm1250_ev1_hw_params, }; +#define WM1250_EV1_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_64000) + static struct snd_soc_dai_driver wm1250_ev1_dai = { .name = "wm1250-ev1", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, + .rates = WM1250_EV1_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, + .rates = WM1250_EV1_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &wm1250_ev1_ops, -- 1.7.10

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: imx-mc13783: Add audmux settings for mx27pdk
by Mark Brown 05 Jul '12

05 Jul '12

On Tue, Jul 03, 2012 at 03:44:58PM -0300, Fabio Estevam wrote: > mx27pdk board also has a mc13783 codec. > > Add support for it and do a run-time machine type check to perform the correct > audiomux settings. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: fsl: remove unneeded AUDMUX register setting from imx-sgtl5000
by Mark Brown 05 Jul '12

05 Jul '12

On Wed, Jul 04, 2012 at 03:38:27PM +0800, Hui Wang wrote: > If we don't set IMX_AUDMUX_V2_PTCR_TCLKDIR in the AUDMUX PTCR > register (means Tx clock pin is input), we don't need to set > IMX_AUDMUX_V2_PTCR_TCSEL as well. Since both i.MX35, i.MX51 and > i.MX6 datasheet says "When Tx clock pin set as an input, the TCSEL > settings are ignored". Applied, thanks.

1 0

[alsa-devel] [PATCH] ALSA: pcm: Make constraints lists const
by Mark Brown 05 Jul '12

05 Jul '12

They aren't modified by the core so the drivers can declare them const. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/pcm.h | 2 +- sound/core/pcm_lib.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e7afcc9..c75c0d1 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -810,7 +810,7 @@ int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_pa int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_list *l); + const struct snd_pcm_hw_constraint_list *l); int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 8f312fa6..8659bc5 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1253,7 +1253,7 @@ int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, struct snd_pcm_hw_constraint_list *l) { return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_list, l, + snd_pcm_hw_rule_list, (void *)l, var, -1); } -- 1.7.10

2 1

[alsa-devel] [PATCH] ALSA: pcm: Mostly constify constraints
by Mark Brown 05 Jul '12

05 Jul '12

There is no need for the constraints to be modified while being applied so they can be declared const, allowing drivers to declare constraint lists const if they wish. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/pcm.h | 6 +++--- sound/core/pcm_lib.c | 10 +++++----- sound/core/pcm_native.c | 4 ++-- 3 files changed, 10 insertions(+), 10 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e7afcc9..3b8f6ca 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -217,7 +217,7 @@ struct snd_pcm_hw_rule { snd_pcm_hw_rule_func_t func; int var; int deps[4]; - void *private; + const void *private; }; struct snd_pcm_hw_constraints { @@ -810,7 +810,7 @@ int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_pa int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_list *l); + const struct snd_pcm_hw_constraint_list *l); int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, @@ -835,7 +835,7 @@ int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime, int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, int var, - snd_pcm_hw_rule_func_t func, void *private, + snd_pcm_hw_rule_func_t func, const void *private, int dep, ...); int snd_pcm_format_signed(snd_pcm_format_t format); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 8f312fa6..4372de5 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1104,7 +1104,7 @@ static int snd_interval_step(struct snd_interval *i, unsigned int min, unsigned */ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, int var, - snd_pcm_hw_rule_func_t func, void *private, + snd_pcm_hw_rule_func_t func, const void *private, int dep, ...) { struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints; @@ -1233,7 +1233,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_minmax); static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hw_constraint_list *list = rule->private; + const struct snd_pcm_hw_constraint_list *list = rule->private; return snd_interval_list(hw_param_interval(params, rule->var), list->count, list->list, list->mask); } @@ -1250,7 +1250,7 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_list *l) + const struct snd_pcm_hw_constraint_list *l) { return snd_pcm_hw_rule_add(runtime, cond, var, snd_pcm_hw_rule_list, l, @@ -1262,7 +1262,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_list); static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hw_constraint_ratnums *r = rule->private; + const struct snd_pcm_hw_constraint_ratnums *r = rule->private; unsigned int num = 0, den = 0; int err; err = snd_interval_ratnum(hw_param_interval(params, rule->var), @@ -1296,7 +1296,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_ratnums); static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hw_constraint_ratdens *r = rule->private; + const struct snd_pcm_hw_constraint_ratdens *r = rule->private; unsigned int num = 0, den = 0; int err = snd_interval_ratden(hw_param_interval(params, rule->var), r->nrats, r->rats, &num, &den); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 53b5ada..5cc8c4f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1775,7 +1775,7 @@ const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { static int snd_pcm_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hardware *hw = rule->private; + const struct snd_pcm_hardware *hw = rule->private; return snd_interval_list(hw_param_interval(params, rule->var), snd_pcm_known_rates.count, snd_pcm_known_rates.list, hw->rates); @@ -1785,7 +1785,7 @@ static int snd_pcm_hw_rule_buffer_bytes_max(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval t; - struct snd_pcm_substream *substream = rule->private; + const struct snd_pcm_substream *substream = rule->private; t.min = 0; t.max = substream->buffer_bytes_max; t.openmin = 0; -- 1.7.10

2 6

[alsa-devel] [PATCH] ALSA: hda - Always call standard unsolicited event
by David Henningsson 05 Jul '12

05 Jul '12

With the model parsers out of the way, we have no custom unsol events to worry about, we can therefore simplify the code. In addition, this fixes a bug on the ASUS TC710, which has only a headphone jack and a mic jack, but no internal mic or speakers. Therefore the unsol_event pointer was not set, and as a result, the jack kcontrols were not correctly updated. BugLink: https://bugs.launchpad.net/bugs/1021192 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Alsa-info: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1021192/+attachment/32… sound/pci/hda/patch_realtek.c | 21 +++------------------ 1 file changed, 3 insertions(+), 18 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c2ff945..04e9243 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -174,7 +174,6 @@ struct alc_spec { /* hooks */ void (*init_hook)(struct hda_codec *codec); - void (*unsol_event)(struct hda_codec *codec, unsigned int res); #ifdef CONFIG_SND_HDA_POWER_SAVE void (*power_hook)(struct hda_codec *codec); #endif @@ -688,7 +687,7 @@ static void alc_update_knob_master(struct hda_codec *codec, hda_nid_t nid) } /* unsolicited event for HP jack sensing */ -static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) +static void alc_unsol_event(struct hda_codec *codec, unsigned int res) { int action; @@ -1024,11 +1023,9 @@ static void alc_init_automute(struct hda_codec *codec) spec->automute_lo = spec->automute_lo_possible; spec->automute_speaker = spec->automute_speaker_possible; - if (spec->automute_speaker_possible || spec->automute_lo_possible) { + if (spec->automute_speaker_possible || spec->automute_lo_possible) /* create a control for automute mode */ alc_add_automute_mode_enum(codec); - spec->unsol_event = alc_sku_unsol_event; - } } /* return the position of NID in the list, or -1 if not found */ @@ -1191,7 +1188,6 @@ static void alc_init_auto_mic(struct hda_codec *codec) snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x/0x%x\n", ext, fixed, dock); - spec->unsol_event = alc_sku_unsol_event; } /* check the availabilities of auto-mute and auto-mic switches */ @@ -2062,14 +2058,6 @@ static int alc_init(struct hda_codec *codec) return 0; } -static void alc_unsol_event(struct hda_codec *codec, unsigned int res) -{ - struct alc_spec *spec = codec->spec; - - if (spec->unsol_event) - spec->unsol_event(codec, res); -} - #ifdef CONFIG_SND_HDA_POWER_SAVE static int alc_check_power_status(struct hda_codec *codec, hda_nid_t nid) { @@ -4271,14 +4259,12 @@ static void set_capture_mixer(struct hda_codec *codec) */ static void alc_auto_init_std(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; alc_auto_init_multi_out(codec); alc_auto_init_extra_out(codec); alc_auto_init_analog_input(codec); alc_auto_init_input_src(codec); alc_auto_init_digital(codec); - if (spec->unsol_event) - alc_inithook(codec); + alc_inithook(codec); } /* @@ -4879,7 +4865,6 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, spec->automute_speaker = 1; spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */ snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT); - spec->unsol_event = alc_sku_unsol_event; snd_hda_gen_add_verbs(&spec->gen, alc_gpio1_init_verbs); } } -- 1.7.9.5

2 2

Re: [alsa-devel] How to handle parameter variation across different boards but same platform
by Arnd Bergmann 05 Jul '12

05 Jul '12

On Thursday 05 July 2012, Vaibhav Hiremath wrote: > On 7/4/2012 7:57 PM, Mark Brown wrote: > > On Wed, Jul 04, 2012 at 02:17:48PM +0000, Hebbar, Gururaja wrote: > >> On Wed, Jul 04, 2012 at 19:31:18, Mark Brown wrote: > > > >>> If these are totally different boards they should have different machine > >>> IDs set so machine_is_() should identify. If that isn't there then you > >>> need to do something custom to your products to identify the boards > >>> further. > > > >> They are different boards with same SoC (AM33xx). So they both are true for > >> machine_is_am33xx(). > > > > What if the device only supports Linux boot from DT, where we set > machine_desc.nr = ~0 ?? > > Does machine ID still gets set? May be I am missing something... No, it doesn't. > As part of my debugging, on AM335xEVM platform machine_is_am33xx() > returns false, since the value of machine_desc.nr is set to ~0. Correct. However, when you are booting using the device tree, parameters such as the clock frequency of a device should be encoded in the device tree itself, so you don't need to know which machine you are on. Arnd

2 1

[alsa-devel] [PATCH] Removing an assert
by Martin Sandsmark 05 Jul '12

05 Jul '12

Hi! There is an assert in snd_hctl_handle_event in alsa-lib that gets triggered on every login for many KDE users. From what I can see it is superflous, and gets triggered because the device in an event has gone away. This case is handled just beneath the assert as well, and I therefore don't see the reason for this assert at all. Entry in bug tracker: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5471 -- Martin Sandsmark

2 1

[PATCH] Remove a useless assert.
by Martin Sandsmark 05 Jul '12

05 Jul '12

The case where the element is unavailable (for example gone away since the event was created) is handled beneath. See also bug 5471. --- src/control/hcontrol.c | 1 - 1 file changed, 1 deletion(-) diff --git a/src/control/hcontrol.c b/src/control/hcontrol.c index 8ffc434..ee1d907 100644 --- a/src/control/hcontrol.c +++ b/src/control/hcontrol.c @@ -761,7 +761,6 @@ static int snd_hctl_handle_event(snd_hctl_t *hctl, snd_ctl_event_t *event) if (event->data.elem.mask & (SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO)) { elem = snd_hctl_find_elem(hctl, &event->data.elem.id); - assert(elem); if (!elem) return -ENOENT; res = snd_hctl_elem_throw_event(elem, event->data.elem.mask & -- 1.7.11.1 --nextPart21993734.SHAHOXH9Wi--

1 0

Re: [alsa-devel] [PATCH] ASoC: fsl: remove unneeded AUDMUX register setting from imx-sgtl5000
by Shawn Guo 05 Jul '12

05 Jul '12

On Wed, Jul 04, 2012 at 03:38:27PM +0800, Hui Wang wrote: > If we don't set IMX_AUDMUX_V2_PTCR_TCLKDIR in the AUDMUX PTCR > register (means Tx clock pin is input), we don't need to set > IMX_AUDMUX_V2_PTCR_TCSEL as well. Since both i.MX35, i.MX51 and > i.MX6 datasheet says "When Tx clock pin set as an input, the TCSEL > settings are ignored". > > Signed-off-by: Hui Wang <jason77.wang(a)gmail.com> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> > --- > sound/soc/fsl/imx-sgtl5000.c | 3 +-- > 1 files changed, 1 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c > index 3a729ca..fb21b17 100644 > --- a/sound/soc/fsl/imx-sgtl5000.c > +++ b/sound/soc/fsl/imx-sgtl5000.c > @@ -95,8 +95,7 @@ static int __devinit imx_sgtl5000_probe(struct platform_device *pdev) > return ret; > } > imx_audmux_v2_configure_port(ext_port, > - IMX_AUDMUX_V2_PTCR_SYN | > - IMX_AUDMUX_V2_PTCR_TCSEL(int_port), > + IMX_AUDMUX_V2_PTCR_SYN, > IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); > if (ret) { > dev_err(&pdev->dev, "audmux external port setup failed\n"); > -- > 1.7.6 >

1 0

[alsa-devel] [PATCH 1/3] ASoC: arizona: Rename current rates tables to bclk_rates
by Mark Brown 04 Jul '12

04 Jul '12

They're the rates for the BCLK, not for the sample rate, so rename so that we don't confuse ourselves. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 3b5730b..67760b4 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -363,7 +363,7 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static const int arizona_48k_rates[] = { +static const int arizona_48k_bclk_rates[] = { -1, 48000, 64000, @@ -385,7 +385,7 @@ static const int arizona_48k_rates[] = { 24576000, }; -static const int arizona_44k1_rates[] = { +static const int arizona_44k1_bclk_rates[] = { -1, 44100, 58800, @@ -445,17 +445,17 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, int bclk, lrclk, wl, frame, sr_val; if (params_rate(params) % 8000) - rates = &arizona_44k1_rates[0]; + rates = &arizona_44k1_bclk_rates[0]; else - rates = &arizona_48k_rates[0]; + rates = &arizona_48k_bclk_rates[0]; - for (i = 0; i < ARRAY_SIZE(arizona_44k1_rates); i++) { + for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { if (rates[i] == snd_soc_params_to_bclk(params)) { bclk = i; break; } } - if (i == ARRAY_SIZE(arizona_44k1_rates)) { + if (i == ARRAY_SIZE(arizona_44k1_bclk_rates)) { arizona_aif_err(dai, "Unsupported sample rate %dHz\n", params_rate(params)); return -EINVAL; -- 1.7.10

1 2

[alsa-devel] [PATCH] ASoC: dwc: Staticise non-exported i2s_start()
by Mark Brown 04 Jul '12

04 Jul '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/dwc/designware_i2s.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 1bd042b..1aa5130 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -107,7 +107,8 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) } } -void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) +static void i2s_start(struct dw_i2s_dev *dev, + struct snd_pcm_substream *substream) { i2s_write_reg(dev->i2s_base, IER, 1); -- 1.7.10

1 0

[alsa-devel] How to handle parameter variation across different boards but same platform
by Hebbar, Gururaja 04 Jul '12

04 Jul '12

hi, Currently on Davinci platform, codec system clock & snd_soc_card parameter is decided using machine_is_xxxx() api. sound/soc/Davinci/davinci-evm.c, inside evm_hw_params() & evm_init(), We have an upcoming board based on TI AM335x soc which has both codec system clock And the codec name (because of i2c bus difference) different from previous am335x based boards. Using machine_is_xxx() is not sufficient because there are 2 other boards on same Platform/machine but with different values. So, how to specify these parameters for the new board? Regards Gururaja

2 5

[alsa-devel] A problem with USB audio devices and the UHCI scheduler
by Monty Montgomery 04 Jul '12

04 Jul '12

A question for Clemens et al... Background: I can run a USB audio device at 6ch/24 bit single duplex, but I cannot run the same device at 16 bit stereo full duplex. UHCI and EHCI both say 'not enough bandwidth' (OHCI works). EHCI's scheduling issues are well known, but I was surprised UHCI could not schedule this either. The cause is simple; UHCI (and EHCI) use the endpoint's wMaxPacketSize for all bandwidth allocation. My device supports all of its sample rates on the same altset, with a wMaxPacketSize matching the highest (96kHz) rate. So, the HCD is reserving most of the USB bus's bandwidth regardless of the actual requirements. How should I approach fixing this in the driver? There's currently no means to communicate a desired bandwidth reservation to the HCD from USB audio, so I'd have to extend even the quirks. I was considering shadowing the endpoint's original wMaxPacketSize in the substream struct, and replacing the value in the endpoint's native descriptor while the stream is running. I have no idea how much trouble that would cause (eg, corruption elsewhere, or middle finger from Linus). Suggestions? For now I have things working by disabling UHCI's initial bandwidth checks (as there is in fact more than enough bandwidth to schedule what I'm doing) but I'd like a more permanent solution. Monty

2 3

[alsa-devel] static build on two machine
by Mohsen Pahlevanzadeh 04 Jul '12

04 Jul '12

Dear all, My co-worker built alsa-lib-1.0.25 and now, i should work on this project, So i get it and compile and compile with the following commands: libtoolize --force --copy --automake aclocal autoheader automake --foreign --copy --add-missing autoconf ./configure --enable-shared=no --enable-static=yes --prefix=/usr/i686 ;make install Now, When i compile my own software, i get two warning however my co-worker doesn't get them: /////////////////////// /usr/i686/lib/libasound.a(pcm_dmix.o):-1: In function `snd_pcm_dmix_sync_ptr': pcm_dmix.c:-1: warning: Warning: snd_pcm_hwsync() is deprecated, consider to use snd_pcm_avail() ///////////////////////// When i run my program my program that my co-worker wrote, crashed, But when he run it, it's crashed. How can i solve this problem? --Best regards Mohsen.

2 1

[alsa-devel] [asoc:for-next 99/101] sound/soc/codecs/sta529.c:400:10: warning: 'ret' may be used uninitialized in this function
by wfg＠linux.intel.com 04 Jul '12

04 Jul '12

Hi Rajeev, There are new compile warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 5a6bb2006960eab999dc7023b344bfdbb82e3059 commit: 3be58dbb92871442191188ae51b449e1a9f0fe64 [99/101] ASoC: STA529: Add support for STA529 Audio Codec All warnings: sound/soc/codecs/sta529.c: In function 'sta529_i2c_probe': sound/soc/codecs/sta529.c:400:10: warning: 'ret' may be used uninitialized in this function [-Wuninitialized] vim +400 sound/soc/codecs/sta529.c 397 398 sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap); 399 if (IS_ERR(sta529->regmap)) { > 400 dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); 401 return PTR_ERR(sta529->regmap); 402 } 403 --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu <wfg(a)linux.intel.com> Intel Corporation

1 0

[alsa-devel] [PATCH V5 1/2] ASoC: STA529: Add support for STA529 Audio Codec
by Rajeev Kumar 04 Jul '12

04 Jul '12

The STA529 is a digital stereo class-D audio amplifier. It includes an audio DSP, an ST proprietary high-efficiency class-D driver and CMOS power output stage. It is intended for high-efficiency digital-to-power-audio conversion for portable applications. Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar(a)st.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta529.c | 441 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 447 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/sta529.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1de24cc..bbcb038 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,6 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SPDIF select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI select SND_SOC_STA32X if I2C + select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -284,6 +285,9 @@ config SND_SOC_SSM2602 config SND_SOC_STA32X tristate +config SND_SOC_STA529 + tristate + config SND_SOC_STAC9766 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index acf8088..8da3d22 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -49,6 +49,7 @@ snd-soc-spdif-tx-objs := spdif_transciever.o snd-soc-spdif-rx-objs := spdif_receiver.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o +snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -163,6 +164,7 @@ obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o +obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c new file mode 100644 index 0000000..a9f34c7 --- /dev/null +++ b/sound/soc/codecs/sta529.c @@ -0,0 +1,441 @@ +/* + * ASoC codec driver for spear platform + * + * sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver + * + * Copyright (C) 2012 ST Microelectronics + * Rajeev Kumar <rajeev-dlh.kumar(a)st.com> + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +/* STA529 Register offsets */ +#define STA529_FFXCFG0 0x00 +#define STA529_FFXCFG1 0x01 +#define STA529_MVOL 0x02 +#define STA529_LVOL 0x03 +#define STA529_RVOL 0x04 +#define STA529_TTF0 0x05 +#define STA529_TTF1 0x06 +#define STA529_TTP0 0x07 +#define STA529_TTP1 0x08 +#define STA529_S2PCFG0 0x0A +#define STA529_S2PCFG1 0x0B +#define STA529_P2SCFG0 0x0C +#define STA529_P2SCFG1 0x0D +#define STA529_PLLCFG0 0x14 +#define STA529_PLLCFG1 0x15 +#define STA529_PLLCFG2 0x16 +#define STA529_PLLCFG3 0x17 +#define STA529_PLLPFE 0x18 +#define STA529_PLLST 0x19 +#define STA529_ADCCFG 0x1E /*mic_select*/ +#define STA529_CKOCFG 0x1F +#define STA529_MISC 0x20 +#define STA529_PADST0 0x21 +#define STA529_PADST1 0x22 +#define STA529_FFXST 0x23 +#define STA529_PWMIN1 0x2D +#define STA529_PWMIN2 0x2E +#define STA529_POWST 0x32 + +#define STA529_MAX_REGISTER 0x32 + +#define STA529_RATES (SNDRV_PCM_RATE_8000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define STA529_FORMAT (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) +#define S2PC_VALUE 0x98 +#define CLOCK_OUT 0x60 +#define LEFT_J_DATA_FORMAT 0x10 +#define I2S_DATA_FORMAT 0x12 +#define RIGHT_J_DATA_FORMAT 0x14 +#define CODEC_MUTE_VAL 0x80 + +#define POWER_CNTLMSAK 0x40 +#define POWER_STDBY 0x40 +#define FFX_MASK 0x80 +#define FFX_OFF 0x80 +#define POWER_UP 0x00 +#define FFX_CLK_ENB 0x01 +#define FFX_CLK_DIS 0x00 +#define FFX_CLK_MSK 0x01 +#define PLAY_FREQ_RANGE_MSK 0x70 +#define CAP_FREQ_RANGE_MSK 0x0C +#define PDATA_LEN_MSK 0xC0 +#define BCLK_TO_FS_MSK 0x30 +#define AUDIO_MUTE_MSK 0x80 + +static const struct reg_default sta529_reg_defaults[] = { + { 0, 0x35 }, /* R0 - FFX Configuration reg 0 */ + { 1, 0xc8 }, /* R1 - FFX Configuration reg 1 */ + { 2, 0x50 }, /* R2 - Master Volume */ + { 3, 0x00 }, /* R3 - Left Volume */ + { 4, 0x00 }, /* R4 - Right Volume */ + { 10, 0xb2 }, /* R10 - S2P Config Reg 0 */ + { 11, 0x41 }, /* R11 - S2P Config Reg 1 */ + { 12, 0x92 }, /* R12 - P2S Config Reg 0 */ + { 13, 0x41 }, /* R13 - P2S Config Reg 1 */ + { 30, 0xd2 }, /* R30 - ADC Config Reg */ + { 31, 0x40 }, /* R31 - clock Out Reg */ + { 32, 0x21 }, /* R32 - Misc Register */ +}; + +struct sta529 { + struct regmap *regmap; +}; + +static bool sta529_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + + case STA529_FFXCFG0: + case STA529_FFXCFG1: + case STA529_MVOL: + case STA529_LVOL: + case STA529_RVOL: + case STA529_S2PCFG0: + case STA529_S2PCFG1: + case STA529_P2SCFG0: + case STA529_P2SCFG1: + case STA529_ADCCFG: + case STA529_CKOCFG: + case STA529_MISC: + return true; + default: + return false; + } +} + + +static const char *pwm_mode_text[] = { "Binary", "Headphone", "Ternary", + "Phase-shift"}; + +static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -9150, 50, 0); +static const DECLARE_TLV_DB_SCALE(master_vol_tlv, -12750, 50, 0); +static const SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text); + +static const struct snd_kcontrol_new sta529_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume", STA529_LVOL, STA529_RVOL, 0, + 127, 0, out_gain_tlv), + SOC_SINGLE_TLV("Master Playback Volume", STA529_MVOL, 0, 127, 1, + master_vol_tlv), + SOC_ENUM("PWM Select", pwm_src), +}; + +static int sta529_set_bias_level(struct snd_soc_codec *codec, enum + snd_soc_bias_level level) +{ + struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, STA529_FFXCFG0, POWER_CNTLMSAK, + POWER_UP); + snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK, + FFX_CLK_ENB); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(sta529->regmap); + snd_soc_update_bits(codec, STA529_FFXCFG0, + POWER_CNTLMSAK, POWER_STDBY); + /* Making FFX output to zero */ + snd_soc_update_bits(codec, STA529_FFXCFG0, FFX_MASK, + FFX_OFF); + snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK, + FFX_CLK_DIS); + break; + case SND_SOC_BIAS_OFF: + break; + } + + /* + * store the label for powers down audio subsystem for suspend.This is + * used by soc core layer + */ + codec->dapm.bias_level = level; + + return 0; + +} + +static int sta529_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + int pdata, play_freq_val, record_freq_val; + int bclk_to_fs_ratio; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + pdata = 1; + bclk_to_fs_ratio = 0; + break; + case SNDRV_PCM_FORMAT_S24_LE: + pdata = 2; + bclk_to_fs_ratio = 1; + break; + case SNDRV_PCM_FORMAT_S32_LE: + pdata = 3; + bclk_to_fs_ratio = 2; + break; + default: + dev_err(codec->dev, "Unsupported format\n"); + return -EINVAL; + } + + switch (params_rate(params)) { + case 8000: + case 11025: + play_freq_val = 0; + record_freq_val = 2; + break; + case 16000: + case 22050: + play_freq_val = 1; + record_freq_val = 0; + break; + + case 32000: + case 44100: + case 48000: + play_freq_val = 2; + record_freq_val = 0; + break; + default: + dev_err(codec->dev, "Unsupported rate\n"); + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + snd_soc_update_bits(codec, STA529_S2PCFG1, PDATA_LEN_MSK, + pdata << 6); + snd_soc_update_bits(codec, STA529_S2PCFG1, BCLK_TO_FS_MSK, + bclk_to_fs_ratio << 4); + snd_soc_update_bits(codec, STA529_MISC, PLAY_FREQ_RANGE_MSK, + play_freq_val << 4); + } else { + snd_soc_update_bits(codec, STA529_P2SCFG1, PDATA_LEN_MSK, + pdata << 6); + snd_soc_update_bits(codec, STA529_P2SCFG1, BCLK_TO_FS_MSK, + bclk_to_fs_ratio << 4); + snd_soc_update_bits(codec, STA529_MISC, CAP_FREQ_RANGE_MSK, + record_freq_val << 2); + } + + return 0; +} + +static int sta529_mute(struct snd_soc_dai *dai, int mute) +{ + u8 val = 0; + + if (mute) + val |= CODEC_MUTE_VAL; + + snd_soc_update_bits(dai->codec, STA529_FFXCFG0, AUDIO_MUTE_MSK, val); + + return 0; +} + +static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 mode = 0; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + mode = LEFT_J_DATA_FORMAT; + break; + case SND_SOC_DAIFMT_I2S: + mode = I2S_DATA_FORMAT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + mode = RIGHT_J_DATA_FORMAT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode); + + return 0; +} + +static const struct snd_soc_dai_ops sta529_dai_ops = { + .hw_params = sta529_hw_params, + .set_fmt = sta529_set_dai_fmt, + .digital_mute = sta529_mute, +}; + +static struct snd_soc_dai_driver sta529_dai = { + .name = "sta529-audio", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA529_RATES, + .formats = STA529_FORMAT, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = STA529_RATES, + .formats = STA529_FORMAT, + }, + .ops = &sta529_dai_ops, +}; + +static int sta529_probe(struct snd_soc_codec *codec) +{ + struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = sta529->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* power down chip */ +static int sta529_remove(struct snd_soc_codec *codec) +{ + sta529_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int sta529_suspend(struct snd_soc_codec *codec) +{ + sta529_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int sta529_resume(struct snd_soc_codec *codec) +{ + sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +struct snd_soc_codec_driver sta529_codec_driver = { + .probe = sta529_probe, + .remove = sta529_remove, + .set_bias_level = sta529_set_bias_level, + .suspend = sta529_suspend, + .resume = sta529_resume, + .controls = sta529_snd_controls, + .num_controls = ARRAY_SIZE(sta529_snd_controls), +}; + +static const struct regmap_config sta529_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = STA529_MAX_REGISTER, + .readable_reg = sta529_readable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = sta529_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(sta529_reg_defaults), +}; + +static __devinit int sta529_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta529 *sta529; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL); + if (sta529 == NULL) { + dev_err(&i2c->dev, "Can not allocate memory\n"); + return -ENOMEM; + } + + sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap); + if (IS_ERR(sta529->regmap)) { + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return PTR_ERR(sta529->regmap); + } + + i2c_set_clientdata(i2c, sta529); + + ret = snd_soc_register_codec(&i2c->dev, + &sta529_codec_driver, &sta529_dai, 1); + if (ret != 0) + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + + return ret; +} + +static int __devexit sta529_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id sta529_i2c_id[] = { + { "sta529", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta529_i2c_id); + +static struct i2c_driver sta529_i2c_driver = { + .driver = { + .name = "sta529", + .owner = THIS_MODULE, + }, + .probe = sta529_i2c_probe, + .remove = __devexit_p(sta529_i2c_remove), + .id_table = sta529_i2c_id, +}; + +module_i2c_driver(sta529_i2c_driver); + +MODULE_DESCRIPTION("ASoC STA529 codec driver"); +MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar(a)st.com>"); +MODULE_LICENSE("GPL"); -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: add deemphasis switch
by Prchal Jiří 04 Jul '12

04 Jul '12

This patch adds missing deemphasis switch. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 4ed6c20..2265e94 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -381,6 +381,9 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("Left AGC Decay time", aic3x_agc_enum[LAGC_DEC_ENUM]), SOC_ENUM("Right AGC Decay time", aic3x_agc_enum[RAGC_DEC_ENUM]), + /* De-emphasis */ + SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), + /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 119, 0, adc_tlv), -- 1.7.7

2 1

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic3x: add missing registers and bits
by Jiri Prchal 04 Jul '12

04 Jul '12

Adds register and bit shift definitions in header file. Changes are for TLV320AIC310x based on data sheet. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x.h | 19 ++++++++++++++++++- 1 files changed, 18 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 06a1978..586c5d5 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -13,7 +13,7 @@ #define _AIC3X_H /* AIC3X register space */ -#define AIC3X_CACHEREGNUM 103 +#define AIC3X_CACHEREGNUM 110 /* Page select register */ #define AIC3X_PAGE_SELECT 0 @@ -74,6 +74,8 @@ #define HPLCOM_CFG 37 /* Right High Power Output control registers */ #define HPRCOM_CFG 38 +/* High Power Output Stage Control Register */ +#define HPOUT_SC 40 /* DAC Output Switching control registers */ #define DAC_LINE_MUX 41 /* High Power Output Driver Pop Reduction registers */ @@ -148,6 +150,17 @@ #define AIC3X_GPIOB_REG 101 /* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 +/* New AGC registers */ +#define LAGCN_ATTACK 103 +#define LAGCN_DECAY 104 +#define RAGCN_ATTACK 105 +#define RAGCN_DECAY 106 +/* New Programmable ADC Digital Path and I2C Bus Condition Register */ +#define NEW_ADC_DIGITALPATH 107 +/* Passive Analog Signal Bypass Selection During Powerdown Register */ +#define PASSIVE_BYPASS 108 +/* DAC Quiescent Current Adjustment Register */ +#define DAC_ICC_ADJ 109 /* Page select register bits */ #define PAGE0_SELECT 0 @@ -163,6 +176,10 @@ #define DUAL_RATE_MODE ((1 << 5) | (1 << 6)) #define LDAC2LCH (0x1 << 3) #define RDAC2RCH (0x1 << 1) +#define LDAC2RCH (0x2 << 3) +#define RDAC2LCH (0x2 << 1) +#define LDAC2MONOMIX (0x3 << 3) +#define RDAC2MONOMIX (0x3 << 1) /* PLL registers bitfields */ #define PLLP_SHIFT 0 -- 1.7.7

2 2

Re: [alsa-devel] QasTools 0.17.0 released
by Raymond Yau 04 Jul '12

04 Jul '12

> QasTools version 0.17.0 is available. > QasTools is a collection of dektop applications for ALSA. > Changes: > * all: setting names for state save/restore changed/regrouped > * all: smart main window sizeHint() to get a useable window > size after leaving fullscreen > * all: new common info dialog - uses QListView page > selection intead of tabs > * shared: device selection style tweaks: selected item > now emphasized by a thin frame > * shared: device selection style tweaks: Rounded selection area > * shared: use QSplitter instead of QDockwidget for device selection > view. It couldn't be detached anyway and size restoring > works more reliable. > * qasconfig: settings directory changed to shared ~/.config/qastools/ > (was ~/.config/qasconfig/) > * qasconfig: new basic command line options (--version, --help, etc.) > * qashctl: new basic command line options (--version, --help, etc.) > * qasmixer: new application icon qasmixer only show a mono Master and Capture volume controls/switch of "pulse" when user selected 4.0 / 5.1/ 7.1 analog output and stereo input in sound setting of ubuntu 12.04 , amixer and alsamixer show the volume control with correct channels apt-cache policy qasmixer qasmixer: Installed: 0.17.1-1 Candidate: 0.17.1-1 amixer -Dpulse Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right - Rear Left - Rear Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 45314 [69%] [on] Front Right: Playback 45314 [69%] [on] Rear Left: Playback 45314 [69%] [on] Rear Right: Playback 45314 [69%] [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch cswitch-joined penum Capture channels: Front Left - Front Right Limits: Capture 0 - 65536 Front Left: Capture 65535 [100%] [off] Front Right: Capture 65535 [100%] [off]

2 1

[alsa-devel] [PATCH 0/4] ALSA: hda beep clean up
by Takashi Iwai 04 Jul '12

04 Jul '12

Hi, this is a series of small cleanups of HD-audio beep code. Since the keyboard driver already handles the simultaneous outputs from the multiple input beep instances, there is no merit to have beep_mode=2 option to switch the beep input device. Takashi

2 9

[alsa-devel] [PATCH V4 0/2] Add support for sta529 audio codec and spdif_out controller
by Rajeev Kumar 04 Jul '12

04 Jul '12

The patch set contains minor modification according to review comments. Patches that got applied(from V3 of patch set) are listed below: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-June/052868.html 1. [PATCH V3 1/4] ASoC: spdif_receiver: Add support for spdif in Audio Codec. Patches that got applied(from V2 of patch set) are listed below: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-June/052868.html 1. [PATCH V2 3/9] sound:asoc: Add support for synopsys i2s controller as per ASoC framework. 2. [PATCH V2 4/9] sound:asoc: Add support for SPEAr ASoC pcm layer. 3. [PATCH V2 5/9] sound:asoc:spdif_in: Add spdif IN support Rajeev Kumar (1): ASoC: STA529: Add support for STA529 Audio Codec Vipin Kumar (1): ASoC: SPEAr spdif_out: Add spdif out support sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta529.c | 440 ++++++++++++++++++++++++++++++++++++++ sound/soc/spear/spdif_out.c | 388 +++++++++++++++++++++++++++++++++ sound/soc/spear/spdif_out_regs.h | 79 +++++++ 5 files changed, 913 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/spear/spdif_out.c create mode 100644 sound/soc/spear/spdif_out_regs.h

2 6

[alsa-devel] [PATCH V3 0/4] Add support for STA529 codec, spdif_receiver and SPEAr spdif_out controller
by Rajeev Kumar 03 Jul '12

03 Jul '12

Earlier I had circulated Patch set V2. Please find below the link for the same. http://mailman.alsa-project.org/pipermail/alsa-devel/2012-June/052868.html. Some of the patches got applied and for rest I am sending Version V3. Patches that got applied(from V2 of patch set) are listed below: 1. [PATCH V2 3/9] sound:asoc: Add support for synopsys i2s controller as per ASoC framework. 2. [PATCH V2 4/9] sound:asoc: Add support for SPEAr ASoC pcm layer. 3. [PATCH V2 5/9] sound:asoc:spdif_in: Add spdif IN support Changes since V2: 1. In-corporated review comments received on V2 Rajeev Kumar (2): ASoC: STA529: Add support for STA529 Audio Codec ASoC: SPEAr: Add Kconfig and Makefile to support SPEAr audio driver Vipin Kumar (2): ASoC: spdif_receiver: Add support for spdif in Audio Codec ASoC: SPEAr spdif_out: Add spdif out support sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/spdif_receiver.c | 67 ++++++ sound/soc/codecs/sta529.c | 441 +++++++++++++++++++++++++++++++++++++ sound/soc/spear/Kconfig | 22 ++ sound/soc/spear/Makefile | 4 + sound/soc/spear/spdif_out.c | 387 ++++++++++++++++++++++++++++++++ sound/soc/spear/spdif_out_regs.h | 79 +++++++ 8 files changed, 1007 insertions(+), 1 deletions(-) create mode 100644 sound/soc/codecs/spdif_receiver.c create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/spear/spdif_out.c create mode 100644 sound/soc/spear/spdif_out_regs.h

3 13

[alsa-devel] [PATCH] ASoC: dapm: Fix snd_soc_dapm_put_volsw() connect
by Benoît Thébaudeau 03 Jul '12

03 Jul '12

snd_soc_dapm_put_volsw() sets connect incorrectly in the case max > 1 with invert. In that case, the raw disconnect value should be max, which corresponds to the userspace value 0. This use case currently does not appear upstream, but it could break SOC_DAPM_SINGLE() or SOC_DAPM_SINGLE_TLV() elsewhere or in the future. Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau(a)advansee.com> --- .../sound/soc/soc-dapm.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git linux-next-HEAD-6c86b58.orig/sound/soc/soc-dapm.c linux-next-HEAD-6c86b58/sound/soc/soc-dapm.c index 405841c..5ef082f 100644 --- linux-next-HEAD-6c86b58.orig/sound/soc/soc-dapm.c +++ linux-next-HEAD-6c86b58/sound/soc/soc-dapm.c @@ -2515,19 +2515,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int wi; val = (ucontrol->value.integer.value[0] & mask); + connect = !!val; if (invert) val = max - val; mask = mask << shift; val = val << shift; - if (val) - /* new connection */ - connect = invert ? 0 : 1; - else - /* old connection must be powered down */ - connect = invert ? 1 : 0; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, reg, mask, val);

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Re: [alsa-devel] [PATCH] ASoC: tegra: use dmaengine based dma driver
by Mark Brown 03 Jul '12

03 Jul '12

On Fri, Jun 29, 2012 at 05:04:33PM +0530, Laxman Dewangan wrote: > Use the dmaengine based Tegra APB DMA driver for > data transfer between SPI fifo and memory in > place of legacy Tegra APB DMA. Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: dapm: Remove stereo from snd_soc_dapm_get_volsw()
by Benoît Thébaudeau 03 Jul '12

03 Jul '12

Stereo is not yet supported by dapm widgets, so remove stereo code from snd_soc_dapm_get_volsw(). Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau(a)advansee.com> --- .../sound/soc/soc-dapm.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git linux-next-HEAD-5d2b7d6.orig/sound/soc/soc-dapm.c linux-next-HEAD-5d2b7d6/sound/soc/soc-dapm.c index c93e832..9333e01 100644 --- linux-next-HEAD-5d2b7d6.orig/sound/soc/soc-dapm.c +++ linux-next-HEAD-5d2b7d6/sound/soc/soc-dapm.c @@ -2464,23 +2464,15 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; int max = mc->max; - unsigned int invert = mc->invert; unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; ucontrol->value.integer.value[0] = (snd_soc_read(widget->codec, reg) >> shift) & mask; - if (shift != rshift) - ucontrol->value.integer.value[1] = - (snd_soc_read(widget->codec, reg) >> rshift) & mask; - if (invert) { + if (invert) ucontrol->value.integer.value[0] = max - ucontrol->value.integer.value[0]; - if (shift != rshift) - ucontrol->value.integer.value[1] = - max - ucontrol->value.integer.value[1]; - } return 0; }

2 2

[alsa-devel] [PATCH v2] ASoC: tlv320aic3x: add missing registers and bits
by Prchal Jiří 03 Jul '12

03 Jul '12

Second version of patch keeping formatting. Adds missing register default values to cache. Adds register and bits definitions in header file. Changes are for TLV320AIC310x based on data sheet. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.orig +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.new @@ -122,7 +122,9 @@ 0x00, 0x00, 0x00, 0x00, /* 88 */ 0x00, 0x00, 0x00, 0x00, /* 92 */ 0x00, 0x00, 0x00, 0x00, /* 96 */ - 0x00, 0x00, 0x02, /* 100 */ + 0x00, 0x00, 0x02, 0x00, /* 100 */ + 0x00, 0x00, 0x00, 0x00, /* 104 */ + 0x00, 0x00, /* 108 */ }; --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h.new @@ -13,7 +13,7 @@ #define _AIC3X_H /* AIC3X register space */ -#define AIC3X_CACHEREGNUM 103 +#define AIC3X_CACHEREGNUM 110 /* Page select register */ #define AIC3X_PAGE_SELECT 0 @@ -74,6 +74,8 @@ #define HPLCOM_CFG 37 /* Right High Power Output control registers */ #define HPRCOM_CFG 38 +/* High Power Output Stage Control Register */ +#define HPOUT_SC 40 /* DAC Output Switching control registers */ #define DAC_LINE_MUX 41 /* High Power Output Driver Pop Reduction registers */ @@ -148,6 +150,17 @@ #define AIC3X_GPIOB_REG 101 /* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 +/* New AGC registers */ +#define LAGCN_ATTACK 103 +#define LAGCN_DECAY 104 +#define RAGCN_ATTACK 105 +#define RAGCN_DECAY 106 +/* New Programmable ADC Digital Path and I2C Bus Condition Register */ +#define NEW_ADC_DIGITALPATH 107 +/* Passive Analog Signal Bypass Selection During Powerdown Register */ +#define PASSIVE_BYPASS 108 +/* DAC Quiescent Current Adjustment Register */ +#define DAC_ICC_ADJ 109 /* Page select register bits */ #define PAGE0_SELECT 0 @@ -163,6 +176,10 @@ #define DUAL_RATE_MODE ((1 << 5) | (1 << 6)) #define LDAC2LCH (0x1 << 3) #define RDAC2RCH (0x1 << 1) +#define LDAC2RCH (0x2 << 3) +#define RDAC2LCH (0x2 << 1) +#define LDAC2MONOMIX (0x3 << 3) +#define RDAC2MONOMIX (0x3 << 1) /* PLL registers bitfields */ #define PLLP_SHIFT 0

2 5

Re: [alsa-devel] [PATCH 1/1] ASoC: Samsung: Convert to use snd_soc_register_card()
by Mark Brown 03 Jul '12

03 Jul '12

On Tue, Jul 03, 2012 at 02:04:04PM +0530, Sachin Kamat wrote: > Current method for machine driver to register with the ASoC core is to use > snd_soc_register_card() instead of creating a "soc-audio" platform device. Applied, thanks (it's not a generic Samsung patch, it's just for the one board BTW).

1 0

Re: [alsa-devel] [PATCH 1/1] ASoC: Samsung: Convert to use snd_soc_register_card()
by Sangbeom Kim 03 Jul '12

03 Jul '12

Hi! On Tuesday, Jul 03, 2012 at 05:34PM, Sachin Kamat wrote: > Current method for machine driver to register with the ASoC core is to use > snd_soc_register_card() instead of creating a "soc-audio" platform device. It looks better to include reason of using snd_soc_register_card instead of creating "soc-audio" or warning message. Acked-by: Sangbeom Kim <sbkim73(a)samsung.com>

1 0

Re: [alsa-devel] [PATCH] ASoC: tegra: use dmaengine based dma driver
by Mark Brown 03 Jul '12

03 Jul '12

On Tue, Jul 03, 2012 at 11:59:31AM +0530, Laxman Dewangan wrote: > Are you taking this patch? > I have other independent patch for the writecombine dma support in > pcm memory library and wanted to avoid any merge conflicts. > So if it goes in your tree then the other patch also need to go in > your tree. You've not even left 24 hours for me to respond!

1 0

Re: [alsa-devel] Regression in sound/usb/
by Takashi Iwai 03 Jul '12

03 Jul '12

At Tue, 3 Jul 2012 10:20:13 +0200, Philipp Dreimann wrote: > > On 3 July 2012 10:00, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 3 Jul 2012 09:41:09 +0200, > > Philipp Dreimann wrote: > >> > >> On 2 July 2012 14:10, Daniel Mack <zonque(a)gmail.com> wrote: > >> > On 01.07.2012 13:40, Philipp Dreimann wrote: > >> >> Hello, > >> >> > >> >> my usb webcam, which happens to have a microphone too, stopped working > >> >> using usb2 since I tried kernel 3.5-rc2. (It works using usb3.) > >> > > >> > Please try 3.5-rc5. A number of regressions were fixed just recently. > >> > > >> > Let me know if that version still causes problems. > >> > >> It still does. > > > > Try to pass ignore_ctl_error=1 to snd-usb-audio module. > > If this works and no other problem happens, we can add a similar quirk > > like some Logitech webcams. > > This does not seem to help. And webcam itself is working with our without snd-usb-audio module? Which device at all? Too little information to analyze. Takashi

1 0

Re: [alsa-devel] Regression in sound/usb/
by Takashi Iwai 03 Jul '12

03 Jul '12

At Tue, 3 Jul 2012 09:41:09 +0200, Philipp Dreimann wrote: > > On 2 July 2012 14:10, Daniel Mack <zonque(a)gmail.com> wrote: > > On 01.07.2012 13:40, Philipp Dreimann wrote: > >> Hello, > >> > >> my usb webcam, which happens to have a microphone too, stopped working > >> using usb2 since I tried kernel 3.5-rc2. (It works using usb3.) > > > > Please try 3.5-rc5. A number of regressions were fixed just recently. > > > > Let me know if that version still causes problems. > > It still does. Try to pass ignore_ctl_error=1 to snd-usb-audio module. If this works and no other problem happens, we can add a similar quirk like some Logitech webcams. Takashi

1 0

[alsa-devel] [PATCH] ASoC: wm8962: Log AIF configuration requested by hw_params()
by Mark Brown 02 Jul '12

02 Jul '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index aa5839c..5781581 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2582,6 +2582,9 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, WM8962_SAMPLE_RATE_INT_MODE | WM8962_SAMPLE_RATE_MASK, adctl3); + dev_dbg(codec->dev, "hw_params set BCLK %dHz LRCLK %dHz\n", + wm8962->bclk, wm8962->lrclk); + if (codec->dapm.bias_level == SND_SOC_BIAS_ON) wm8962_configure_bclk(codec); -- 1.7.10

1 0

Re: [alsa-devel] [PATCH] ASoC: tegra: use dmaengine based dma driver
by Stephen Warren 02 Jul '12

02 Jul '12

On 07/02/2012 02:25 AM, Laxman Dewangan wrote: > On Saturday 30 June 2012 08:38 PM, Laxman Dewangan wrote: >> On Friday 29 June 2012 10:32 PM, Stephen Warren wrote: >>> On 06/29/2012 05:34 AM, Laxman Dewangan wrote: >>>> Use the dmaengine based Tegra APB DMA driver for >>>> data transfer between SPI fifo and memory in >>>> place of legacy Tegra APB DMA. >>>> >>>> Because generic soc-dmaengine-pcm uses the DMAs API >>>> based on dmaengine, using the exported APIs provided >>>> by this generic driver. >>>> >>>> The new driver is selected if legacy driver is not >>>> selected and new dma driver is enabled through config >>>> file. >>> >>> This works just fine with the existing non-dmaengine DMA driver enabled. >>> >>> However, I can't get it to work with dmaengine: ... >> I think I should send that patches so that you can test it by: >> - Taking already applied dma driver change in linux-next. >> - Apply my new patches which I am going to send. >> >> And do local change in the tegra_defconfig to disable SYSTEM_DMA and >> enable dmaengine based dma driver. > > The new dma patches are: > > dma: tegra: fix residual calculation for cyclic case > dma: tegra: rename driver and compatible to match with dts > ... OK, with those two patches plus this one, everything works on Cardhu/Tegra30 using dmaengine. However, on Tegra20 DMA doesn't appear to work. No audio is heard, and aplay eventually exits with the following error message: aplay: pcm_write:1603: write error: Input/output error I believe this means that the DMA simply isn't operating. I'll assume this is a Tegra20-specific issue in the dmaengine driver itself, and hence not a problem with this patch, so I'll ack this patch.

2 1

Re: [alsa-devel] [PATCH] ASoC: tegra: use dmaengine based dma driver
by Stephen Warren 02 Jul '12

02 Jul '12

On 06/29/2012 05:34 AM, Laxman Dewangan wrote: > Use the dmaengine based Tegra APB DMA driver for > data transfer between SPI fifo and memory in > place of legacy Tegra APB DMA. > > Because generic soc-dmaengine-pcm uses the DMAs API > based on dmaengine, using the exported APIs provided > by this generic driver. > > The new driver is selected if legacy driver is not > selected and new dma driver is enabled through config > file. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> Acked-by: Stephen Warren <swarren(a)wwwdotorg.org> Tested-by: Stephen Warren <swarren(a)wwwdotorg.org> (Tested on Cardhu/Tegra30. It fails on Whistler/Tegra20, but I assume that's due to a bug in the dmaengine driver not this patch. There won't be a regression when applying this patch since we haven't switched to dmaengine as the default yet).

1 0

Re: [alsa-devel] Regression in sound/usb/
by Daniel Mack 02 Jul '12

02 Jul '12

On 01.07.2012 13:40, Philipp Dreimann wrote: > Hello, > > my usb webcam, which happens to have a microphone too, stopped working > using usb2 since I tried kernel 3.5-rc2. (It works using usb3.) Please try 3.5-rc5. A number of regressions were fixed just recently. Let me know if that version still causes problems. Daniel

1 0

[alsa-devel] 24 bit sample embedded in 32 bits
by Vipin Kumar 02 Jul '12

02 Jul '12

Hello, I assume that the alsa framework expects the 24 bit sample data to be packed one after the other. for example while recording a 2 channel, 24 bit sample data, typically the dma reads a fifo (which receives data from adc) and prepares a buffer which is then passed to the upper layers. This buffer is expeted to be 24byte left channel and then 24 byte right channel. Am I right? If the above assumption is correct, I need to find a way to tell the framework that the data is 24bit but is embedded in 32 bit. ie 32bit left and then 32 bit right channel data I am tending to think that there has to be a way but I am unable to find it. Can you please point me to the right direction Regards Vipin

3 3

Re: [alsa-devel] [PATCH 2/2] ASoC: tegra: use sound pcm library for substream dma buffer
by Takashi Iwai 02 Jul '12

02 Jul '12

At Mon, 2 Jul 2012 14:24:33 +0530, Laxman Dewangan wrote: > > The sound pcm library support the writecombine DMA buffer > allocation/deallocation/mapping and hence using the APIs > form sound pcm library inplace of implementing the same > locally. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> > --- > Changes from V1: > Changes done to use the existing APIs for sound pcm memory management > which is now supporting writecombine. > > sound/soc/tegra/tegra_pcm.c | 112 ++++++++++-------------------------------- > 1 files changed, 27 insertions(+), 85 deletions(-) > > diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c > index 127348d..ec1fa25 100644 > --- a/sound/soc/tegra/tegra_pcm.c > +++ b/sound/soc/tegra/tegra_pcm.c > @@ -202,8 +202,14 @@ static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, > { > struct snd_pcm_runtime *runtime = substream->runtime; > struct tegra_runtime_data *prtd = runtime->private_data; > + int ret; > > - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); > + ret = snd_pcm_lib_malloc_pages(substream, > + params_buffer_bytes(params)); > + if (ret < 0) { > + pr_err("page allocation failed, err %d\n", ret); Do you really want to spew error messages at each time? > + return ret; > + } > > prtd->dma_req[0].size = params_period_bytes(params); > prtd->dma_req[1].size = prtd->dma_req[0].size; > @@ -213,9 +219,7 @@ static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, > > static int tegra_pcm_hw_free(struct snd_pcm_substream *substream) > { > - snd_pcm_set_runtime_buffer(substream, NULL); > - > - return 0; > + return snd_pcm_lib_free_pages(substream); > } > > static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd) > @@ -263,18 +267,6 @@ static snd_pcm_uframes_t tegra_pcm_pointer(struct snd_pcm_substream *substream) > return prtd->period_index * runtime->period_size; > } > > - > -static int tegra_pcm_mmap(struct snd_pcm_substream *substream, > - struct vm_area_struct *vma) > -{ > - struct snd_pcm_runtime *runtime = substream->runtime; > - > - return dma_mmap_writecombine(substream->pcm->card->dev, vma, > - runtime->dma_area, > - runtime->dma_addr, > - runtime->dma_bytes); > -} > - > static struct snd_pcm_ops tegra_pcm_ops = { > .open = tegra_pcm_open, > .close = tegra_pcm_close, > @@ -283,87 +275,37 @@ static struct snd_pcm_ops tegra_pcm_ops = { > .hw_free = tegra_pcm_hw_free, > .trigger = tegra_pcm_trigger, > .pointer = tegra_pcm_pointer, > - .mmap = tegra_pcm_mmap, > + .mmap = snd_pcm_lib_default_mmap, You can simply remove .mmap definition. NULL falls back to the default mmap handler. > }; > > -static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) > -{ > - struct snd_pcm_substream *substream = pcm->streams[stream].substream; > - struct snd_dma_buffer *buf = &substream->dma_buffer; > - size_t size = tegra_pcm_hardware.buffer_bytes_max; > - > - buf->area = dma_alloc_writecombine(pcm->card->dev, size, > - &buf->addr, GFP_KERNEL); > - if (!buf->area) > - return -ENOMEM; > - > - buf->dev.type = SNDRV_DMA_TYPE_DEV; > - buf->dev.dev = pcm->card->dev; > - buf->private_data = NULL; > - buf->bytes = size; > - > - return 0; > -} > - > -static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream) > -{ > - struct snd_pcm_substream *substream; > - struct snd_dma_buffer *buf; > - > - substream = pcm->streams[stream].substream; > - if (!substream) > - return; > - > - buf = &substream->dma_buffer; > - if (!buf->area) > - return; > - > - dma_free_writecombine(pcm->card->dev, buf->bytes, > - buf->area, buf->addr); > - buf->area = NULL; > -} > - > -static u64 tegra_dma_mask = DMA_BIT_MASK(32); > - > +#define TEGRA_PCM_PREALLOC_BUFFER (32 * 1024) > +#define TEGRA_PCM_PREALLOC_BUFFER_MAX (32 * 1024) > static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > - struct snd_card *card = rtd->card->snd_card; > struct snd_pcm *pcm = rtd->pcm; > - int ret = 0; > - > - if (!card->dev->dma_mask) > - card->dev->dma_mask = &tegra_dma_mask; > - if (!card->dev->coherent_dma_mask) > - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); The mask setup is still required even if you use the preallocation helper. > - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { > - ret = tegra_pcm_preallocate_dma_buffer(pcm, > - SNDRV_PCM_STREAM_PLAYBACK); > - if (ret) > - goto err; > - } > - > - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { > - ret = tegra_pcm_preallocate_dma_buffer(pcm, > - SNDRV_PCM_STREAM_CAPTURE); > - if (ret) > - goto err_free_play; > + struct snd_card *card = rtd->card->snd_card; > + int retval = 0; > + > + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || > + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { You can omit this check. snd_pcm_lib_preallocate_pages_for_all() will ignore the streams that have no substreams assigned. Takashi > + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, > + SNDRV_DMA_TYPE_DEV_WC, > + card->dev, > + TEGRA_PCM_PREALLOC_BUFFER, > + TEGRA_PCM_PREALLOC_BUFFER_MAX); > + if (retval < 0) > + dev_err(card->dev, > + "dma buffer pre-alloc failied %d\n", retval); > } > - > - return 0; > - > -err_free_play: > - tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); > -err: > - return ret; > + return retval; > } > > static void tegra_pcm_free(struct snd_pcm *pcm) > { > - tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); > - tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); > + snd_pcm_lib_preallocate_free_for_all(pcm); > } > > + > static struct snd_soc_platform_driver tegra_pcm_platform = { > .ops = &tegra_pcm_ops, > .pcm_new = tegra_pcm_new, > -- > 1.7.1.1 >

2 1

Re: [alsa-devel] [PATCH V2 1/2] ALSA: Support writecombine DMA buffer memory page alloc/free
by Takashi Iwai 02 Jul '12

02 Jul '12

At Mon, 2 Jul 2012 14:24:32 +0530, Laxman Dewangan wrote: > > Some of architecture like ARM support the writecombine DMA > buffer which is used for pcm substream DMA buffer. > > Supporting writecombine DMA buffer allocation/free/mmap > through the sound library memory management. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> > --- > Changes from V1: > Integrate the changes from Takashi's sample code for supporting > writecombine. Could you give a short comment in each file that ifdef CONFIG_ARM is a workaround? thanks, Takashi > > include/sound/memalloc.h | 1 + > sound/core/memalloc.c | 74 +++++++++++++++++++++++++++++++++++++++++----- > sound/core/pcm_native.c | 10 ++++++ > 3 files changed, 77 insertions(+), 8 deletions(-) > > diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h > index c425062..0328e37 100644 > --- a/include/sound/memalloc.h > +++ b/include/sound/memalloc.h > @@ -52,6 +52,7 @@ struct snd_dma_device { > #else > #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ > #endif > +#define SNDRV_DMA_TYPE_DEV_WC 4 /* writecombine page */ > > /* > * info for buffer allocation > diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c > index 6915692..7468251 100644 > --- a/sound/core/memalloc.c > +++ b/sound/core/memalloc.c > @@ -124,8 +124,10 @@ void snd_free_pages(void *ptr, size_t size) > */ > > #ifdef CONFIG_HAS_DMA > -/* allocate the coherent DMA pages */ > -static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) > +static void *_snd_malloc_dev_pages(struct device *dev, size_t size, > + dma_addr_t *dma, > + void *(*alloc_func)(struct device *dev, size_t size, > + dma_addr_t *dma, gfp_t gfp_flags)) > { > int pg; > void *res; > @@ -138,16 +140,18 @@ static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *d > | __GFP_COMP /* compound page lets parts be mapped */ > | __GFP_NORETRY /* don't trigger OOM-killer */ > | __GFP_NOWARN; /* no stack trace print - this call is non-critical */ > - res = dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags); > + res = alloc_func(dev, PAGE_SIZE << pg, dma, gfp_flags); > if (res != NULL) > inc_snd_pages(pg); > > return res; > } > > -/* free the coherent DMA pages */ > -static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, > - dma_addr_t dma) > +static void _snd_free_dev_pages(struct device *dev, size_t size, void *ptr, > + dma_addr_t dma, > + void (*free_func)(struct device *dev, > + size_t size, void *vaddr, > + dma_addr_t dma_handle)) > { > int pg; > > @@ -155,8 +159,51 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, > return; > pg = get_order(size); > dec_snd_pages(pg); > - dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); > + free_func(dev, PAGE_SIZE << pg, ptr, dma); > +} > + > +static void *_dma_alloc_coherent(struct device *dev, size_t size, > + dma_addr_t *dma_handle, gfp_t flag) > +{ > + return dma_alloc_coherent(dev, size, dma_handle, flag); > +} > + > +static void _dma_free_coherent(struct device *dev, size_t size, > + void *vaddr, dma_addr_t dma_handle) > +{ > + dma_free_coherent(dev, size, vaddr, dma_handle); > +} > + > + > +/* allocate the coherent DMA pages */ > +static void *snd_malloc_dev_pages(struct device *dev, size_t size, > + dma_addr_t *dma) > +{ > + return _snd_malloc_dev_pages(dev, size, dma, _dma_alloc_coherent); > +} > + > +/* free the coherent DMA pages */ > +static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, > + dma_addr_t dma) > +{ > + _snd_free_dev_pages(dev, size, ptr, dma, _dma_free_coherent); > } > + > +#ifdef CONFIG_ARM > +/* allocate the writecombine DMA pages */ > +static void *snd_malloc_dev_wc_pages(struct device *dev, size_t size, > + dma_addr_t *dma) > +{ > + return _snd_malloc_dev_pages(dev, size, dma, dma_alloc_writecombine); > +} > + > +/* free the writecombine DMA pages */ > +static void snd_free_dev_wc_pages(struct device *dev, size_t size, void *ptr, > + dma_addr_t dma) > +{ > + _snd_free_dev_pages(dev, size, ptr, dma, dma_free_writecombine); > +} > +#endif > #endif /* CONFIG_HAS_DMA */ > > /* > @@ -200,6 +247,11 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, > case SNDRV_DMA_TYPE_DEV: > dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); > break; > +#ifdef CONFIG_ARM > + case SNDRV_DMA_TYPE_DEV_WC: > + dmab->area = snd_malloc_dev_wc_pages(device, size, &dmab->addr); > + break; > +#endif > #endif > #ifdef CONFIG_SND_DMA_SGBUF > case SNDRV_DMA_TYPE_DEV_SG: > @@ -272,6 +324,12 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) > case SNDRV_DMA_TYPE_DEV: > snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); > break; > +#ifdef CONFIG_ARM > + case SNDRV_DMA_TYPE_DEV_WC: > + snd_free_dev_wc_pages(dmab->dev.dev, dmab->bytes, dmab->area, > + dmab->addr); > + break; > +#endif > #endif > #ifdef CONFIG_SND_DMA_SGBUF > case SNDRV_DMA_TYPE_DEV_SG: > @@ -378,7 +436,7 @@ static int snd_mem_proc_read(struct seq_file *seq, void *offset) > long pages = snd_allocated_pages >> (PAGE_SHIFT-12); > struct snd_mem_list *mem; > int devno; > - static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" }; > + static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" , "WC"}; > > mutex_lock(&list_mutex); > seq_printf(seq, "pages : %li bytes (%li pages per %likB)\n", > diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c > index 53b5ada..28d949c 100644 > --- a/sound/core/pcm_native.c > +++ b/sound/core/pcm_native.c > @@ -3170,6 +3170,16 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, > struct vm_area_struct *area) > { > area->vm_flags |= VM_RESERVED; > + > +#ifdef CONFIG_ARM > + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_WC) > + return dma_mmap_writecombine(substream->dma_buffer.dev.dev, > + area, > + substream->runtime->dma_area, > + substream->runtime->dma_addr, > + area->vm_end - area->vm_start); > +#endif > + > #ifdef ARCH_HAS_DMA_MMAP_COHERENT > if (!substream->ops->page && > substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) > -- > 1.7.1.1 >

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[alsa-devel] 2.6.18 unknown symbol ksize error
by Patrick Shirkey 02 Jul '12

02 Jul '12

Hi, I have compiled the latest stable alsa drivers against 2.6.18 kernel with gcc-4.1. I am getting an error when I insert the sound card modules. snd_rawmidi: Unknown symbol ksize As far as I can tell 2.6.18 has support for ksize so I am not sure where this error is coming from. -- Patrick Shirkey Boost Hardware Ltd

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[alsa-devel] alsa-driver cross compile problem - upgrading ALSA
by Max Klein 02 Jul '12

02 Jul '12

I am running 2.6.27 kernel on arm board with ALSA driver version 1.0.17 (cat /proc/asound/version) and I would like to update ALSA driver version to current 1.0.25. I am trying to build alsa-drivers with "./configure --with-cross=arm-none-linux-gnueabi- --enable-dynamic-minors --with-debug=full --with-isapnp=no --with-oss=no --with-kernel=/data/src/kernel/linux-davinci --with-build=/data/src/kernel/linux-davinci --with-sequencer=no" Afterwards I compile it via "make ARCH=arm" but this fails we the following error: In file included from /data/VRM/davinci33/alsa-1.0.25/alsa-driver-1.0.25/acore/hrtimer.c:1: /data/VRM/davinci33/alsa-1.0.25/alsa-driver-1.0.25/include/adriver.h:753: error: static declaration of 'jiffies_to_msecs' follows non-static declaration include/linux/jiffies.h:286: error: previous declaration of 'jiffies_to_msecs' was here /data/VRM/davinci33/alsa-1.0.25/alsa-driver-1.0.25/include/adriver.h:772: error: static declaration of 'msecs_to_jiffies' follows non-static declaration include/linux/jiffies.h:288: error: previous declaration of 'msecs_to_jiffies' was here Can someone please help me how to get ALSA upgraded on my platform? Obviously I am either doing something wrong when configuring or I am on the absolutely wrong way? I am looking forward for any advice, thanks in advance! -- NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone! Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a

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[alsa-devel] [PATCH] mioa701 convert to snd_soc_register_card API
by Robert Jarzmik 01 Jul '12

01 Jul '12

The mioa701 board code is converted to the snd_soc_register_card() and snd_soc_unregister_card() APIs. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/soc/pxa/mioa701_wm9713.c | 33 ++++++++++++--------------------- 1 file changed, 12 insertions(+), 21 deletions(-) diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 9c585af..8687c1c 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -186,36 +186,27 @@ static struct snd_soc_card mioa701 = { .num_links = ARRAY_SIZE(mioa701_dai), }; -static struct platform_device *mioa701_snd_device; - -static int mioa701_wm9713_probe(struct platform_device *pdev) +static int __devinit mioa701_wm9713_probe(struct platform_device *pdev) { - int ret; + int rc; if (!machine_is_mioa701()) return -ENODEV; - dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" - "lead to overheating and possible destruction of your device." - "Do not use without a good knowledge of mio's board design!\n"); - - mioa701_snd_device = platform_device_alloc("soc-audio", -1); - if (!mioa701_snd_device) - return -ENOMEM; - - platform_set_drvdata(mioa701_snd_device, &mioa701); - - ret = platform_device_add(mioa701_snd_device); - if (!ret) - return 0; - - platform_device_put(mioa701_snd_device); - return ret; + mioa701.dev = &pdev->dev; + rc = snd_soc_register_card(&mioa701); + if (!rc) + dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" + "lead to overheating and possible destruction of your device." + " Do not use without a good knowledge of mio's board design!\n"); + return rc; } static int __devexit mioa701_wm9713_remove(struct platform_device *pdev) { - platform_device_unregister(mioa701_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); return 0; } -- 1.7.10

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[alsa-devel] [PATCH] ASoC: soc-dapm: Use ASYNC_DOMAIN_EXCLUSIVE
by Fabio Estevam 01 Jul '12

01 Jul '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> commit e895407d (async: introduce 'async_domain' type) changed the parameter types of async_schedule_domain() and async_synchronize_full_domain(). Use ASYNC_DOMAIN_EXCLUSIVE and fix the following build warnings: sound/soc/soc-dapm.c:1646:6: warning: passing argument 3 of 'async_schedule_domain' from incompatible pointer type sound/soc/soc-dapm.c:1647:2: warning: passing argument 1 of 'async_synchronize_full_domain' from incompatible pointer type sound/soc/soc-dapm.c:1660:6: warning: passing argument 3 of 'async_schedule_domain' from incompatible pointer type sound/soc/soc-dapm.c:1661:2: warning: passing argument 1 of 'async_synchronize_full_domain' from incompatible pointer type Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Build tested only. sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9670668..c93e832 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1572,7 +1572,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) struct snd_soc_dapm_context *d; LIST_HEAD(up_list); LIST_HEAD(down_list); - LIST_HEAD(async_domain); + ASYNC_DOMAIN_EXCLUSIVE(async_domain); enum snd_soc_bias_level bias; trace_snd_soc_dapm_start(card); -- 1.7.1

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Re: [alsa-devel] [pulseaudio-discuss] asoundrc, configure one virtual device for both input and output
by Tanu Kaskinen 01 Jul '12

01 Jul '12

(CC'ing alsa-devel, because I had problems with building alsa-plugins) On Sun, 2012-07-01 at 14:16 +0900, Amar Akshat wrote: > Help me here with two things, > > 1) How do I fix this and test, as in should I clone the alsa-project, > or download alsa code and fix and compile ? In case I have to clone > it, can you also provide me the clone instructions. git clone git://git.alsa-project.org/alsa-plugins.git alsa-plugins cd alsa-plugins ./gitcompile --enable-pulseaudio If the configure script complains about missing dependencies, you'll need to install them. After you have made your changes, I believe you should be able to test them by running "make && sudo make install". I haven't tried that myself. I tried the "./gitcompile --enable-pulseaudio" command myself, and had this problem: Making all in pulse make[2]: Entering directory `/home/tanu/dev/alsa-plugins/pulse' /bin/bash ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I.. -Wall -g -I/usr/include/alsa -D_REENTRANT -I/usr/local/include -D_GNU_SOURCE -g -O2 -MT ctl_pulse.lo -MD -MP -MF .deps/ctl_pulse.Tpo -c -o ctl_pulse.lo ctl_pulse.c libtool: compile: gcc -DHAVE_CONFIG_H -I. -I.. -Wall -g -I/usr/include/alsa -D_REENTRANT -I/usr/local/include -D_GNU_SOURCE -g -O2 -MT ctl_pulse.lo -MD -MP -MF .deps/ctl_pulse.Tpo -c ctl_pulse.c -fPIC -DPIC -o .libs/ctl_pulse.o mv -f .deps/ctl_pulse.Tpo .deps/ctl_pulse.Plo /bin/bash ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I.. -Wall -g -I/usr/include/alsa -D_REENTRANT -I/usr/local/include -D_GNU_SOURCE -g -O2 -MT pulse.lo -MD -MP -MF .deps/pulse.Tpo -c -o pulse.lo pulse.c libtool: compile: gcc -DHAVE_CONFIG_H -I. -I.. -Wall -g -I/usr/include/alsa -D_REENTRANT -I/usr/local/include -D_GNU_SOURCE -g -O2 -MT pulse.lo -MD -MP -MF .deps/pulse.Tpo -c pulse.c -fPIC -DPIC -o .libs/pulse.o mv -f .deps/pulse.Tpo .deps/pulse.Plo /bin/bash ../libtool --tag=CC --mode=link gcc -Wall -g -I/usr/include/alsa -D_REENTRANT -I/usr/local/include -D_GNU_SOURCE -g -O2 -module -avoid-version -export-dynamic -no-undefined -o libasound_module_ctl_pulse.la -rpath /usr/lib/alsa-lib ctl_pulse.lo pulse.lo -lasound -L/usr/local/lib -lpulse -lasound libtool: link: gcc -shared -fPIC -DPIC .libs/ctl_pulse.o .libs/pulse.o -L/usr/local/lib /usr/local/lib/libpulse.so -lasound -O2 -pthread -Wl,-soname -Wl,libasound_module_ctl_pulse.so -o .libs/libasound_module_ctl_pulse.so /bin/sed: can't read /usr/lib/x86_64-linux-gnu/libsndfile.la: No such file or directory libtool: link: `/usr/lib/x86_64-linux-gnu/libsndfile.la' is not a valid libtool archive make[2]: *** [libasound_module_ctl_pulse.la] Error 1 make[2]: Leaving directory `/home/tanu/dev/alsa-plugins/pulse' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/tanu/dev/alsa-plugins' make: *** [all] Error 2 Debian doesn't ship libsndfile.la, so I tried installing libsndfile from source. That didn't help at all. I eventually figured that since the alsa build system doesn't reference libsndfile at all, libpulse must somehow have a reference to /usr/lib/x86_64-linux-gnu/libsndfile.la (even though my system has never had that file installed). I rebuilt pulseaudio and installed it, and now it works (apparently because the rebuilt version references the version of libsndfile that was built from source, which includes libsndfile.la, instead of referencing the system version of libsndfile). Does anyone know if the build system of alsa-plugins could be improved so that it doesn't use any external .la files when linking? > 2) and after my modifications, my config in asoundrc should look like this ? > > pcm.pulse_dev1 { > type pulse > source alsa_input.pci-0000_00_1b.0.analog-stereo > sink alsa_output.pci-0000_00_1b.0.analog-mono > } Yes. -- Tanu

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Re: [alsa-devel] [PATCH] ASoC: tegra: use dmaengine based dma driver
by Stephen Warren 29 Jun '12

29 Jun '12

On 06/29/2012 05:34 AM, Laxman Dewangan wrote: > Use the dmaengine based Tegra APB DMA driver for > data transfer between SPI fifo and memory in > place of legacy Tegra APB DMA. > > Because generic soc-dmaengine-pcm uses the DMAs API > based on dmaengine, using the exported APIs provided > by this generic driver. > > The new driver is selected if legacy driver is not > selected and new dma driver is enabled through config > file. This works just fine with the existing non-dmaengine DMA driver enabled. However, I can't get it to work with dmaengine: > # aplay ~/abba-dq-48000-stereo.wav > [ 151.613476] tegra20-i2s tegra20-i2s.0: dmaengine pcm open failed with err -6 > [ 151.620557] tegra20-i2s tegra20-i2s.0: can't open platform tegra20-i2s.0: -6 > aplay: main:654: audio open error: No such device or address I do have the following in my local tree: 68a67b8 ARM: tegra: add device tree AUXDATA for APBDMA 0db7a96 ARM: tegra: dma: rename driver name for clock to "tegra-apbdma" I also fixed the compatible values in drivers/dma/tegra20-apb-dma.c so the driver would get instantiated, which it does; /sys/devices/tegra-apbdma/dma has a bunch of dmaengine channels in it. (Note: This is on Ventana, although I doubt that makes much difference) Is there something else I need to do to test this?

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Takashi Iwai 29 Jun '12

29 Jun '12

At Thu, 28 Jun 2012 19:06:07 +0530, Laxman Dewangan wrote: > > On Thursday 28 June 2012 06:58 PM, Takashi Iwai wrote: > > At Thu, 28 Jun 2012 18:34:02 +0530, > > Laxman Dewangan wrote: > >> On Thursday 28 June 2012 06:27 PM, Takashi Iwai wrote: > >>> At Thu, 28 Jun 2012 18:05:53 +0530, > >>> Laxman Dewangan wrote: > >>>> On Thursday 28 June 2012 05:58 PM, Mark Brown wrote: > >>>>> * PGP Signed by an unknown key > >>>>> > >>>>> On Thu, Jun 28, 2012 at 02:30:26PM +0200, Lars-Peter Clausen wrote: > >>>>>> On 06/28/2012 02:18 PM, Mark Brown wrote: > >>>>>>> Oh, dear - this means we don't actually have a generic API at all. Is > >>>>>>> there any effort being made to make this generally available? > >>>>>> Yes there is, and in fact in next/master there is generic > >>>>>> dma_mmap_writecombine. I'm not quite sure how it behaves on non ARM archs > >>>>>> though. But in theory it should fallback to a normal coherent mapping. Marek > >>>>>> did a lot of work on this, I've put him on Cc. > >>>> So can we put the function snd_pcm_lib_writecombine_mmap() in the > >>>> pcm_native.c and only export this api for ARM i.e. under macro #ifdef > >>>> CONFIG_ARM so that ARM based SOCs like Tegra/epa3xx/mxs/spear can > >>>> directly use that one? > >>>> Or, wait for this common API until all ARCH support it? > >>> I think it's fine to put it first in ALSA side with some ifdef. > >>> A similar trick is already found for snd_pcm_lib_mmap_iomem. > >>> See include/sound/pcm.h. > >>> > >>> But, actually it's a still question what if an architecture doesn't > >>> support the mmap of writecombine at all. The proposed patch doesn't > >>> allow you to know whether writecombine-mmap is possible or not on the > >>> running architecture until you really try to call it and fail. > >>> It's a missing piece, IMO. > >> Lars wanted to move the buffer allocation also to common place. > > Which common place? > > He wanted to refactor allocation part also at same time and so I guessed > that it will be in some common place and though about the sound/core. But the memory allocation rework is basically irrelevant from what's needed for pcm_mmap_writecombine() now. It'd need a lot more reworks than the simple addition of dma_mmap_writecombine() wrapper. > >> Then how about this? > >> create new file and header for snd-pcm-writecombine-buffer.c/.h and put > >> this in sound/core. > >> Select this file compilation through config SND_PCM_WRITECOMBINE_BUFFER > >> which will be select this config through sound/soc/xxx/Kconfig if they > >> want to use. > >> This will provide three apis: new, free and mmap. > > Way too much hustles than necessary... > Ooh no. :-( I had taken the idea from snd_dmaengine_pcm driver. > Bit I like to hear simple way which solves the purpose. As Lars suggested, a simple ifdef should suffice for now. With the upcoming generic dma_mmap_writecombine() stuff, it'd be even simpler in future. But, still we need to be careful about this. As mentioned, there is no flag to know the possibility of writecombine mmap beforehand. It'd be nice if we have either a compile-time or a run-time flag / function to check that. Then the driver can also expose the mmap capability to user-space depending on the flag. Takashi

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Re: [alsa-devel] [PATCH 0/3] ASoC: Move pcm writecombine dma buffer allocation to core
by Lars-Peter Clausen 29 Jun '12

29 Jun '12

On 06/29/2012 05:04 PM, Laxman Dewangan wrote: > [...] > > >> +/* allocate the coherent DMA pages */ >> +static void *snd_malloc_dev_pages(struct device *dev, size_t size, >> dma_addr_t *dma) >> +{ >> + return __snd_malloc_dev_pages(dev, size, dma, dma_alloc_coherent); > > This does not get compiled in ARM because dma_alloc_coherant is macro > defined as > > #define dma_alloc_coherent(d, s, h, f) dma_alloc_attrs(d, s, h, f, NULL) > > static inline void *dma_alloc_attrs(struct device *dev, size_t size, > dma_addr_t *dma_handle, gfp_t flag, > struct dma_attrs *attrs) > > > I fixed this by > static void *_dma_alloc_coherent(struct device *dev, size_t size, > dma_addr_t *dma_handle, gfp_t flag) > { > return dma_alloc_coherent(dev, size, dma_handle, flag); > } > > /* allocate the coherent DMA pages */ > static void *snd_malloc_dev_pages(struct device *dev, size_t size, > dma_addr_t *dma) > { > return _snd_malloc_dev_pages(dev, size, dma, _dma_alloc_coherent); > } I think all dma_alloc_* functions use dma_alloc_attrs internally these days. I think it might be a better idea just to use dma_alloc_attrs in __snd_malloc_dev_pages and just pass a dma_attrs struct instead of a function. Something like: static void *__snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma, struct dma_attrs *attrs) { .... res = dma_alloc_attrs(dev, PAGE_SIZE << pg, dma, gfp_flags, attrs); .... } static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) { return __snd_malloc_dev_pages(dev, size, dma, NULL); } static void *snd_malloc_dev_wc_pages(struct device *dev, size_t size, dma_addr_t *dma) { DEFINE_DMA_ATTRS(attrs); dma_set_attr(DMA_ATTR_WRITE_COMBINE, &attrs); return __snd_malloc_dev_pages(dev, size, dma, &attrs); } Same applies for freeing the memory again. - Lars

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Re: [alsa-devel] [PATCH 0/3] ASoC: Move pcm writecombine dma buffer allocation to core
by Takashi Iwai 29 Jun '12

29 Jun '12

At Fri, 29 Jun 2012 20:34:24 +0530, Laxman Dewangan wrote: > > Hi Takashi, > Thanks for sample code. It helps lot. > > On Friday 29 June 2012 05:43 PM, Takashi Iwai wrote: > > At Fri, 29 Jun 2012 15:53:15 +0530, > > Laxman Dewangan wrote: > >> Some of the ARM based soc allocate the writecombine dma buffer for > >> pcm substreams. They have the same codes for managing this buffer. > >> Moving this to the core/pcm files so that they can use that directly. > >> > >> Remove the code from Tegra PCM and use these new library function. > >> > >> This is enabled only for ARM specific and can be extended to other > >> architecture if they support the writecombine dma buffer. > >> > >> This patch is based on detail discussion on patch: > >> [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap > >> And suggestion from Lars and Takashi. > > Looking through your patch, I think an easier integration is just to > > add writecombine option to memalloc.c which calls > > dma_alloc_writecombine() instead of dma_alloc_coherent(). > > > > The addition for mmap is still an open question. Again, an easier > > option so far looks like just add the call of dma_alloc_writecombine() > > in snd_pcm_lib_default_mmap(). Alternatively, we can create an > > individual mmap pcm_ops as we discussed. > > > > Below is a totally untested patch, but you can imagine what I meant. > > > > After this change, replace with snd_pcm_lib_malloc_pages with > > SNDRV_DMA_TYPE_DEV_WC, and that's all you need in the driver side. > > We can not use the snd_pcm_lib_malloc_pages() in the pcm_new callback > because at this time the substream->runtime is not initialized and it > leads to the kernel crash. > > I used the apis as > > int snd_soc_pcm_new_wc_dma_buffer(struct snd_soc_pcm_runtime *rtd, > size_t max_bytes) > { > ::::::::::::: > substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; > if (substream) { > substream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_WC; > ret = snd_pcm_lib_malloc_pages(substream, max_bytes); > if (ret) > goto err; > } > :::::::: > } > > > int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t > size) > { > struct snd_pcm_runtime *runtime; > struct snd_dma_buffer *dmab = NULL; > > if (PCM_RUNTIME_CHECK(substream)) > return -EINVAL; > > if (snd_BUG_ON(substream->dma_buffer.dev.type == > SNDRV_DMA_TYPE_UNKNOWN)) > return -EINVAL; > > runtime = substream->runtime; > > if (runtime->dma_buffer_p) { > ---------------Kernel crash at this point ---------- > > So I used the snd_dma_alloc_pages() from the driver to allocate WC memory. snd_pcm_lib_malloc_pages() should be called only from hw_params. Prior to that, you need to set up the dam_buffer type by calling snd_pcm_lib_preallocate_pages() or snd_pcm_lib_preallocate_pages_for_all() at the time to create a PCM instance. The size can be 0 if not necessary to allocate there but later on demand. > > > +/* allocate the coherent DMA pages */ > > +static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) > > +{ > > + return __snd_malloc_dev_pages(dev, size, dma, dma_alloc_coherent); > > This does not get compiled in ARM because dma_alloc_coherant is macro > defined as > > #define dma_alloc_coherent(d, s, h, f) dma_alloc_attrs(d, s, h, f, NULL) Ah, OK. thanks, Takashi > static inline void *dma_alloc_attrs(struct device *dev, size_t size, > dma_addr_t *dma_handle, gfp_t flag, > struct dma_attrs *attrs) > > > I fixed this by > static void *_dma_alloc_coherent(struct device *dev, size_t size, > dma_addr_t *dma_handle, gfp_t flag) > { > return dma_alloc_coherent(dev, size, dma_handle, flag); > } > > /* allocate the coherent DMA pages */ > static void *snd_malloc_dev_pages(struct device *dev, size_t size, > dma_addr_t *dma) > { > return _snd_malloc_dev_pages(dev, size, dma, _dma_alloc_coherent); > } >

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Re: [alsa-devel] [PATCH 0/3] ASoC: Move pcm writecombine dma buffer allocation to core
by Takashi Iwai 29 Jun '12

29 Jun '12

At Fri, 29 Jun 2012 15:53:15 +0530, Laxman Dewangan wrote: > > Some of the ARM based soc allocate the writecombine dma buffer for > pcm substreams. They have the same codes for managing this buffer. > Moving this to the core/pcm files so that they can use that directly. > > Remove the code from Tegra PCM and use these new library function. > > This is enabled only for ARM specific and can be extended to other > architecture if they support the writecombine dma buffer. > > This patch is based on detail discussion on patch: > [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap > And suggestion from Lars and Takashi. Looking through your patch, I think an easier integration is just to add writecombine option to memalloc.c which calls dma_alloc_writecombine() instead of dma_alloc_coherent(). The addition for mmap is still an open question. Again, an easier option so far looks like just add the call of dma_alloc_writecombine() in snd_pcm_lib_default_mmap(). Alternatively, we can create an individual mmap pcm_ops as we discussed. Below is a totally untested patch, but you can imagine what I meant. After this change, replace with snd_pcm_lib_malloc_pages with SNDRV_DMA_TYPE_DEV_WC, and that's all you need in the driver side. Takashi --- diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index c425062..7840340 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,7 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#define SNDRV_DMA_TYPE_DEV_WC 4 /* writecombine pages */ /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 6915692..11108e8 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -124,8 +124,12 @@ void snd_free_pages(void *ptr, size_t size) */ #ifdef CONFIG_HAS_DMA -/* allocate the coherent DMA pages */ -static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) +static void *__snd_malloc_dev_pages(struct device *dev, size_t size, + dma_addr_t *dma, + void *(*func)(struct device *dev, + size_t size, + dma_addr_t *dma_handle, + gfp_t flag)) { int pg; void *res; @@ -138,16 +142,18 @@ static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *d | __GFP_COMP /* compound page lets parts be mapped */ | __GFP_NORETRY /* don't trigger OOM-killer */ | __GFP_NOWARN; /* no stack trace print - this call is non-critical */ - res = dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags); + res = func(dev, PAGE_SIZE << pg, dma, gfp_flags); if (res != NULL) inc_snd_pages(pg); return res; } -/* free the coherent DMA pages */ -static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, - dma_addr_t dma) +static void __snd_free_dev_pages(struct device *dev, size_t size, void *ptr, + dma_addr_t dma, + void (*func)(struct device *dev, + size_t size, void *vaddr, + dma_addr_t dma_handle)) { int pg; @@ -155,8 +161,36 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, return; pg = get_order(size); dec_snd_pages(pg); - dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); + func(dev, PAGE_SIZE << pg, ptr, dma); +} + +/* allocate the coherent DMA pages */ +static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma) +{ + return __snd_malloc_dev_pages(dev, size, dma, dma_alloc_coherent); +} + +/* free the coherent DMA pages */ +static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, + dma_addr_t dma) +{ + __snd_free_dev_pages(dev, ptr, dma, dma_free_coherent); } + +#ifdef CONFIG_ARM +/* allocate the writecombine DMA pages */ +static void *snd_malloc_dev_wc_pages(struct device *dev, size_t size, dma_addr_t *dma) +{ + return __snd_malloc_dev_pages(dev, size, dma, dma_alloc_writecombine); +} + +/* free the writecombine DMA pages */ +static void snd_free_dev_wc_pages(struct device *dev, size_t size, void *ptr, + dma_addr_t dma) +{ + __snd_free_dev_pages(dev, ptr, dma, dma_free_writecombine); +} +#endif /* CONFIG_ARM */ #endif /* CONFIG_HAS_DMA */ /* @@ -200,6 +234,11 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; +#ifdef CONFIG_ARM + case SNDRV_DMA_TYPE_DEV_WC: + dmab->area = snd_malloc_dev_wc_pages(device, size, &dmab->addr); + break; +#endif #endif #ifdef CONFIG_SND_DMA_SGBUF case SNDRV_DMA_TYPE_DEV_SG: @@ -272,6 +311,11 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; +#ifdef CONFIG_ARM + case SNDRV_DMA_TYPE_DEV_WC: + snd_free_dev_wc_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); + break; +#endif #endif #ifdef CONFIG_SND_DMA_SGBUF case SNDRV_DMA_TYPE_DEV_SG: @@ -378,7 +422,7 @@ static int snd_mem_proc_read(struct seq_file *seq, void *offset) long pages = snd_allocated_pages >> (PAGE_SHIFT-12); struct snd_mem_list *mem; int devno; - static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" }; + static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG", "WC" }; mutex_lock(&list_mutex); seq_printf(seq, "pages : %li bytes (%li pages per %likB)\n", diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 53b5ada..c499791 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3170,6 +3170,14 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { area->vm_flags |= VM_RESERVED; +#ifdef CONFIG_ARM + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_WC) + return dma_mmap_writecombine(substream->dma_buffer.dev.dev, + area, + substream->runtime->dma_area, + substream->runtime->dma_addr, + area->vm_end - area->vm_start); +#endif #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV)

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: add AGC settings
by Prchal Jiří 29 Jun '12

29 Jun '12

This patch adds AGC target level and times settings for TLV320AIC3106. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x.c | 25 +++++++++++++++++++++++++ 1 files changed, 25 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 50a7112..5335ebc 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -204,6 +204,10 @@ static const char *aic3x_right_hpcom_mux[] = static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; static const char *aic3x_adc_hpf[] = { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; +static const char *aic3x_agc_level[] = + { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; +static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; +static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; #define LDAC_ENUM 0 #define RDAC_ENUM 1 @@ -231,6 +235,21 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; +#define LAGC_LEV_ENUM 0 +#define RAGC_LEV_ENUM 1 +#define LAGC_ATT_ENUM 2 +#define RAGC_ATT_ENUM 3 +#define LAGC_DEC_ENUM 4 +#define RAGC_DEC_ENUM 5 +static const struct soc_enum aic3x_agc_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), +}; + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -355,6 +374,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + SOC_ENUM("Left AGC Target level", aic3x_agc_enum[LAGC_LEV_ENUM]), + SOC_ENUM("Right AGC Target level", aic3x_agc_enum[RAGC_LEV_ENUM]), + SOC_ENUM("Left AGC Attack time", aic3x_agc_enum[LAGC_ATT_ENUM]), + SOC_ENUM("Right AGC Attack time", aic3x_agc_enum[RAGC_ATT_ENUM]), + SOC_ENUM("Left AGC Decay time", aic3x_agc_enum[LAGC_DEC_ENUM]), + SOC_ENUM("Right AGC Decay time", aic3x_agc_enum[RAGC_DEC_ENUM]), /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, -- 1.7.7

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[alsa-devel] ASoC updates for 3.5
by Mark Brown 29 Jun '12

29 Jun '12

The following changes since commit 3419ae781f1592b3d367107db6500090495490cd: ASoC: tegra+wm8903: turn of mic detect when card is removed (2012-06-12 11:38:48 +0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.5 for you to fetch changes up to 5e531fc87875fc0bbfb3ca46a8a178674c4e9b7c: ASoC: tlv320aic3x: Fix codec pll configure bug (2012-06-27 12:46:25 +0100) ---------------------------------------------------------------- ASoC: Updates for 3.5 A couple of unexciting driver specific changes plus a fix for gain controls used as DAPM controls which isn't yet needed in mainline but is a definite bug. ---------------------------------------------------------------- Benoît Thébaudeau (1): ASoC: dapm: Fix snd_soc_dapm_put_volsw() connect Hebbar, Gururaja (1): ASoC: tlv320aic3x: Fix codec pll configure bug Mark Brown (1): ASoC: wm2200: Add missing BCLK rate sound/soc/codecs/tlv320aic3x.c | 4 +--- sound/soc/codecs/tlv320aic3x.h | 1 + sound/soc/codecs/wm2200.c | 1 + sound/soc/soc-dapm.c | 8 +------- 4 files changed, 4 insertions(+), 10 deletions(-)

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[alsa-devel] [PATCH] ASoC: spdif: Build separate RX and TX objects
by Mark Brown 29 Jun '12

29 Jun '12

Otherwise we fail to link when building as modules due to multiple init/exit functions. Reported-by: Fengguang Wu <wfg(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Makefile | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 62c3d4d..acf8088 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -45,7 +45,8 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o snd-soc-sn95031-objs := sn95031.o -snd-soc-spdif-objs := spdif_transciever.o spdif_receiver.o +snd-soc-spdif-tx-objs := spdif_transciever.o +snd-soc-spdif-rx-objs := spdif_receiver.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-stac9766-objs := stac9766.o @@ -159,7 +160,7 @@ obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o -obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o +obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o -- 1.7.10

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Re: [alsa-devel] [pulseaudio-discuss] [PATCH 2/4] alsa-mixer: Add Phantom Jack support
by Raymond Yau 29 Jun '12

29 Jun '12

2012/6/28 David Henningsson <david.henningsson(a)canonical.com>: > For kernel 3.6, "phantom jack" kctls have been added. They serve as > a marker that a particular port exist. They were made so we can detect > that there actually are speakers and internal mic on a laptop, even if > there are no other indications (volume controls etc). > > Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> > --- It seem that there will be numerous phantom ports created e.g. two internal mic phantom ports, spdif and spdif in, cd on the intel d975 desktop ./hda-emu codecs/stac9274d-intel-d975 CTRL: add: Front Headphone Jack:0 JACK created Front Headphone, type 1 JACK report Front Headphone, status 0 CTRL: add: Front Mic Jack:0 JACK created Front Mic, type 2 JACK report Front Mic, status 0 CTRL: add: Rear Mic Jack:0 JACK created Rear Mic, type 2 JACK report Rear Mic, status 0 CTRL: add: Internal Mic Phantom Jack:0 CTL Notify: Internal Mic Phantom Jack:0, mask=1 CTRL: add: Internal Mic Phantom Jack:1 CTL Notify: Internal Mic Phantom Jack:1, mask=1 CTRL: add: Line Jack:0 JACK created Line, type 32 JACK report Line, status 0 CTRL: add: CD Phantom Jack:0 CTL Notify: CD Phantom Jack:0, mask=1 CTRL: add: SPDIF Phantom Jack:0 CTL Notify: SPDIF Phantom Jack:0, mask=1 CTRL: add: SPDIF In Phantom Jack:0 CTL Notify: SPDIF In Phantom Jack:0, mask=1

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[alsa-devel] [PATCH] pcm: support for audio wallclock
by Pierre-Louis Bossart 28 Jun '12

28 Jun '12

add new snd_pcm_status_get_htstamp_wallclk() routine to query the audio timestamps read from the wallclock. Also add snd_pcm_hw_params_supports_audio_wallclock_ts() to query what the hardware supports. TODO: check protocol compatibility? Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/pcm.h | 2 ++ include/sound/asound.h | 7 +++++-- src/pcm/pcm.c | 31 +++++++++++++++++++++++++++++++ 3 files changed, 38 insertions(+), 2 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 4997557..542d9a8 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -550,6 +550,7 @@ int snd_pcm_hw_params_is_half_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_is_joint_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_sync_start(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *params); +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_get_rate_numden(const snd_pcm_hw_params_t *params, unsigned int *rate_num, unsigned int *rate_den); @@ -858,6 +859,7 @@ void snd_pcm_status_get_trigger_tstamp(const snd_pcm_status_t *obj, snd_timestam void snd_pcm_status_get_trigger_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); void snd_pcm_status_get_tstamp(const snd_pcm_status_t *obj, snd_timestamp_t *ptr); void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); +void snd_pcm_status_get_htstamp_wallclk(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); snd_pcm_sframes_t snd_pcm_status_get_delay(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail_max(const snd_pcm_status_t *obj); diff --git a/include/sound/asound.h b/include/sound/asound.h index 07c03fa..f0d23d6 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -160,7 +160,7 @@ enum { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11) typedef unsigned long sndrv_pcm_uframes_t; typedef long sndrv_pcm_sframes_t; @@ -285,6 +285,7 @@ enum sndrv_pcm_subformat { #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ enum sndrv_pcm_state { SNDRV_PCM_STATE_OPEN = 0, /* stream is open */ @@ -426,7 +427,8 @@ struct sndrv_pcm_status { sndrv_pcm_uframes_t avail_max; /* max frames available on hw since last status */ sndrv_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ int suspended_state; /* suspended stream state */ - unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* audio wall clock timestamp */ + unsigned char reserved[60-sizeof(struct timespec)]; /* must be filled with zero */ }; struct sndrv_pcm_mmap_status { @@ -435,6 +437,7 @@ struct sndrv_pcm_mmap_status { sndrv_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ int suspended_state; /* RO: suspended stream state */ + struct timespec audio_tstamp; /* audio wall clock timestamp */ }; struct sndrv_pcm_mmap_control { diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 18b43b3..49d7b99 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -3123,6 +3123,26 @@ int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *param } /** + * \brief Check if hardware supports audio wallclock timestamps + * \param params Configuration space + * \retval 0 Hardware doesn't support audio wallclock timestamps + * \retval 1 Hardware supports audio wallclock timestamps + * + * This function should only be called when the configuration space + * contains a single configuration. Call #snd_pcm_hw_params to choose + * a single configuration from the configuration space. + */ +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params) +{ + assert(params); + if (CHECK_SANITY(params->info == ~0U)) { + SNDMSG("invalid PCM info field"); + return 0; /* FIXME: should be a negative error? */ + } + return !!(params->info & SNDRV_PCM_INFO_HAS_WALL_CLOCK); +} + +/** * \brief Get rate exact info from a configuration space * \param params Configuration space * \param rate_num Pointer to returned rate numerator @@ -6213,6 +6233,17 @@ void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *p use_default_symbol_version(__snd_pcm_status_get_htstamp, snd_pcm_status_get_htstamp, ALSA_0.9.0rc8); /** + * \brief Get "now" hi-res audio wallclock timestamp from a PCM status container + * \param obj pointer to #snd_pcm_status_t + * \param ptr Pointer to returned timestamp + */ +void snd_pcm_status_get_htstamp_wallclk(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +{ + assert(obj && ptr); + *ptr = obj->audio_tstamp; +} + +/** * \brief Get delay from a PCM status container (see #snd_pcm_delay) * \return Delay in frames * -- 1.7.6.5

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Takashi Iwai 28 Jun '12

28 Jun '12

At Thu, 28 Jun 2012 19:30:57 +0530, Laxman Dewangan wrote: > > On Thursday 28 June 2012 07:21 PM, Takashi Iwai wrote: > > At Thu, 28 Jun 2012 19:06:07 +0530, > > Laxman Dewangan wrote: > > > >>>> Then how about this? > >>>> create new file and header for snd-pcm-writecombine-buffer.c/.h and put > >>>> this in sound/core. > >>>> Select this file compilation through config SND_PCM_WRITECOMBINE_BUFFER > >>>> which will be select this config through sound/soc/xxx/Kconfig if they > >>>> want to use. > >>>> This will provide three apis: new, free and mmap. > >>> Way too much hustles than necessary... > >> Ooh no. :-( I had taken the idea from snd_dmaengine_pcm driver. > >> Bit I like to hear simple way which solves the purpose. > > As Lars suggested, a simple ifdef should suffice for now. > > With the upcoming generic dma_mmap_writecombine() stuff, it'd be even > > simpler in future. > > > > But, still we need to be careful about this. As mentioned, there is > > no flag to know the possibility of writecombine mmap beforehand. > > It'd be nice if we have either a compile-time or a run-time flag / > > function to check that. Then the driver can also expose the mmap > > capability to user-space depending on the flag. > > Yaah, this seems really simple. Thanks for suggestion. > If I understand it fully, the new apis will be declare in sound/pcm.h, > implement it in pcm_native.c and use the config variable > CONFIG_SND_PCM_WRITECOMBINE_BUFFER for ifdef. Or this could be simply ifdef CONFIG_ARM. Since this ifdef will be unnecessary soon later, I don't think we need to give yet another kconfig. Of course, we should put a good comment around there for not keeping it for long time. Takashi

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Lars-Peter Clausen 28 Jun '12

28 Jun '12

On 06/28/2012 01:56 PM, Laxman Dewangan wrote: > On Thursday 28 June 2012 04:21 PM, Mark Brown wrote: >> * PGP Signed by an unknown key >> >> On Thu, Jun 28, 2012 at 12:31:26PM +0530, Laxman Dewangan wrote: >>> The pcm mmap for the substream dma buffer is done using the >>> dma_mmap_writecombine(). >>> Some of the chip sound driver like Spear, Tegra, Davinci, >>> ep93xx,snd_imx, snd_mxs, NUC900, OMAP, Samsung are doing this >>> mapping locally. >>> Add api in the snd dmaengine driver so that different client of >>> snd dmaengine can use this rather than implementing locally. >> OK, but I'd expect this to come along with one or more patches >> converting at least some of the users to the new API... > > This is effort towards moving the Tegra pcm driver for dmangine based dma > driver inplace of legacy dma driver for Tegra which have private API. > > Currently, following 4 pcm driver can use this API directly keeping that > they already using the generic snd dmaengine pcm driver. > ep93xx, > spear, > fsl, > mxs. > And Tegra will be next. > > Other driver which can use this API are davinci, OMAP, samsung but because > they are not using the snd_dmaengine_pcm driver's api as of now so not be > very much useful for them. Btw. all these drivers basically use the same code for buffer preallocation. Since how the buffer is mmaped is related to how it is allocated it might be worth to factor this out at the same time as well. - Lars

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Takashi Iwai 28 Jun '12

28 Jun '12

At Thu, 28 Jun 2012 18:34:02 +0530, Laxman Dewangan wrote: > > On Thursday 28 June 2012 06:27 PM, Takashi Iwai wrote: > > At Thu, 28 Jun 2012 18:05:53 +0530, > > Laxman Dewangan wrote: > >> On Thursday 28 June 2012 05:58 PM, Mark Brown wrote: > >>> * PGP Signed by an unknown key > >>> > >>> On Thu, Jun 28, 2012 at 02:30:26PM +0200, Lars-Peter Clausen wrote: > >>>> On 06/28/2012 02:18 PM, Mark Brown wrote: > >>>>> Oh, dear - this means we don't actually have a generic API at all. Is > >>>>> there any effort being made to make this generally available? > >>>> Yes there is, and in fact in next/master there is generic > >>>> dma_mmap_writecombine. I'm not quite sure how it behaves on non ARM archs > >>>> though. But in theory it should fallback to a normal coherent mapping. Marek > >>>> did a lot of work on this, I've put him on Cc. > >> So can we put the function snd_pcm_lib_writecombine_mmap() in the > >> pcm_native.c and only export this api for ARM i.e. under macro #ifdef > >> CONFIG_ARM so that ARM based SOCs like Tegra/epa3xx/mxs/spear can > >> directly use that one? > >> Or, wait for this common API until all ARCH support it? > > I think it's fine to put it first in ALSA side with some ifdef. > > A similar trick is already found for snd_pcm_lib_mmap_iomem. > > See include/sound/pcm.h. > > > > But, actually it's a still question what if an architecture doesn't > > support the mmap of writecombine at all. The proposed patch doesn't > > allow you to know whether writecombine-mmap is possible or not on the > > running architecture until you really try to call it and fail. > > It's a missing piece, IMO. > > Lars wanted to move the buffer allocation also to common place. Which common place? > Then how about this? > create new file and header for snd-pcm-writecombine-buffer.c/.h and put > this in sound/core. > Select this file compilation through config SND_PCM_WRITECOMBINE_BUFFER > which will be select this config through sound/soc/xxx/Kconfig if they > want to use. > This will provide three apis: new, free and mmap. Way too much hustles than necessary... Takashi

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Takashi Iwai 28 Jun '12

28 Jun '12

At Thu, 28 Jun 2012 18:05:53 +0530, Laxman Dewangan wrote: > > On Thursday 28 June 2012 05:58 PM, Mark Brown wrote: > > * PGP Signed by an unknown key > > > > On Thu, Jun 28, 2012 at 02:30:26PM +0200, Lars-Peter Clausen wrote: > >> On 06/28/2012 02:18 PM, Mark Brown wrote: > >>> Oh, dear - this means we don't actually have a generic API at all. Is > >>> there any effort being made to make this generally available? > >> Yes there is, and in fact in next/master there is generic > >> dma_mmap_writecombine. I'm not quite sure how it behaves on non ARM archs > >> though. But in theory it should fallback to a normal coherent mapping. Marek > >> did a lot of work on this, I've put him on Cc. > > So can we put the function snd_pcm_lib_writecombine_mmap() in the > pcm_native.c and only export this api for ARM i.e. under macro #ifdef > CONFIG_ARM so that ARM based SOCs like Tegra/epa3xx/mxs/spear can > directly use that one? > Or, wait for this common API until all ARCH support it? I think it's fine to put it first in ALSA side with some ifdef. A similar trick is already found for snd_pcm_lib_mmap_iomem. See include/sound/pcm.h. But, actually it's a still question what if an architecture doesn't support the mmap of writecombine at all. The proposed patch doesn't allow you to know whether writecombine-mmap is possible or not on the running architecture until you really try to call it and fail. It's a missing piece, IMO. Takashi

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Lars-Peter Clausen 28 Jun '12

28 Jun '12

On 06/28/2012 09:01 AM, Laxman Dewangan wrote: > The pcm mmap for the substream dma buffer is done using the > dma_mmap_writecombine(). > Some of the chip sound driver like Spear, Tegra, Davinci, > ep93xx,snd_imx, snd_mxs, NUC900, OMAP, Samsung are doing this > mapping locally. > Add api in the snd dmaengine driver so that different client of > snd dmaengine can use this rather than implementing locally. > This is not really related to the dmaengine pcm driver. It's more of a coincidence that all upstream drivers which use the dmaengine pcm driver also use write-combined memory. In my opinion it would be better to add this to the ALSA core. Now that there is a generic dma_mmap_writecombine it may make sense to integrate this with snd_pcm_lib_default_mmap. - Lars > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> > --- > include/sound/dmaengine_pcm.h | 2 ++ > sound/soc/soc-dmaengine-pcm.c | 17 +++++++++++++++++ > 2 files changed, 19 insertions(+), 0 deletions(-) > > diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h > index b877334..15c0d3c 100644 > --- a/include/sound/dmaengine_pcm.h > +++ b/include/sound/dmaengine_pcm.h > @@ -47,4 +47,6 @@ int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream); > > struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream); > > +int snd_dmaengine_pcm_mmap_writecombine(struct snd_pcm_substream *substream, > + struct vm_area_struct *vma); > #endif > diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c > index 5df529e..6535eb3 100644 > --- a/sound/soc/soc-dmaengine-pcm.c > +++ b/sound/soc/soc-dmaengine-pcm.c > @@ -21,6 +21,7 @@ > #include <linux/module.h> > #include <linux/init.h> > #include <linux/dmaengine.h> > +#include <linux/dma-mapping.h> > #include <linux/slab.h> > #include <sound/pcm.h> > #include <sound/pcm_params.h> > @@ -313,3 +314,19 @@ int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) > return 0; > } > EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close); > + > +/** > + * snd_dmaengine_pcm_mmap_writecombine - mmap stream dma buffer using > + * dma_mmap_writecombine(). > + * @substream: PCM substream > + * @vma: VM area. > + */ > +int snd_dmaengine_pcm_mmap_writecombine(struct snd_pcm_substream *substream, > + struct vm_area_struct *vma) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + > + return dma_mmap_writecombine(substream->pcm->card->dev, vma, > + runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); > +} > +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_mmap_writecombine);

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[alsa-devel] [PATCH] ASoC: dwc: Add missing __iomem annotations
by Mark Brown 28 Jun '12

28 Jun '12

Otherwise sparse gets very upset with us. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/dwc/designware_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index e667e2b..1bd042b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -71,12 +71,12 @@ struct dw_i2s_dev { int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); }; -static inline void i2s_write_reg(void *io_base, int reg, u32 val) +static inline void i2s_write_reg(void __iomem *io_base, int reg, u32 val) { writel(val, io_base + reg); } -static inline u32 i2s_read_reg(void *io_base, int reg) +static inline u32 i2s_read_reg(void __iomem *io_base, int reg) { return readl(io_base + reg); } -- 1.7.10

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Mark Brown 28 Jun '12

28 Jun '12

On Thu, Jun 28, 2012 at 05:26:58PM +0530, Laxman Dewangan wrote: > Other driver which can use this API are davinci, OMAP, samsung but > because they are not using the snd_dmaengine_pcm driver's api as of > now so not be very much useful for them. There's no reason why we can't factor out more common code - this is essentially what Lars-Peter was saying. > Should I send one patch per driver change or squash all 4 file > changes in one patch? Whatever will be easy or recommended? It doesn't really matter that much, but please take on board Lars-Peter's comments.

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Re: [alsa-devel] [PATCH] ASoC: snd_dmaengine: add common api for pcm_mmap
by Mark Brown 28 Jun '12

28 Jun '12

On Thu, Jun 28, 2012 at 12:31:26PM +0530, Laxman Dewangan wrote: > The pcm mmap for the substream dma buffer is done using the > dma_mmap_writecombine(). > Some of the chip sound driver like Spear, Tegra, Davinci, > ep93xx,snd_imx, snd_mxs, NUC900, OMAP, Samsung are doing this > mapping locally. > Add api in the snd dmaengine driver so that different client of > snd dmaengine can use this rather than implementing locally. OK, but I'd expect this to come along with one or more patches converting at least some of the users to the new API...

1 0

[alsa-devel] kcontrol get method
by raj raja 28 Jun '12

28 Jun '12

Hi, Generally get and set methods of Kcontrol called via mixer controls. I want to update variable value based on my kcontrol set method. I could not call get method, like as normal functions. So any way to call particular kcontrol get method in driver itself Thanks Raj

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[alsa-devel] ALS100: no interrupts in Linux but works fine in DOS
by Ondrej Zary 27 Jun '12

27 Jun '12

Hello, I have a Freetek PRO32PNP card - an ISA card with ALS100 chip and onboard Crystal wavetable MIDI. The card works fine as SB16 in DOS - without any initialization utility, PnP BIOS configures the card to 0x220, irq5, dma1, dma5. However it does not work in Linux. Driver loads fine (snd-als100, even tried old OSS sb) but when I try to play anything, the player hangs without producing any sound. /proc/interrupts show that no interrupts are generated. Mixer works fine. The only suspicious thing is "unknown tag 0xff" and "invalid checksum" in dmesg (this is from patched kernel that shows size of the unknown tag): [ 0.159650] isapnp: Scanning for PnP cards... [ 0.246683] pnp 01:01.00: dependent set 0 (preferred) io min 0x220 max 0x220 align 0 size 16 flags 0x2 [ 0.246889] pnp 01:01.00: dependent set 0 (preferred) irq 5 flags 0x1 [ 0.247072] pnp 01:01.00: dependent set 0 (preferred) dma 5 (bitmask 0x20) flags 0x72 [ 0.247253] pnp 01:01.00: dependent set 0 (preferred) dma 1 (bitmask 0x2) flags 0x68 [ 0.247872] pnp 01:01.00: dependent set 1 (acceptable) io min 0x220 max 0x280 align 32 size 16 flags 0x0 [ 0.248078] pnp 01:01.00: dependent set 1 (acceptable) irq 5 7 9 10 flags 0x1 [ 0.248265] pnp 01:01.00: dependent set 1 (acceptable) dma 5 6 7 (bitmask 0xe0) flags 0x72 [ 0.248450] pnp 01:01.00: dependent set 1 (acceptable) dma 0 1 3 (bitmask 0xb) flags 0x68 [ 0.248963] pnp 01:01.00: dependent set 2 (functional) io min 0x100 max 0x3f0 align 16 size 16 flags 0x0 [ 0.249213] pnp 01:01.00: dependent set 2 (functional) irq 5 7 9 10 11 12 15 flags 0xf [ 0.249404] pnp 01:01.00: dependent set 2 (functional) dma 0 1 3 5 6 7 (bitmask 0xeb) flags 0x79 [ 0.249996] pnp 01:01.01: independent io min 0x388 max 0x388 align 0 size 8 flags 0x2 [ 0.250535] pnp 01:01.02: independent io min 0x200 max 0x200 align 0 size 2 flags 0x2 [ 0.251171] pnp 01:01.03: dependent set 0 (preferred) io min 0x330 max 0x330 align 0 size 2 flags 0x2 [ 0.251365] pnp 01:01.03: dependent set 0 (preferred) irq 9 flags 0x1 [ 0.251984] pnp 01:01.03: dependent set 1 (acceptable) io min 0x100 max 0x3f0 align 16 size 2 flags 0x0 [ 0.252244] pnp 01:01.03: dependent set 1 (acceptable) irq 5 7 9 10 11 12 15 flags 0xf [ 0.256256] pnp 01:01.03: unknown tag 0xff (card 1) size=18041, ignored [ 4.527093] 01:01: invalid checksum 0xad [ 4.527257] Switching to clocksource tsc [ 4.528594] 01:01: card 'ALS100 Media Audio Controller' [ 4.528699] isapnp: 1 Plug & Play card detected total I've even tried to disable Linux pnp completely (noisapnp pnpbios=off) and load snd-sb16 module manually (modprobe snd-sb16 isapnp=0) with exactly the same result (everything seems to be OK but there's no sound and interrupts). Any ideas? -- Ondrej Zary

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[alsa-devel] periods_max initialization in snd_usb_hardware
by Amit Nagal 27 Jun '12

27 Jun '12

Hi , snd_usb_hardware is defined in sound/usb/pcm.c as : static struct snd_pcm_hardware snd_usb_hardware = { ..... .buffer_bytes_max = 1024 * 1024, .period_bytes_min = 64, .period_bytes_max = 512 * 1024, .periods_min = 2, .periods_max = 1024, }; here periods_max is intialized with value = 1024 . but going by buffer_bytes_max and period_bytes_min value defined above , periods_max = ( buffer_bytes_max / period_bytes_min ) = 16384 . why this upper cap of 1024 is maintained for periods_max even though from calculation it comes as 16384 ? Thanx & Regards Amit Nagal

2 1

[alsa-devel] how to update hw_ptr and appl_ptr
by Zhen Fu 27 Jun '12

27 Jun '12

Hi developer, I recently look the function snd_pcm_update_hw_ptr0 want to find how to update status->hw_ptr and control->appl_ptr. If you have to know this, can you tell me? Thank you very much. -- Best Regards, Zhen Fu

2 6

[alsa-devel] [PATCH 1/2] ALSA: HDA: Support single 3-pin jack without VREF on the actual pin
by David Henningsson 27 Jun '12

27 Jun '12

Some ASUS device has a single 3-pin jack that can either be a mic or a headphone, but the pin does not have VREF capabilities. We've been told by Realtek to instead enable VREF on pin 0x18 in that case. BugLink: https://bugs.launchpad.net/bugs/1018262 Tested-by: Chih-Hsyuan Ho <chih.ho(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 49 ++++++++++++++++++++++++++++------------- 1 file changed, 34 insertions(+), 15 deletions(-) Alsa info: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1018262/+attachment/32… diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5c81ee9..6174d4a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -303,6 +303,38 @@ static inline hda_nid_t get_capsrc(struct alc_spec *spec, int idx) static void call_update_outputs(struct hda_codec *codec); static void alc_inv_dmic_sync(struct hda_codec *codec, bool force); +/* for shared I/O, change the pin-control accordingly */ +static void update_shared_mic_hp(struct hda_codec *codec, bool set_as_mic) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + hda_nid_t pin = spec->autocfg.inputs[1].pin; + /* NOTE: this assumes that there are only two inputs, the + * first is the real internal mic and the second is HP/mic jack. + */ + + val = snd_hda_get_default_vref(codec, pin); + + /* This pin does not have vref caps - let's instead enable vref on pin 0x18 + instead, as suggested by Realtek */ + if (val == AC_PINCTL_VREF_HIZ) { + const hda_nid_t vref_pin = 0x18; + /* Sanity check pin 0x18 */ + if (get_wcaps_type(get_wcaps(codec, vref_pin)) == AC_WID_PIN && + get_defcfg_connect(snd_hda_codec_get_pincfg(codec, vref_pin)) == AC_JACK_PORT_NONE) { + unsigned int vref_val = snd_hda_get_default_vref(codec, vref_pin); + if (vref_val != AC_PINCTL_VREF_HIZ) + snd_hda_set_pin_ctl(codec, vref_pin, PIN_IN | (set_as_mic ? vref_val : 0)); + } + } + + val = set_as_mic ? val | PIN_IN : PIN_HP; + snd_hda_set_pin_ctl(codec, pin, val); + + spec->automute_speaker = !set_as_mic; + call_update_outputs(codec); +} + /* select the given imux item; either unmute exclusively or select the route */ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, unsigned int idx, bool force) @@ -329,21 +361,8 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, return 0; spec->cur_mux[adc_idx] = idx; - /* for shared I/O, change the pin-control accordingly */ - if (spec->shared_mic_hp) { - unsigned int val; - hda_nid_t pin = spec->autocfg.inputs[1].pin; - /* NOTE: this assumes that there are only two inputs, the - * first is the real internal mic and the second is HP jack. - */ - if (spec->cur_mux[adc_idx]) - val = snd_hda_get_default_vref(codec, pin) | PIN_IN; - else - val = PIN_HP; - snd_hda_set_pin_ctl(codec, pin, val); - spec->automute_speaker = !spec->cur_mux[adc_idx]; - call_update_outputs(codec); - } + if (spec->shared_mic_hp) + update_shared_mic_hp(codec, spec->cur_mux[adc_idx]); if (spec->dyn_adc_switch) { alc_dyn_adc_pcm_resetup(codec, idx); -- 1.7.9.5

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Re: [alsa-devel] emu10k1 driver can't allocate memory
by Takashi Iwai 27 Jun '12

27 Jun '12

At Sun, 20 May 2012 03:04:14 +0200, Mikael Magnusson wrote: > > Hi, (please CC me on replies, not subscribed) > > I recently upgraded to 3.3.6 from 2.6.33.15 (yeah, I know). Since then > I noticed sometimes sounds were sometimes missing, then I noticed > thousands of these in dmesg, > [37280.283484] max memory size is 0x7fffffff (addr = 0x1d5d70071038000)!! I guess you are enabling IOMMU? I guess it translates the address somehow in a way emu10k1 doesn't like. There are really little changes in emu10k1 driver itself, so if the kernel upgrade changes the behavior, it must be in the core part such like IOMMU handling. Try to boot without iommu and see whether it changes the behavior again. Takashi > [37280.283486] emu: failure page = 32 > I also checked asfxload -M output, right after boot it shows all the > synth memory free as you would expect, but after a while of usage, it > is close to 0 (around 50KiB or so). Trying to run a simple program > that writes to /dev/audio fails with > write(3</dev/audio>, > "\32\21\t\3\0\0\4\n\23\34&/6:;94,#\31\17\10\2\0\0\4\v\24\36(07"..., > 4096) = -1 ENOMEM (Cannot allocate memory) > You can also note the output from asfxload -M dropping rapidly while > the program is making sound before it fails. Native alsa apps seem to > fare slightly better as the memory decrease caused by them is actually > returned when they close the device again. > > Loading and unloading soundfonts seem to work as normal, the memory is > returned when the soundfont is unloaded afaict. (I have the patch > applied that makes it possible to load soundfonts (changes the DMA32 > zone to 2GB). I don't think this is the cause of my problem since the > given addresses are way way outside the 4GB range as well.) > > Note that even well before the reported memory is at 0, apps randomly > fail to open the sound output and are silent. Curiously espeak when > successful doesn't cause a decrease in available memory, but when it > fails, it causes about 100KiB drop. > > I can't (don't dare) bisect because this is my production machine, but > I can test patches, or check the major releases between 2.6.33 and > 3.3.6 if you have no idea what causes this. > > I'm not the only one with the problem either, > http://www.google.com/search?nfpr=1&tbs=li:1&q=emu%3A+failure+page+%3D+32+m… > > -- > Mikael Magnusson >

2 3

[alsa-devel] Configuring an audio path within ALSA
by rashmi narayan 27 Jun '12

27 Jun '12

Dear ALSA Folks, I am a newbie to ALSA .I am trying to configure a path within the driver to open up /power up the path within the codec from MIC-IN to LineOUT. How can I do this within ALSA. Please send me some pointers. Thank you, Rashmi

3 6

[alsa-devel] [PATCH V3 1/1] ASoC: tlv320aic3x: Fix codec pll configure bug
by Hebbar, Gururaja 27 Jun '12

27 Jun '12

In sound/soc/codecs/tlv320aic3x.c data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); In the above code, pll-p value is OR'ed with previous value without clearing it. Bug is not seen if pll-p value doesn't change across Sampling frequency. However on some platforms (like AM335x EVM-SK), pll-p may have different values across different sampling frequencies. In such case, above code configures the pll with a wrong value. Because of this bug, when a audio stream is played with pll value different from previous stream, audio is heard as differently(like its stretched). Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- chnages in V2: modify subject to indicate ASOC and codec as tlv320aic3x changes in V3: use snd_soc_update_bits() instead of read-mask-write sound/soc/codecs/tlv320aic3x.c | 4 +--- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 64d2a4f..e9b62b5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -935,9 +935,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, - data | (pll_p << PLLP_SHIFT)); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p); snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); snd_soc_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 6f097fb..08c7f66 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -166,6 +166,7 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLP_MASK 7 #define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 -- 1.7.1

2 1

[alsa-devel] [PATCH] ALSA: ASoC: Remove const attribute from lm49453's snd_soc_dai_driver
by Daniel Mack 27 Jun '12

27 Jun '12

Fixes the following build warning. CC sound/soc/codecs/lm49453.o sound/soc/codecs/lm49453.c: In function ‘lm49453_i2c_probe’: sound/soc/codecs/lm49453.c:1511:24: warning: passing argument 3 of ‘snd_soc_register_codec’ discards qualifiers from pointer target type include/sound/soc.h:333:5: note: expected ‘struct snd_soc_dai_driver *’ but argument is of type ‘const struct snd_soc_dai_driver *’ Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/lm49453.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 802b9f1..5bd1294 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1358,7 +1358,7 @@ static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { }; /* LM49453 dai structure. */ -static const struct snd_soc_dai_driver lm49453_dai[] = { +static struct snd_soc_dai_driver lm49453_dai[] = { { .name = "LM49453 Headset", .playback = { -- 1.7.10.2

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: add AGC, MIC_BIAS
by Prchal Jiří 27 Jun '12

27 Jun '12

This patch adds settings for TLV320AIC3106 of AGC, MIC_BIAS, De-emphasis. Also added default settings of output pop-up reduction, short circuit protection and mix both channels to mono_out to get maximum level. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.new @@ -194,6 +194,11 @@ static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; static const char *aic3x_adc_hpf[] = { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; +static const char *aic3x_agc_level[] = + { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; +static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; +static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; +static const char *aic3x_mic_bias[] = { "off", "2V", "2.5V", "AVDD" }; #define LDAC_ENUM 0 #define RDAC_ENUM 1 @@ -206,6 +211,13 @@ #define LINE2L_ENUM 8 #define LINE2R_ENUM 9 #define ADC_HPF_ENUM 10 +#define LAGC_LEV_ENUM 11 +#define RAGC_LEV_ENUM 12 +#define LAGC_ATT_ENUM 13 +#define RAGC_ATT_ENUM 14 +#define LAGC_DEC_ENUM 15 +#define RAGC_DEC_ENUM 16 +#define MIC_BIAS_ENUM 17 static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), @@ -219,6 +231,13 @@ SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(MICBIAS_CTRL, 6, 4, aic3x_mic_bias), }; /* @@ -345,6 +364,15 @@ * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + SOC_ENUM("Left AGC Target level", aic3x_enum[LAGC_LEV_ENUM]), + SOC_ENUM("Right AGC Target level", aic3x_enum[RAGC_LEV_ENUM]), + SOC_ENUM("Left AGC Attack time", aic3x_enum[LAGC_ATT_ENUM]), + SOC_ENUM("Right AGC Attack time", aic3x_enum[RAGC_ATT_ENUM]), + SOC_ENUM("Left AGC Decay time", aic3x_enum[LAGC_DEC_ENUM]), + SOC_ENUM("Right AGC Decay time", aic3x_enum[RAGC_DEC_ENUM]), + + /* De-emphasis */ + SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, @@ -352,6 +380,8 @@ SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), + + SOC_ENUM("Mic Bias", aic3x_enum[MIC_BIAS_ENUM]), }; /* @@ -1261,6 +1291,13 @@ snd_soc_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); + /* set to avoid artifacts on the audio output during power-on/off */ + snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_B, 0x80); /*ac-coupled*/ + snd_soc_write(codec, HPOUT_POP_REDUCTION, 0x4e); /* 10 + 4 ms, reference*/ + + /* short circuit protection */ + snd_soc_write(codec, HPRCOM_CFG, 0x04); + /* DAC default volume and mute */ snd_soc_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON); snd_soc_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON); @@ -1274,8 +1311,9 @@ snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* DAC to Mono Line Out default volume and route to Output mixer */ - snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + /* mix both channels with -6dB level */ + snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, (DEFAULT_VOL + 12) | ROUTE_ON); + snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, (DEFAULT_VOL + 12) | ROUTE_ON); /* unmute all outputs */ snd_soc_update_bits(codec, LLOPM_CTRL, UNMUTE, UNMUTE); @@ -1292,6 +1330,12 @@ /* By default route Line1 to ADC PGA mixer */ snd_soc_write(codec, LINE1L_2_LADC_CTRL, 0x0); snd_soc_write(codec, LINE1R_2_RADC_CTRL, 0x0); + + /* AGC to -10dB, 20 / 500ms, no clip stepping, noise gate -90dB, hysteresis 3dB*/ + snd_soc_write(codec, LAGC_CTRL_A, 0x2f); + snd_soc_write(codec, RAGC_CTRL_A, 0x2f); + snd_soc_write(codec, LAGC_CTRL_C, 0xbe); + snd_soc_write(codec, RAGC_CTRL_C, 0xbe); /* PGA to HP Bypass default volume, disconnect from Output Mixer */ snd_soc_write(codec, PGAL_2_HPLOUT_VOL, DEFAULT_VOL);

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[alsa-devel] [PATCH V2 0/9] Add ASoC drivers for SPEAr platform & I2S driver for Designware module
by Rajeev Kumar 27 Jun '12

27 Jun '12

Changes since V1: 1. Move Synopsys i2s designware controller from 'spear' folder to a new folder 'dwc' 2. Implemented regmap in sta529 codec. 3. In-corporated review comments received on V1. Rajeev Kumar (6): sound:asoc: Add support for STA529 Audio Codec sound:asoc: Add support for synopsys i2s controller as per ASoC framework. sound:asoc: Add support for SPEAr ASoC pcm layer. sound:asoc: Add support for SPEAr ASoC machine driver. sound:asoc: Add Kconfig and Makefile to support SPEAr audio driver sound:asoc: Update Kconfig and Makefile(sound/soc/) to support SPEAr audio Vipin Kumar (3): sound:asoc: Add support for spdif in Audio Codec sound:asoc:spdif_in: Add spdif IN support sound:asoc:spdif_out: Add spdif out support include/sound/designware_i2s.h | 69 ++++++ include/sound/spear_dma.h | 35 +++ include/sound/spear_spdif.h | 29 +++ sound/soc/Kconfig | 2 + sound/soc/Makefile | 2 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/spdif_receiver.c | 76 ++++++ sound/soc/codecs/sta529.c | 477 +++++++++++++++++++++++++++++++++++++ sound/soc/dwc/Kconfig | 8 + sound/soc/dwc/Makefile | 3 + sound/soc/dwc/designware_i2s.c | 454 +++++++++++++++++++++++++++++++++++ sound/soc/spear/Kconfig | 40 +++ sound/soc/spear/Makefile | 7 + sound/soc/spear/spdif_in.c | 297 +++++++++++++++++++++++ sound/soc/spear/spdif_in_regs.h | 60 +++++ sound/soc/spear/spdif_out.c | 397 ++++++++++++++++++++++++++++++ sound/soc/spear/spdif_out_regs.h | 79 ++++++ sound/soc/spear/spear_evb.c | 206 ++++++++++++++++ sound/soc/spear/spear_pcm.c | 214 +++++++++++++++++ 20 files changed, 2462 insertions(+), 1 deletions(-) create mode 100644 include/sound/designware_i2s.h create mode 100644 include/sound/spear_dma.h create mode 100644 include/sound/spear_spdif.h create mode 100644 sound/soc/codecs/spdif_receiver.c create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/dwc/Kconfig create mode 100644 sound/soc/dwc/Makefile create mode 100644 sound/soc/dwc/designware_i2s.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/spear/spdif_in.c create mode 100644 sound/soc/spear/spdif_in_regs.h create mode 100644 sound/soc/spear/spdif_out.c create mode 100644 sound/soc/spear/spdif_out_regs.h create mode 100644 sound/soc/spear/spear_evb.c create mode 100644 sound/soc/spear/spear_pcm.c

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[alsa-devel] [PATCH] ALSA: snd-usb-caiaq: initialize card pointer
by Daniel Mack 27 Jun '12

27 Jun '12

Fixes the following warning: CC [M] sound/usb/caiaq/device.o sound/usb/caiaq/device.c: In function ‘snd_probe’: sound/usb/caiaq/device.c:500:16: warning: ‘card’ may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/caiaq/device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 64aed43..7da0d0a 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -485,7 +485,7 @@ static int __devinit snd_probe(struct usb_interface *intf, const struct usb_device_id *id) { int ret; - struct snd_card *card; + struct snd_card *card = NULL; struct usb_device *device = interface_to_usbdev(intf); ret = create_card(device, intf, &card); -- 1.7.10.2

2 2

[alsa-devel] [Regression] No sound on HP Pavilion dv6, STAC92xx
by Luis Henriques 27 Jun '12

27 Jun '12

Hi, After commit b0791dda813c179e539b0fc1ecd3f5f30f2571e2 ("ALSA: hda/idt - Fix power-map for speaker-pins with some HP laptops") we have several bug reports about not having sound on HP Pavilion laptops. Here's a link to one of these bug reports: http://bugs.launchpad.net/bugs/1013183 After reverting the referred commit, everything seems to be working fine again. Any ideas/fixes? Cheers, -- Luis

2 9

[alsa-devel] [PATCH] ASoC: tlv320aic3x: add missing registers and bits
by Prchal Jiří 26 Jun '12

26 Jun '12

This patch adds missing registers and bits for TLV320AIC3106. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.new @@ -97,32 +97,20 @@ * There is no point in caching the reset register */ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { - 0x00, 0x00, 0x00, 0x10, /* 0 */ - 0x04, 0x00, 0x00, 0x00, /* 4 */ - 0x00, 0x00, 0x00, 0x01, /* 8 */ - 0x00, 0x00, 0x00, 0x80, /* 12 */ - 0x80, 0xff, 0xff, 0x78, /* 16 */ - 0x78, 0x78, 0x78, 0x78, /* 20 */ - 0x78, 0x00, 0x00, 0xfe, /* 24 */ - 0x00, 0x00, 0xfe, 0x00, /* 28 */ - 0x18, 0x18, 0x00, 0x00, /* 32 */ - 0x00, 0x00, 0x00, 0x00, /* 36 */ - 0x00, 0x00, 0x00, 0x80, /* 40 */ - 0x80, 0x00, 0x00, 0x00, /* 44 */ - 0x00, 0x00, 0x00, 0x04, /* 48 */ - 0x00, 0x00, 0x00, 0x00, /* 52 */ - 0x00, 0x00, 0x04, 0x00, /* 56 */ - 0x00, 0x00, 0x00, 0x00, /* 60 */ - 0x00, 0x04, 0x00, 0x00, /* 64 */ - 0x00, 0x00, 0x00, 0x00, /* 68 */ - 0x04, 0x00, 0x00, 0x00, /* 72 */ - 0x00, 0x00, 0x00, 0x00, /* 76 */ - 0x00, 0x00, 0x00, 0x00, /* 80 */ - 0x00, 0x00, 0x00, 0x00, /* 84 */ - 0x00, 0x00, 0x00, 0x00, /* 88 */ - 0x00, 0x00, 0x00, 0x00, /* 92 */ - 0x00, 0x00, 0x00, 0x00, /* 96 */ - 0x00, 0x00, 0x02, /* 100 */ + 0x00, 0x00, 0x00, 0x10, 0x04, 0x00, 0x00, 0x00, /* 0 */ + 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x80, /* 8 */ + 0x80, 0xff, 0xff, 0x78, 0x78, 0x78, 0x78, 0x78, /* 16 */ + 0x78, 0x00, 0x00, 0xfe, 0x00, 0x00, 0xfe, 0x00, /* 24 */ + 0x18, 0x18, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 32 */ + 0x00, 0x00, 0x00, 0x80, 0x80, 0x00, 0x00, 0x00, /* 40 */ + 0x00, 0x00, 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, /* 48 */ + 0x00, 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, /* 56 */ + 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 64 */ + 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 72 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 80 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 88 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 96 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 104 */ }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h.new @@ -13,7 +13,7 @@ #define _AIC3X_H /* AIC3X register space */ -#define AIC3X_CACHEREGNUM 103 +#define AIC3X_CACHEREGNUM 110 /* Page select register */ #define AIC3X_PAGE_SELECT 0 @@ -74,6 +74,8 @@ #define HPLCOM_CFG 37 /* Right High Power Output control registers */ #define HPRCOM_CFG 38 +/* High Power Output Stage Control Register */ +#define HPOUT_SC 40 /* DAC Output Switching control registers */ #define DAC_LINE_MUX 41 /* High Power Output Driver Pop Reduction registers */ @@ -148,6 +150,17 @@ #define AIC3X_GPIOB_REG 101 /* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 +/* New AGC registers */ +#define LAGCN_ATTACK 103 +#define LAGCN_DECAY 104 +#define RAGCN_ATTACK 105 +#define RAGCN_DECAY 106 +/* New Programmable ADC Digital Path and I2C Bus Condition Register */ +#define NEW_ADC_DIGITALPATH 107 +/* Passive Analog Signal Bypass Selection During Powerdown Register */ +#define PASSIVE_BYPASS 108 +/* DAC Quiescent Current Adjustment Register */ +#define DAC_ICC_ADJ 109 /* Page select register bits */ #define PAGE0_SELECT 0 @@ -163,6 +176,10 @@ #define DUAL_RATE_MODE ((1 << 5) | (1 << 6)) #define LDAC2LCH (0x1 << 3) #define RDAC2RCH (0x1 << 1) +#define LDAC2RCH (0x2 << 3) +#define RDAC2LCH (0x2 << 1) +#define LDAC2MONOMIX (0x3 << 3) +#define RDAC2MONOMIX (0x3 << 1) /* PLL registers bitfields */ #define PLLP_SHIFT 0

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Re: [alsa-devel] [PATCH V2] ASoC: tlv320aic3x: Fix codec pll configure bug
by Hebbar, Gururaja 26 Jun '12

26 Jun '12

On Tue, Jun 26, 2012 at 14:31:42, Prchal Jiří wrote: > Hi Gururaja, > shouldn't be better to use: > > snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p); > > instead of "read mask write" ? Sure will resend the patch. Thanks for the review > Even at this place you don't need to keep rest of register (PLL_Q and PLL_enable), so yust writing is OK. > > snd_soc_write(codec, AIC3X_PLL_PROGA_REG, pll_p << PLLP_SHIFT); > > Dne 26.6.2012 08:03, Hebbar, Gururaja napsal(a): > > In sound/soc/codecs/tlv320aic3x.c > > > > data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); > > snd_soc_write(codec, AIC3X_PLL_PROGA_REG, > > data | (pll_p<< PLLP_SHIFT)); > > > > In the above code, pll-p value is OR'ed with previous value without > > clearing it. Bug is not seen if pll-p value doesn't change across > > Sampling frequency. > > > > However on some platforms (like AM335x EVM-SK), pll-p may have different > > values across different sampling frequencies. In such case, above code > > configures the pll with a wrong value. > > Because of this bug, when a audio stream is played with pll value > > different from previous stream, audio is heard as differently. > > > > Signed-off-by: Hebbar, Gururaja<gururaja.hebbar(a)ti.com> > > --- > > chnages in V2: > > modify subject to indicate ASOC and codec as tlv320aic3x > > > > sound/soc/codecs/tlv320aic3x.c | 2 +- > > sound/soc/codecs/tlv320aic3x.h | 1 + > > 2 files changed, 2 insertions(+), 1 deletions(-) > > > > diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c > > index d0dbac1..8c1977b 100644 > > --- a/sound/soc/codecs/tlv320aic3x.c > > +++ b/sound/soc/codecs/tlv320aic3x.c > > @@ -963,7 +963,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, > > } > > > > found: > > - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); > > + data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG)& ~PLLP_MASK; > > snd_soc_write(codec, AIC3X_PLL_PROGA_REG, > > data | (pll_p<< PLLP_SHIFT)); > > snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, > > diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h > > index 06a1978..16d9999 100644 > > --- a/sound/soc/codecs/tlv320aic3x.h > > +++ b/sound/soc/codecs/tlv320aic3x.h > > @@ -166,6 +166,7 @@ > > > > /* PLL registers bitfields */ > > #define PLLP_SHIFT 0 > > +#define PLLP_MASK 7 > > #define PLLQ_SHIFT 3 > > #define PLLR_SHIFT 0 > > #define PLLJ_SHIFT 2 > > Regards, Gururaja

1 0

[alsa-devel] [PATCH V2] ASoC: tlv320aic3x: Fix codec pll configure bug
by Hebbar, Gururaja 26 Jun '12

26 Jun '12

In sound/soc/codecs/tlv320aic3x.c data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); In the above code, pll-p value is OR'ed with previous value without clearing it. Bug is not seen if pll-p value doesn't change across Sampling frequency. However on some platforms (like AM335x EVM-SK), pll-p may have different values across different sampling frequencies. In such case, above code configures the pll with a wrong value. Because of this bug, when a audio stream is played with pll value different from previous stream, audio is heard as differently. Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- chnages in V2: modify subject to indicate ASOC and codec as tlv320aic3x sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d0dbac1..8c1977b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -963,7 +963,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG) & ~PLLP_MASK; snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 06a1978..16d9999 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -166,6 +166,7 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLP_MASK 7 #define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 -- 1.7.1

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: add input clock selection
by Prchal Jiří 26 Jun '12

26 Jun '12

This patch adds input selection of main codec clock - from what pin. Tested with TLV320AIC3106 on BCLK and MCLK. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.orig +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c @@ -972,6 +972,10 @@ struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + /* set clock on MCLK or GPIO2 or BCLK */ + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK, clk_id); + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, CLKDIV_IN_MASK, clk_id); + aic3x->sysclk = freq; return 0; } --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h.orig +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h @@ -178,6 +178,13 @@ #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 +#define PLLCLK_IN_MASK 0x30 +#define CLKDIV_IN_MASK 0xc0 +/* clock in source */ +#define CLKIN_MCLK 0 +#define CLKIN_GPIO2 1 +#define CLKIN_BCLK 2 + /* Software reset register bits */ #define SOFT_RESET 0x80

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[alsa-devel] [PATCH] ASoC: wm8994: Don't suspend accessory detection
by Mark Brown 26 Jun '12

26 Jun '12

Leave it up to the machine driver to disable accessory detection if desired, the common pattern is to have accessory detection be a wake source. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 37 ------------------------------------- 1 file changed, 37 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6474bae..a4aafdb 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2967,23 +2967,8 @@ static struct snd_soc_dai_driver wm8994_dai[] = { static int wm8994_codec_suspend(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->wm8994; int i, ret; - switch (control->type) { - case WM8994: - snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); - break; - case WM1811: - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, 0); - /* Fall through */ - case WM8958: - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, - WM8958_MICD_ENA, 0); - break; - } - for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], sizeof(struct wm8994_fll_config)); @@ -3033,28 +3018,6 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec) i + 1, ret); } - switch (control->type) { - case WM8994: - if (wm8994->micdet[0].jack || wm8994->micdet[1].jack) - snd_soc_update_bits(codec, WM8994_MICBIAS, - WM8994_MICD_ENA, WM8994_MICD_ENA); - break; - case WM1811: - if (wm8994->jackdet && wm8994->jack_cb) { - /* Restart from idle */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, - WM1811_JACKDET_MODE_JACK); - break; - } - break; - case WM8958: - if (wm8994->jack_cb) - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, - WM8958_MICD_ENA, WM8958_MICD_ENA); - break; - } - return 0; } #else -- 1.7.10

1 0

[alsa-devel] [PATCH] davinci:mcasp: Fix codec pll configure bug
by Hebbar, Gururaja 26 Jun '12

26 Jun '12

In sound/soc/codecs/tlv320aic3x.c data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); In the above code, pll-p value is OR'ed with previous value without clearing it. Bug is not seen if pll-p value doesn't change across Sampling frequency. However on some platforms (like AM335x EVM-SK), pll-p may have different values across different sampling frequencies. In such case, above code configures the pll with a wrong value. Because of this bug, when a audio stream is played with pll value different from previous stream, audio is heard as stretched. Fix the same by clearing the mask and then writing the value. Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d0dbac1..8c1977b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -963,7 +963,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG) & ~PLLP_MASK; snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 06a1978..16d9999 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -166,6 +166,7 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLP_MASK 7 #define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 -- 1.7.1

1 1

[alsa-devel] [PATCH] ALSA: hda - Add Realtek ALC280 codec support
by David Henningsson 26 Jun '12

26 Jun '12

This chip looks very similar to ALC269 and ALC27* variants. The bug reporter has verified that sound was working after this patch had been applied. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/1017017 Tested-by: Richard Crossley <richardcrossley(a)o2.co.uk> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) Alsa info available at: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1017017/+attachment/32… diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 41475ae..a5534b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6800,6 +6800,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 }, { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 }, { .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 }, + { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, -- 1.7.9.5

2 1

Re: [alsa-devel] Using UCM with PulseAudio
by Arun Raghavan 25 Jun '12

25 Jun '12

On Thu, 2012-06-21 at 16:30 +0800, Feng Wei wrote: > 2012/6/20 Arun Raghavan <arun.raghavan(a)collabora.co.uk>: > > Liam, > > Thanks for clearing things up. > > > > On Fri, 2012-06-15 at 18:08 +0100, Liam Girdwood wrote: > > [...] > >> > The first problem is mutual exclusivity of verbs. From what I can > >> > understand, verbs are intended to be mutually exclusive -- if you have a > >> > HiFi verb and a VoiceCall verb, only one may be used at a time. We have > >> > mapped verbs to card profiles, which offer the same guarantee. However, > >> > on Android (which is a fair example of the kind of audio policy we might > >> > want), the HiFi verb PCMs maybe used while the VoiceCall PCMs are open. > >> > This is done, for example, to play an end-of-call tone from the CPU > >> > while the modem PCMs are still held open. Is there some way to do this > >> > with UCM? > >> > > >> > >> A verb is not tied to any specific PCM device here. It's intended to be > >> the highest level of audio use case where it can configure any audio > >> resource (including multiple cards) to enable the use case action. > >> > >> So for OMAP4 we would have the HiFi verb for all use case where we are > >> playing or capturing HiFi quality and the voice call verb where we are > >> making a telephone voice call. > >> > >> UCM provides the "modifier" to allow ad-hoc modifications to the audio > >> use case like above where we want to play an end of call tone. In this > >> case the verb is still VoiceCall, but PA would enable the "PlayTone" > >> modifer to play the tone (UCM can also tell Pulseaudio the sink PCM and > >> volume control for the tone data). > > > > Just as an observation, I think the Android HAL just uses PCM 0 and not > > the PCM 3, but in the PA case, I'll do it the way you describe since > > that makes more sense. > > > >> > The second problem is having separate PCMs for modifiers. In the OMAP4 > >> > profile, ringtone playback is exposed via a PlayTone modifier which > >> > corresponds to a separate PCM from regular HiFi playback. In the UCM-PA > >> > mapping we decided on, modifiers were implemented as device intended > >> > roles on a sink, so that when a stream with that role came in, we could > >> > enable the modifier, and disable it when such a stream ends. However, > >> > this doesn't account for switching the PCM on which playback is > >> > occurring. Should we be creating a separate sink for such modifiers > >> > (with lower priority, so they're not routed to unless there's a stream > >> > with the required role coming in)? Or should we be reopening the PCM for > >> > this? > >> > >> The intention for modifiers that use separate ALSA PCM sinks/sources > >> from the verb is to keep the main stream on the verb PCM source/sink and > >> the modifier stream will use the modifier PCM sink/source (this can be > >> the same PCM for some hardware). > >> > >> e.g. MP3 will be played to pcm 0 sink and ringtone to pcm 1 sink. The HW > >> will then mix both streams before they are rendered. > > > > Okay, so what this means is that we need to extend the current UCM work > > to create a second, lower priority sink for each modifier that has a > > distinct PlaybackPCM, and let the role-based routing pick that in cases > > where it makes sense. It's good that this doesn't change how we've > > mapped existing concepts -- just adds to it. > Sounds fine to me. The only problem is can we always open the modifier > pcm when modifier is not enabled. If not, I prefer to create/destroy > the sink by events. The alternative is to always add the sink and enable/disable the modifier in the I/O thread since we will get a message when a sink-input is added/removed. I'm just hacking this bit up, and will post patches if it all works out. There is another question here -- whether the modifier sink should have any ports or not. If I understand this correctly, we could let the actual routing be handled by the "main" sink's port and have the modifier sinks not touch that at all by not having any ports. In the ideal world though, it would be nice to have the ability to mark ports as being "dependent" somehow, so that port changes on the main sink would reflect in all others. This would be particularly useful for sinks and sources that have correlated paths like they do in the OMAP4 case. -- Arun

1 1

Re: [alsa-devel] Possible message repeat
by Clemens Ladisch 25 Jun '12

25 Jun '12

Brent Weatherall wrote: > our goal is to specify a special XU that WOULD require vendors > supporting it to modify their firmware. We fully expect a vendor to > modify their firmware to support our specification. That being said, > is there a simple way to use ALSA to communicate with a vendor > specific XU as long as the software understands the XU? For example, > in the video4linux2 world, we can specify a UVC XU with a set of > controls for special processing, then vendors that want to support it > modify their firmware to support it. Then anyone can use our custom > user-space drivers to easily talk to the video device. V4L2 has > controls made for using XUs in this manner. The audio class specification does not define any mechanism for a device to specify what those controls would be. And because nobody has ever needed to use XUs so far, the current ALSA driver has no other way to create such controls. If you insist on not changing ALSA, you must bypass it. Regards, Clemens

2 1

[alsa-devel] wm8994 trivial merge fix
by Takashi Iwai 25 Jun '12

25 Jun '12

FYI, the patch below is what I applied from Jaroslav's tree. I simply picked it up directly into my tree (fix/asoc and for-linus branches) because of triviality. Takashi --- From 6cab3e1e65bfe070f09bb091eeda182b171d5929 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela <perex(a)perex.cz> Date: Mon, 28 May 2012 11:14:33 +0200 Subject: [PATCH] ASoC: wm8994: remove duplicate code It seems that the code duplication was added at a merge operation. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/wm8994.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index aa8c98b..1436b6c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -727,9 +727,6 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) if (!wm8994->jackdet || !wm8994->jack_cb) return; - if (!wm8994->jackdet || !wm8994->jack_cb) - return; - if (wm8994->active_refcount) mode = WM1811_JACKDET_MODE_AUDIO; -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: dwc: Bodge around continuing absence of clock API stubs
by Mark Brown 25 Jun '12

25 Jun '12

The patches for stubbing out the generic clock API still haven't been applied so we need to either add ifdefs here or add a dependency until someone decides to actually apply them. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/dwc/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index 93e9fc3..e334900 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -1,5 +1,6 @@ config SND_DESIGNWARE_I2S tristate "Synopsys I2S Device Driver" + depends on CLKDEV_LOOKUP help Say Y or M if you want to add support for I2S driver for Synopsys desigwnware I2S device. The device supports upto -- 1.7.10

1 0

[alsa-devel] [PATCH] ASoC: dmaengine_pcm: fix typo in comment
by Uwe Kleine-König 25 Jun '12

25 Jun '12

Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/soc-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 4756952..5647449 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -243,7 +243,7 @@ static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd * Note that this function will use private_data field of the substream's * runtime. So it is not availabe to your pcm driver implementation. If you need * to keep additional data attached to a substream use - * snd_dmaeinge_pcm_{set,get}_data. + * snd_dmaengine_pcm_{set,get}_data. */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) -- 1.7.10

2 1

[alsa-devel] [RFC PATCH] Inverted internal mic
by David Henningsson 25 Jun '12

25 Jun '12

Hi, One of the more common problems on laptop machines, is that the internal mic provides a stereo signal but with one channel phase inverted, or differential output. This means problems for applications recording two channels but later merging them into one, leaving them with zero or near-zero output. There are various ways we can work around this in both the kernel, alsa-lib and PulseAudio layers. It's a matter of picking the right one. I'm leaning towards trying to fix it in the kernel's codec drivers, because 1) we already have quirking infrastructure there 2) we already have some working quirks already in that layer 3) it would benefit other sound applications that use ALSA directly. The downside to that is really that we're silencing out one channel for everyone, leading to no application being able to use both channels, even if they would implement some kind of "auto-detect-and-reverse-one-channel" functionality. For the most part, this has been some Acer Aspire machines running Realtek ALC268 / ALC269 / ALC271X / ALC272X, and for two of these we have proc coefs we can set, but for the other two these proc coefs, if they exist, are unknown to us. Recently I came across a Conexant as well, and I decided to write a patch for it, that would take the approach that the internal mic is forced mono on the kcontrol, and make sure the right channel is always muted. The patch is verified by the reporter to fix this problem. It could use some perfection though - it would make sense to to the same to the internal mic boost as well, and the strcmp('Internal Mic') call could maybe be turned into something more elegant. But before going ahead with that, I'd like to hear your opinion on the matter, if you agree that this is a good approach to the problem? References: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/903853/+attachme… https://bugs.launchpad.net/bugs/903853 -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

5 38

[alsa-devel] Exclusive access to capture device
by Guy Rutenberg 24 Jun '12

24 Jun '12

Hi, Is there a way to open a capture device exclusively? E.g. I want to capture audio and be sure no other application can capture from that device at the same time. I've went through the API and searched the mailing list and haven't found a way to do so. Thanks a lot, Guy ---- http://www.guyrutenberg.com/

2 3

[alsa-devel] [PATCH 1/2] ASoC: Add shared code for Wolfson Arizona class devices
by Mark Brown 23 Jun '12

23 Jun '12

The Wolfson Arizona series of audio hub CODECs can share a large amount of their driver code as the result of a common register map. This patch adds some of this core support, providing a basis for the initial WM5102 audio driver. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 781 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 149 +++++++++ 4 files changed, 937 insertions(+) create mode 100644 sound/soc/codecs/arizona.c create mode 100644 sound/soc/codecs/arizona.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 43f5240..2ae8082 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -129,6 +129,11 @@ config SND_SOC_ALL_CODECS config SND_SOC_88PM860X tristate +config SND_SOC_ARIZONA + tristate + default y if SND_SOC_WM5102=y + default m if SND_SOC_WM5102=m + config SND_SOC_WM_HUBS tristate default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3d30654..3005ea6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -14,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l52-objs := cs42l52.o @@ -128,6 +129,7 @@ obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o +obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c new file mode 100644 index 0000000..3b5730b --- /dev/null +++ b/sound/soc/codecs/arizona.c @@ -0,0 +1,781 @@ +/* + * arizona.c - Wolfson Arizona class device shared support + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/gcd.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" + +#define ARIZONA_AIF_BCLK_CTRL 0x00 +#define ARIZONA_AIF_TX_PIN_CTRL 0x01 +#define ARIZONA_AIF_RX_PIN_CTRL 0x02 +#define ARIZONA_AIF_RATE_CTRL 0x03 +#define ARIZONA_AIF_FORMAT 0x04 +#define ARIZONA_AIF_TX_BCLK_RATE 0x05 +#define ARIZONA_AIF_RX_BCLK_RATE 0x06 +#define ARIZONA_AIF_FRAME_CTRL_1 0x07 +#define ARIZONA_AIF_FRAME_CTRL_2 0x08 +#define ARIZONA_AIF_FRAME_CTRL_3 0x09 +#define ARIZONA_AIF_FRAME_CTRL_4 0x0A +#define ARIZONA_AIF_FRAME_CTRL_5 0x0B +#define ARIZONA_AIF_FRAME_CTRL_6 0x0C +#define ARIZONA_AIF_FRAME_CTRL_7 0x0D +#define ARIZONA_AIF_FRAME_CTRL_8 0x0E +#define ARIZONA_AIF_FRAME_CTRL_9 0x0F +#define ARIZONA_AIF_FRAME_CTRL_10 0x10 +#define ARIZONA_AIF_FRAME_CTRL_11 0x11 +#define ARIZONA_AIF_FRAME_CTRL_12 0x12 +#define ARIZONA_AIF_FRAME_CTRL_13 0x13 +#define ARIZONA_AIF_FRAME_CTRL_14 0x14 +#define ARIZONA_AIF_FRAME_CTRL_15 0x15 +#define ARIZONA_AIF_FRAME_CTRL_16 0x16 +#define ARIZONA_AIF_FRAME_CTRL_17 0x17 +#define ARIZONA_AIF_FRAME_CTRL_18 0x18 +#define ARIZONA_AIF_TX_ENABLES 0x19 +#define ARIZONA_AIF_RX_ENABLES 0x1A +#define ARIZONA_AIF_FORCE_WRITE 0x1B + +#define arizona_fll_err(_fll, fmt, ...) \ + dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) +#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__) + +#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__) + +const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { + "None", + "Tone Generator 1", + "Tone Generator 2", + "Haptics", + "AEC", + "Mic Mute Mixer", + "Noise Generator", + "IN1L", + "IN1R", + "IN2L", + "IN2R", + "IN3L", + "IN3R", + "AIF1RX1", + "AIF1RX2", + "AIF1RX3", + "AIF1RX4", + "AIF1RX5", + "AIF1RX6", + "AIF1RX7", + "AIF1RX8", + "AIF2RX1", + "AIF2RX2", + "AIF3RX1", + "AIF3RX2", + "SLIMRX1", + "SLIMRX2", + "SLIMRX3", + "SLIMRX4", + "SLIMRX5", + "SLIMRX6", + "SLIMRX7", + "SLIMRX8", + "EQ1", + "EQ2", + "EQ3", + "EQ4", + "DRC1L", + "DRC1R", + "DRC2L", + "DRC2R", + "LHPF1", + "LHPF2", + "LHPF3", + "LHPF4", + "DSP1.1", + "DSP1.2", + "DSP1.3", + "DSP1.4", + "DSP1.5", + "DSP1.6", + "ASRC1L", + "ASRC1R", + "ASRC2L", + "ASRC2R", +}; +EXPORT_SYMBOL_GPL(arizona_mixer_texts); + +int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { + 0x00, /* None */ + 0x04, /* Tone */ + 0x05, + 0x06, /* Haptics */ + 0x08, /* AEC */ + 0x0c, /* Noise mixer */ + 0x0d, /* Comfort noise */ + 0x10, /* IN1L */ + 0x11, + 0x12, + 0x13, + 0x14, + 0x15, + 0x20, /* AIF1RX1 */ + 0x21, + 0x22, + 0x23, + 0x24, + 0x25, + 0x26, + 0x27, + 0x28, /* AIF2RX1 */ + 0x29, + 0x30, /* AIF3RX1 */ + 0x31, + 0x38, /* SLIMRX1 */ + 0x39, + 0x3a, + 0x3b, + 0x3c, + 0x3d, + 0x3e, + 0x3f, + 0x50, /* EQ1 */ + 0x51, + 0x52, + 0x53, + 0x58, /* DRC1L */ + 0x59, + 0x5a, + 0x5b, + 0x60, /* LHPF1 */ + 0x61, + 0x62, + 0x63, + 0x68, /* DSP1.1 */ + 0x69, + 0x6a, + 0x6b, + 0x6c, + 0x6d, + 0x90, /* ASRC1L */ + 0x91, + 0x92, + 0x93, +}; +EXPORT_SYMBOL_GPL(arizona_mixer_values); + +const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0); +EXPORT_SYMBOL_GPL(arizona_mixer_tlv); + +static const char *arizona_lhpf_mode_text[] = { + "Low-pass", "High-pass" +}; + +const struct soc_enum arizona_lhpf1_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF1_1, ARIZONA_LHPF1_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf1_mode); + +const struct soc_enum arizona_lhpf2_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF2_1, ARIZONA_LHPF2_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf2_mode); + +const struct soc_enum arizona_lhpf3_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF3_1, ARIZONA_LHPF3_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf3_mode); + +const struct soc_enum arizona_lhpf4_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF4_1, ARIZONA_LHPF4_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); + +int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, + int event) +{ + return 0; +} +EXPORT_SYMBOL_GPL(arizona_in_ev); + +int arizona_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + return 0; +} +EXPORT_SYMBOL_GPL(arizona_out_ev); + +int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + char *name; + unsigned int reg; + unsigned int mask = ARIZONA_SYSCLK_FREQ_MASK | ARIZONA_SYSCLK_SRC_MASK; + unsigned int val = source << ARIZONA_SYSCLK_SRC_SHIFT; + unsigned int *clk; + + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + name = "SYSCLK"; + reg = ARIZONA_SYSTEM_CLOCK_1; + clk = &priv->sysclk; + mask |= ARIZONA_SYSCLK_FRAC; + break; + case ARIZONA_CLK_ASYNCCLK: + name = "ASYNCCLK"; + reg = ARIZONA_ASYNC_CLOCK_1; + clk = &priv->asyncclk; + break; + default: + return -EINVAL; + } + + switch (freq) { + case 5644800: + case 6144000: + break; + case 11289600: + case 12288000: + val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + case 22579200: + case 24576000: + val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + case 45158400: + case 49152000: + val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + default: + return -EINVAL; + } + + *clk = freq; + + if (freq % 6144000) + val |= ARIZONA_SYSCLK_FRAC; + + dev_dbg(arizona->dev, "%s set to %uHz", name, freq); + + return regmap_update_bits(arizona->regmap, reg, mask, val); +} +EXPORT_SYMBOL_GPL(arizona_set_sysclk); + +static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int lrclk, bclk, mode, base; + + base = dai->driver->base; + + lrclk = 0; + bclk = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + mode = 0; + break; + case SND_SOC_DAIFMT_DSP_B: + mode = 1; + break; + case SND_SOC_DAIFMT_I2S: + mode = 2; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode = 3; + break; + default: + arizona_aif_err(dai, "Unsupported DAI format %d\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + bclk |= ARIZONA_AIF1_BCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + bclk |= ARIZONA_AIF1_BCLK_MSTR; + lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR; + break; + default: + arizona_aif_err(dai, "Unsupported master mode %d\n", + fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + bclk |= ARIZONA_AIF1_BCLK_INV; + lrclk |= ARIZONA_AIF1TX_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + bclk |= ARIZONA_AIF1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk |= ARIZONA_AIF1TX_LRCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_INV | ARIZONA_AIF1_BCLK_MSTR, + bclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_PIN_CTRL, + ARIZONA_AIF1TX_LRCLK_INV | + ARIZONA_AIF1TX_LRCLK_MSTR, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_PIN_CTRL, + ARIZONA_AIF1RX_LRCLK_INV | + ARIZONA_AIF1RX_LRCLK_MSTR, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FORMAT, + ARIZONA_AIF1_FMT_MASK, mode); + + return 0; +} + +static const int arizona_48k_rates[] = { + -1, + 48000, + 64000, + 96000, + 128000, + 192000, + 256000, + 384000, + 512000, + 768000, + 1024000, + 1536000, + 2048000, + 3072000, + 4096000, + 6144000, + 8192000, + 12288000, + 24576000, +}; + +static const int arizona_44k1_rates[] = { + -1, + 44100, + 58800, + 88200, + 117600, + 177640, + 235200, + 352800, + 470400, + 705600, + 940800, + 1411200, + 1881600, + 2882400, + 3763200, + 5644800, + 7526400, + 11289600, + 22579200, +}; + +static int arizona_sr_vals[] = { + 0, + 12000, + 24000, + 48000, + 96000, + 192000, + 384000, + 768000, + 0, + 11025, + 22050, + 44100, + 88200, + 176400, + 352800, + 705600, + 4000, + 8000, + 16000, + 32000, + 64000, + 128000, + 256000, + 512000, +}; + +static int arizona_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + int base = dai->driver->base; + const int *rates; + int i; + int bclk, lrclk, wl, frame, sr_val; + + if (params_rate(params) % 8000) + rates = &arizona_44k1_rates[0]; + else + rates = &arizona_48k_rates[0]; + + for (i = 0; i < ARRAY_SIZE(arizona_44k1_rates); i++) { + if (rates[i] == snd_soc_params_to_bclk(params)) { + bclk = i; + break; + } + } + if (i == ARRAY_SIZE(arizona_44k1_rates)) { + arizona_aif_err(dai, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + + /* + * We will need to be more flexible than this in future, + * currently we use a single sample rate for the chip. + */ + for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++) + if (arizona_sr_vals[i] == params_rate(params)) + break; + if (i == ARRAY_SIZE(arizona_sr_vals)) { + arizona_aif_err(dai, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + sr_val = i; + + lrclk = snd_soc_params_to_bclk(params) / params_rate(params); + + arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", + rates[bclk], rates[bclk] / lrclk); + + wl = snd_pcm_format_width(params_format(params)); + frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, + ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE, + ARIZONA_AIF1TX_BCPF_MASK, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE, + ARIZONA_AIF1RX_BCPF_MASK, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1, + ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2, + ARIZONA_AIF1RX_WL_MASK | + ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + + return 0; +} + +const struct snd_soc_dai_ops arizona_dai_ops = { + .set_fmt = arizona_set_fmt, + .hw_params = arizona_hw_params, +}; + +static irqreturn_t arizona_fll_lock(int irq, void *data) +{ + struct arizona_fll *fll = data; + + arizona_fll_dbg(fll, "Locked\n"); + + complete(&fll->lock); + + return IRQ_HANDLED; +} + +static irqreturn_t arizona_fll_clock_ok(int irq, void *data) +{ + struct arizona_fll *fll = data; + + arizona_fll_dbg(fll, "clock OK\n"); + + complete(&fll->ok); + + return IRQ_HANDLED; +} + +static struct { + unsigned int min; + unsigned int max; + u16 fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +struct arizona_fll_cfg { + int n; + int theta; + int lambda; + int refdiv; + int outdiv; + int fratio; +}; + +static int arizona_calc_fll(struct arizona_fll *fll, + struct arizona_fll_cfg *cfg, + unsigned int Fref, + unsigned int Fout) +{ + unsigned int target, div, gcd_fll; + int i, ratio; + + arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout); + + /* Fref must be <=13.5MHz */ + div = 1; + cfg->refdiv = 0; + while ((Fref / div) > 13500000) { + div *= 2; + cfg->refdiv++; + + if (div > 8) { + arizona_fll_err(fll, + "Can't scale %dMHz in to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be 90-100MHz; don't check the upper bound */ + div = 1; + while (Fout * div < 90000000) { + div++; + if (div > 7) { + arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * div; + cfg->outdiv = div; + + arizona_fll_dbg(fll, "Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + cfg->fratio = fll_fratios[i].fratio; + ratio = fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n", + Fref); + return -EINVAL; + } + + cfg->n = target / (ratio * Fref); + + if (target % Fref) { + gcd_fll = gcd(target, ratio * Fref); + arizona_fll_dbg(fll, "GCD=%u\n", gcd_fll); + + cfg->theta = (target - (cfg->n * ratio * Fref)) + / gcd_fll; + cfg->lambda = (ratio * Fref) / gcd_fll; + } else { + cfg->theta = 0; + cfg->lambda = 0; + } + + arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n", + cfg->n, cfg->theta, cfg->lambda); + arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", + cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv); + + return 0; + +} + +static void arizona_apply_fll(struct arizona *arizona, unsigned int base, + struct arizona_fll_cfg *cfg, int source) +{ + regmap_update_bits(arizona->regmap, base + 3, + ARIZONA_FLL1_THETA_MASK, cfg->theta); + regmap_update_bits(arizona->regmap, base + 4, + ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda); + regmap_update_bits(arizona->regmap, base + 5, + ARIZONA_FLL1_FRATIO_MASK, + cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT); + regmap_update_bits(arizona->regmap, base + 6, + ARIZONA_FLL1_CLK_REF_DIV_MASK | + ARIZONA_FLL1_CLK_REF_SRC_MASK, + cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | + source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); + + regmap_update_bits(arizona->regmap, base + 2, + ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, + ARIZONA_FLL1_CTRL_UPD | cfg->n); +} + +int arizona_set_fll(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout) +{ + struct arizona *arizona = fll->arizona; + struct arizona_fll_cfg cfg, sync; + unsigned int reg, val; + int syncsrc; + bool ena; + int ret; + + ret = regmap_read(arizona->regmap, fll->base + 1, &reg); + if (ret != 0) { + arizona_fll_err(fll, "Failed to read current state: %d\n", + ret); + return ret; + } + ena = reg & ARIZONA_FLL1_ENA; + + if (Fout) { + /* Do we have a 32kHz reference? */ + regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val); + switch (val & ARIZONA_CLK_32K_SRC_MASK) { + case ARIZONA_CLK_SRC_MCLK1: + case ARIZONA_CLK_SRC_MCLK2: + syncsrc = val & ARIZONA_CLK_32K_SRC_MASK; + break; + default: + syncsrc = -1; + } + + if (source == syncsrc) + syncsrc = -1; + + if (syncsrc >= 0) { + ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (ret != 0) + return ret; + + ret = arizona_calc_fll(fll, &cfg, 32768, Fout); + if (ret != 0) + return ret; + } else { + ret = arizona_calc_fll(fll, &cfg, Fref, Fout); + if (ret != 0) + return ret; + } + } else { + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, 0); + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, 0); + + if (ena) + pm_runtime_put_autosuspend(arizona->dev); + + return 0; + } + + regmap_update_bits(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + cfg.outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + + if (syncsrc >= 0) { + arizona_apply_fll(arizona, fll->base, &cfg, syncsrc); + arizona_apply_fll(arizona, fll->base + 0x10, &sync, source); + } else { + arizona_apply_fll(arizona, fll->base, &cfg, source); + } + + if (!ena) + pm_runtime_get(arizona->dev); + + /* Clear any pending completions */ + try_wait_for_completion(&fll->ok); + + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); + if (syncsrc >= 0) + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, + ARIZONA_FLL1_SYNC_ENA); + + ret = wait_for_completion_timeout(&fll->ok, + msecs_to_jiffies(25)); + if (ret == 0) + arizona_fll_warn(fll, "Timed out waiting for lock\n"); + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_set_fll); + +int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, + int ok_irq, struct arizona_fll *fll) +{ + int ret; + + init_completion(&fll->lock); + init_completion(&fll->ok); + + fll->id = id; + fll->base = base; + fll->arizona = arizona; + + snprintf(fll->lock_name, sizeof(fll->lock_name), "FLL%d lock", id); + snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), + "FLL%d clock OK", id); + + ret = arizona_request_irq(arizona, lock_irq, fll->lock_name, + arizona_fll_lock, fll); + if (ret != 0) { + dev_err(arizona->dev, "Failed to get FLL%d lock IRQ: %d\n", + id, ret); + } + + ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, + arizona_fll_clock_ok, fll); + if (ret != 0) { + dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n", + id, ret); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_fll); + +MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h new file mode 100644 index 0000000..8c2ca1d --- /dev/null +++ b/sound/soc/codecs/arizona.h @@ -0,0 +1,149 @@ +/* + * arizona.h - Wolfson Arizona class device shared support + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ASOC_ARIZONA_H +#define _ASOC_ARIZONA_H + +#include <linux/completion.h> + +#include <sound/soc.h> + +#define ARIZONA_CLK_SYSCLK 1 +#define ARIZONA_CLK_ASYNCCLK 2 + +#define ARIZONA_CLK_SRC_MCLK1 0x0 +#define ARIZONA_CLK_SRC_MCLK2 0x1 +#define ARIZONA_CLK_SRC_FLL1 0x4 +#define ARIZONA_CLK_SRC_FLL2 0x5 +#define ARIZONA_CLK_SRC_AIF1BCLK 0x8 +#define ARIZONA_CLK_SRC_AIF2BCLK 0x9 +#define ARIZONA_CLK_SRC_AIF3BCLK 0xa + +#define ARIZONA_FLL_SRC_MCLK1 0 +#define ARIZONA_FLL_SRC_MCLK2 1 +#define ARIZONA_FLL_SRC_SLIMCLK 2 +#define ARIZONA_FLL_SRC_FLL1 3 +#define ARIZONA_FLL_SRC_FLL2 4 +#define ARIZONA_FLL_SRC_AIF1BCLK 5 +#define ARIZONA_FLL_SRC_AIF2BCLK 6 +#define ARIZONA_FLL_SRC_AIF3BCLK 7 +#define ARIZONA_FLL_SRC_AIF1LRCLK 8 +#define ARIZONA_FLL_SRC_AIF2LRCLK 9 +#define ARIZONA_FLL_SRC_AIF3LRCLK 10 + +#define ARIZONA_MIXER_VOL_MASK 0x00FE +#define ARIZONA_MIXER_VOL_SHIFT 1 +#define ARIZONA_MIXER_VOL_WIDTH 7 + +struct arizona; + +struct arizona_priv { + struct arizona *arizona; + int sysclk; + int asyncclk; +}; + +#define ARIZONA_NUM_MIXER_INPUTS 55 + +extern const unsigned int arizona_mixer_tlv[]; +extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; +extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; + +#define ARIZONA_MIXER_CONTROLS(name, base) \ + SOC_SINGLE_RANGE_TLV(name " Input 1 Volume", base + 1, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 2 Volume", base + 3, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 3 Volume", base + 5, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 4 Volume", base + 7, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv) + +#define ARIZONA_MUX_ENUM_DECL(name, reg) \ + SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff, \ + arizona_mixer_texts, arizona_mixer_values) + +#define ARIZONA_MUX_CTL_DECL(name) \ + const struct snd_kcontrol_new name##_mux = \ + SOC_DAPM_VALUE_ENUM("Route", name##_enum) + +#define ARIZONA_MIXER_ENUMS(name, base_reg) \ + static ARIZONA_MUX_ENUM_DECL(name##_in1_enum, base_reg); \ + static ARIZONA_MUX_ENUM_DECL(name##_in2_enum, base_reg + 2); \ + static ARIZONA_MUX_ENUM_DECL(name##_in3_enum, base_reg + 4); \ + static ARIZONA_MUX_ENUM_DECL(name##_in4_enum, base_reg + 6); \ + static ARIZONA_MUX_CTL_DECL(name##_in1); \ + static ARIZONA_MUX_CTL_DECL(name##_in2); \ + static ARIZONA_MUX_CTL_DECL(name##_in3); \ + static ARIZONA_MUX_CTL_DECL(name##_in4) + +#define ARIZONA_MUX(name, ctrl) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl) + +#define ARIZONA_MIXER_WIDGETS(name, name_str) \ + ARIZONA_MUX(name_str " Input 1", &name##_in1_mux), \ + ARIZONA_MUX(name_str " Input 2", &name##_in2_mux), \ + ARIZONA_MUX(name_str " Input 3", &name##_in3_mux), \ + ARIZONA_MUX(name_str " Input 4", &name##_in4_mux), \ + SND_SOC_DAPM_MIXER(name_str " Mixer", SND_SOC_NOPM, 0, 0, NULL, 0) + +#define ARIZONA_MIXER_ROUTES(widget, name) \ + { widget, NULL, name " Mixer" }, \ + { name " Mixer", NULL, name " Input 1" }, \ + { name " Mixer", NULL, name " Input 2" }, \ + { name " Mixer", NULL, name " Input 3" }, \ + { name " Mixer", NULL, name " Input 4" }, \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 1"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 2"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 3"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 4") + +extern const struct soc_enum arizona_lhpf1_mode; +extern const struct soc_enum arizona_lhpf2_mode; +extern const struct soc_enum arizona_lhpf3_mode; +extern const struct soc_enum arizona_lhpf4_mode; + +extern int arizona_in_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); +extern int arizona_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); + +extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir); + +extern const struct snd_soc_dai_ops arizona_dai_ops; + +#define ARIZONA_FLL_NAME_LEN 20 + +struct arizona_fll { + struct arizona *arizona; + int id; + unsigned int base; + struct completion lock; + struct completion ok; + + char lock_name[ARIZONA_FLL_NAME_LEN]; + char clock_ok_name[ARIZONA_FLL_NAME_LEN]; +}; + +extern int arizona_init_fll(struct arizona *arizona, int id, int base, + int lock_irq, int ok_irq, struct arizona_fll *fll); +extern int arizona_set_fll(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout); + +#endif -- 1.7.10

1 1

Re: [alsa-devel] [PATCH] sound: fix MODULE_LICENSE typos
by Mark Brown 23 Jun '12

23 Jun '12

On Fri, Jun 22, 2012 at 04:21:49PM -0700, Kees Cook wrote: > include/linux/license.h does not recognize "GPLv2", so adjust these > declarations to add the needed space before the version. This patch isn't against current -next which doesn't have this issue, I've no idea what you generated against as a version with this has never even been in -next. Please do also use subject lines that resemble those used in the rest of the subsystem, having to hand edit commit logs is tedious. It seems more sensible to fix license.h, GPLv2 is a pretty common way of writing the license name in the wider community.

1 0

[alsa-devel] [PATCH] ASoC: core: Add DOUBLE_R variants of the _RANGE controls
by Mark Brown 22 Jun '12

22 Jun '12

The code handles this fine already, we just need new macros in the header for drivers to create the controls. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index e4348d2..e063380 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -42,6 +42,10 @@ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ .max = xmax, .platform_max = xmax, .invert = xinvert}) +#define SOC_DOUBLE_R_RANGE_VALUE(xlreg, xrreg, xshift, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_mixer_control) \ + {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ + .min = xmin, .max = xmax, .platform_max = xmax, .invert = xinvert}) #define SOC_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ @@ -96,6 +100,13 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_RANGE(xname, reg_left, reg_right, xshift, xmin, \ + xmax, xinvert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .info = snd_soc_info_volsw_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ + xshift, xmin, xmax, xinvert) } #define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -114,6 +125,16 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_RANGE_TLV(xname, reg_left, reg_right, xshift, xmin, \ + xmax, xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ + xshift, xmin, xmax, xinvert) } #define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ -- 1.7.10

2 1

[alsa-devel] [PATCH] ASoC: dapm: Try to add all routes even if one fails
by Mark Brown 22 Jun '12

22 Jun '12

We may as well print as many errors as we can in one go rather than requiring developers to iterate through all their typos. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 667a941..3e7baab 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2276,15 +2276,15 @@ err: int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num) { - int i, ret = 0; + int i, r, ret = 0; mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { - ret = snd_soc_dapm_add_route(dapm, route); - if (ret < 0) { + r = snd_soc_dapm_add_route(dapm, route); + if (r < 0) { dev_err(dapm->dev, "Failed to add route %s->%s\n", route->source, route->sink); - break; + ret = r; } route++; } -- 1.7.10

2 1

[PATCH] ALSA: hda - Add inverted mic quirks for Asus U41SV, Acer 1810TZ and AOD260
by David Henningsson 22 Jun '12

22 Jun '12

These machines have inverted phase on right channel for their internal mics. BugLink: https://bugs.launchpad.net/bugs/997227 BugLink: https://bugs.launchpad.net/bugs/996611 BugLink: https://bugs.launchpad.net/bugs/1006089 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++------ 1 file changed, 17 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 53cd730..9be7da1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5918,6 +5918,14 @@ static void alc269_fixup_mic2_mute(struct hda_codec *codec, } } +static void alc269_fixup_inv_dmic(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PROBE) + alc_add_inv_dmic_mixer(codec, 0x12); +} + + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5936,6 +5944,7 @@ enum { ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, ALC269_FIXUP_MIC2_MUTE_LED, + ALC269_FIXUP_INV_DMIC, }; static const struct alc_fixup alc269_fixups[] = { @@ -6060,12 +6069,19 @@ static const struct alc_fixup alc269_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, }, + [ALC269_FIXUP_INV_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_inv_dmic, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), + SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), @@ -6534,12 +6550,7 @@ static void alc272_fixup_mario(struct hda_codec *codec, "hda_codec: failed to override amp caps for NID 0x2\n"); } -static void alc662_fixup_inv_dmic(struct hda_codec *codec, - const struct alc_fixup *fix, int action) -{ - if (action == ALC_FIXUP_ACT_PROBE) - alc_add_inv_dmic_mixer(codec, 0x12); -} +#define alc662_fixup_inv_dmic alc269_fixup_inv_dmic enum { ALC662_FIXUP_ASPIRE, -- 1.7.9.5 --------------030106040506020209090705--

1 0

[alsa-devel] [PATCH] ASoC: da732x: Staticise non-exported symbol soc_codec_dev_da732x
by Mark Brown 22 Jun '12

22 Jun '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/da732x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 04af369..01be2a3 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1543,7 +1543,7 @@ static int da732x_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver soc_codec_dev_da732x = { +static struct snd_soc_codec_driver soc_codec_dev_da732x = { .probe = da732x_probe, .remove = da732x_remove, .set_bias_level = da732x_set_bias_level, -- 1.7.10

2 1

[alsa-devel] [PATCH] ASoC: isabelle: Staticise non-exported isabelle_dai
by Mark Brown 22 Jun '12

22 Jun '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/isabelle.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 0d62f3b..5d8f39e 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1036,7 +1036,7 @@ static struct snd_soc_dai_ops isabelle_ul_dai_ops = { }; /* ISABELLE dai structure */ -struct snd_soc_dai_driver isabelle_dai[] = { +static struct snd_soc_dai_driver isabelle_dai[] = { { .name = "isabelle-dl1", .playback = { -- 1.7.10

1 0

[alsa-devel] [PATCH] ALSA: hda - Don't power up when not powered down.
by Dylan Reid 22 Jun '12

22 Jun '12

After cancel_delayed_work_sync returns, the power down work either never started (power_on == 1) or finished (power_on == 0). In the former case there is no need to power up again. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_codec.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 604699c..045b5e7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4444,6 +4444,13 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) cancel_delayed_work_sync(&codec->power_work); spin_lock(&codec->power_lock); + /* If the power down delayed work was cancelled above before starting, + * then there is no need to go through power up here. + */ + if (codec->power_on) { + spin_unlock(&codec->power_lock); + return; + } trace_hda_power_up(codec); snd_hda_update_power_acct(codec); codec->power_on = 1; -- 1.7.9.rc0

2 3

[alsa-devel] Need ALSA soc architecture Documents
by raj raja 22 Jun '12

22 Jun '12

Hi, I am very new to ALSA soc core . I want to have some basic knowledge on ALSA soc core architecture Where can i find the related pfd or ppt format documents Thanks Yuvaraj R

1 0

[alsa-devel] [PATCH V3 12/16] ALSA: HDA: Make hda sound card usable for Loongson.
by Huacai Chen 22 Jun '12

22 Jun '12

Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec, this patch modify patch_conexant.c to add Lemote specific code. Signed-off-by: Huacai Chen <chenhc(a)lemote.com> Signed-off-by: Hongliang Tao <taohl(a)lemote.com> Signed-off-by: Hua Yan <yanh(a)lemote.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Reviewed-by: Clemens Ladisch <clemens(a)ladisch.de> Cc: alsa-devel(a)alsa-project.org --- include/linux/pci_ids.h | 2 + sound/pci/hda/patch_conexant.c | 52 ++++++++++++++++++++++++++++++++++++++- 2 files changed, 52 insertions(+), 2 deletions(-) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index ab741b0..d8b0a52 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -2906,3 +2906,5 @@ #define PCI_DEVICE_ID_XEN_PLATFORM 0x0001 #define PCI_VENDOR_ID_OCZ 0x1b85 + +#define PCI_VENDOR_ID_LEMOTE 0x1c06 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 172370b..ca3cc237 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -142,6 +142,7 @@ struct conexant_spec { unsigned int thinkpad:1; unsigned int hp_laptop:1; unsigned int asus:1; + unsigned int lemote:1; unsigned int pin_eapd_ctrls:1; unsigned int fixup_stereo_dmic:1; @@ -2278,7 +2279,7 @@ static void cxt5066_automic(struct hda_codec *codec) cxt5066_thinkpad_automic(codec); else if (spec->hp_laptop) cxt5066_hp_laptop_automic(codec); - else if (spec->asus) + else if (spec->asus || spec->lemote) cxt5066_asus_automic(codec); } @@ -2911,6 +2912,32 @@ static const struct hda_verb cxt5066_init_verbs_hp_laptop[] = { { } /* end */ }; +static struct hda_verb cxt5066_init_verbs_lemote[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1: Connection index: 0 */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + + /* DAC2: unused */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + /* ADC2, ADC3: unused */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* Disable digital microphone port */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Disable SPDIF */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { @@ -2948,6 +2975,8 @@ enum { CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ CXT5066_HP_LAPTOP, /* HP Laptop */ + CXT5066_LEMOTE_A1004, /* Lemote Laptop A1004 */ + CXT5066_LEMOTE_A1205, /* Lemote All-In-One A1205 */ CXT5066_AUTO, /* BIOS auto-parser */ CXT5066_MODELS }; @@ -2961,6 +2990,8 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_THINKPAD] = "thinkpad", [CXT5066_ASUS] = "asus", [CXT5066_HP_LAPTOP] = "hp-laptop", + [CXT5066_LEMOTE_A1004] = "lemote-laptop-a1004", + [CXT5066_LEMOTE_A1205] = "lemote-aio-a1205", [CXT5066_AUTO] = "auto", }; @@ -2993,6 +3024,8 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), + SND_PCI_QUIRK(PCI_VENDOR_ID_LEMOTE, 0x2011, "Lemote A1004", CXT5066_LEMOTE_A1004), + SND_PCI_QUIRK(PCI_VENDOR_ID_LEMOTE, 0x2012, "Lemote A1205", CXT5066_LEMOTE_A1205), SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; @@ -3073,7 +3106,22 @@ static int patch_cxt5066(struct hda_codec *codec) spec->port_d_mode = 0; spec->mic_boost = 3; /* default 30dB gain */ break; - + case CXT5066_LEMOTE_A1004: + case CXT5066_LEMOTE_A1205: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_unsol_event; + spec->init_verbs[spec->num_init_verbs] = + cxt5066_init_verbs_lemote; + spec->num_init_verbs++; + spec->lemote = 1; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + /* no S/PDIF out */ + /* input source automatically selected */ + spec->input_mux = NULL; + spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ + break; case CXT5066_OLPC_XO_1_5: codec->patch_ops.init = cxt5066_olpc_init; codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; -- 1.7.7.3

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