November 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC updates for 2.6.37
by Mark Brown 22 Nov '10

22 Nov '10

The following changes since commit 75e3f3137cb570661c2ad3035a139dda671fbb63: ASoC: tpa6130a2: Get rid of compile warning from tpa6130a2_power (2010-11-03 15:50:46 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.37 Mostly small fixes - a lot of them due to your holiday (hope you had a good time!). The big changes are the i.MX DMA update which was postponed until after the i.MX tree had been merged into mainline. The new support for the Maxim chip is just a new device ID, no functional code changes. Axel Lin (3): ASoC: Fix snd_soc_register_dais error handling ASoC: Return proper error if snd_soc_register_dais fails in psc_i2s_of_probe ASoC: uda134x - set reg_cache_default to uda134x_reg Chris Paulson-Ellis (1): ASoC: davinci: fixes for multi-component Dimitris Papastamos (1): ASoC: WM8776: Removed unneeded struct member Jesse Marroquin (1): ASoC: Add support for MAX98089 CODEC Marek Belisko (1): ASoC: s3c24xx: Fix compilation problem for mini2440 Mark Brown (7): Merge remote branch 'takashi/fix/asoc' into for-2.6.37 ASoC: Check return value of strict_strtoul() in WM8962 ASoC: Lock the CODEC in PXA external jack controls ASoC: Remove broken WM8350 direction constants ASoC: Ensure sane WM835x AIF configuration by default ASoC: Fix dapm_seq_compare() for multi-component ASoC: Fix register cache setup WM8994 for multi-component Ryan Mallon (1): Fix Atmel soc audio boards Kconfig dependency Sascha Hauer (5): ASoC i.MX eukrea tlv320: Fix for multicomponent ASoC i.MX phycore ac97: remove unnecessary includes ASoC i.MX: register dma audio device ASoC i.MX: switch to new DMA api phycore-ac97: add ac97 to cardname Vasily Khoruzhick (1): ASoC: RX1950: Fix hw_params function arch/arm/mach-davinci/dm355.c | 6 +- arch/arm/mach-davinci/dm365.c | 6 +- arch/arm/mach-davinci/dm644x.c | 4 +- include/linux/mfd/wm8350/audio.h | 3 - sound/soc/atmel/Kconfig | 5 +- sound/soc/codecs/max98088.c | 11 ++- sound/soc/codecs/uda134x.c | 1 + sound/soc/codecs/wm8350.c | 9 ++- sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8962.c | 5 +- sound/soc/codecs/wm8994.c | 2 + sound/soc/davinci/davinci-evm.c | 40 +++++-- sound/soc/davinci/davinci-i2s.c | 15 ++- sound/soc/davinci/davinci-mcasp.c | 13 ++- sound/soc/davinci/davinci-sffsdr.c | 2 +- sound/soc/davinci/davinci-vcif.c | 13 ++- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/imx/eukrea-tlv320.c | 8 +- sound/soc/imx/imx-pcm-dma-mx2.c | 221 ++++++++++++++++-------------------- sound/soc/imx/imx-ssi.c | 44 +++++--- sound/soc/imx/imx-ssi.h | 4 + sound/soc/imx/phycore-ac97.c | 5 +- sound/soc/pxa/corgi.c | 5 + sound/soc/pxa/magician.c | 4 + sound/soc/pxa/poodle.c | 5 + sound/soc/pxa/spitz.c | 5 + sound/soc/pxa/tosa.c | 5 + sound/soc/s3c24xx/Kconfig | 1 + sound/soc/s3c24xx/rx1950_uda1380.c | 20 +--- sound/soc/soc-core.c | 9 +- sound/soc/soc-dapm.c | 4 +- 31 files changed, 268 insertions(+), 210 deletions(-)

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[alsa-devel] [PATCH] ASoC: Restore WM8994 volatile and readable register operations
by Mark Brown 22 Nov '10

22 Nov '10

They went AWOL during the multi-component merge. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index de1447c..736cbf4 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3021,6 +3021,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { .resume = wm8994_resume, .read = wm8994_read, .write = wm8994_write, + .readable_register = wm8994_readable, + .volatile_register = wm8994_volatile, .set_bias_level = wm8994_set_bias_level, .reg_cache_size = WM8994_CACHE_SIZE, -- 1.7.1

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[alsa-devel] [PATCH 1/3] ALSA: HDA: Create mixers on ALC887
by David Henningsson 22 Nov '10

22 Nov '10

BugLink: http://launchpad.net/bugs/669092 ALC887 does not have any volume control ability on the mixer NIDs, so put the volume controls on the dac NIDs instead. Without this patch, ALC887 users cannot use alsamixer at all. Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] Found bug in latest sound/pci/hda/patch_realtek.c from git, the bug is about jack sensing
by Kim Therkelsen 22 Nov '10

22 Nov '10

Hi, I think I have found a bug in patch_realtek.c. I have a 272X codec in my laptop and have discovered that it is only reported through the ALSA driver API when my headphone minijack is disconnected and not when it is connected. This is happening because alc_report_jack(...) is only called when the minijack connector is disconnected and not when it is connected. In order to make jack sensing work properly i modified patch_realtek.c and added alc_report_jack(codec, spec->autocfg.hp_pins[i]); at line 1132 (see code below). The code shown below is from: http://git.kernel.org/?p=linux/kernel/git/next/linux-next.git;a=blob;f=soun… 1118 static void alc_automute_speaker(struct hda_codec *codec, int pinctl) 1119 { 1120 struct alc_spec *spec = codec->spec; 1121 unsigned int mute; 1122 hda_nid_t nid; 1123 int i; 1124 1125 spec->jack_present = 0; 1126 for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { 1127 nid = spec->autocfg.hp_pins[i]; 1128 if (!nid) 1129 break; 1130 if (snd_hda_jack_detect(codec, nid)) { 1131 spec->jack_present = 1; //ADDED BY KT: alc_report_jack(codec, spec->autocfg.hp_pins[i]); 1132 break; 1133 } 1134 alc_report_jack(codec, spec->autocfg.hp_pins[i]); 1135 } 1136 1137 mute = spec->jack_present ? HDA_AMP_MUTE : 0; 1138 /* Toggle internal speakers muting */ 1139 for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { 1140 nid = spec->autocfg.speaker_pins[i]; 1141 if (!nid) 1142 break; 1143 if (pinctl) { 1144 snd_hda_codec_write(codec, nid, 0, 1145 AC_VERB_SET_PIN_WIDGET_CONTROL, 1146 spec->jack_present ? 0 : PIN_OUT); 1147 } else { 1148 snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, 1149 HDA_AMP_MUTE, mute); 1150 } 1151 } 1152 } Best regards, Kim Therkelsen

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[alsa-devel] [PATCH] ALSA: snd-atmel-abdac: test wrong variable
by Vasiliy Kulikov 22 Nov '10

22 Nov '10

After clk_get() pclk is checked second time instead of sample_clk check. Signed-off-by: Vasiliy Kulikov <segoon(a)openwall.com> --- Compile tested only. sound/atmel/abdac.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index f2f41c8..4e47baa 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -420,7 +420,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev) return PTR_ERR(pclk); } sample_clk = clk_get(&pdev->dev, "sample_clk"); - if (IS_ERR(pclk)) { + if (IS_ERR(sample_clk)) { dev_dbg(&pdev->dev, "no sample clock\n"); retval = PTR_ERR(pclk); goto out_put_pclk; -- 1.7.0.4

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Re: [alsa-devel] [PATCH] sound/mixart: avoid redefining {readl, write}_{le, be} accessors
by Takashi Iwai 22 Nov '10

22 Nov '10

At Fri, 12 Nov 2010 18:29:11 +0100, Florian Fainelli wrote: > > On Tuesday 09 November 2010 18:29:08 Florian Fainelli wrote: > > From: Florian Fainelli <florian(a)openwrt.org> > > Takashi, do you have any comments on this patch? Sorry for the late reply. Yes, I already applied it weeks ago, but the reply mail wasn't sent out but stuck in my output queue by some reason during my vacation. Anyway, thanks for the patch. I think it's OK in the way you did for the time being. Ideally, read*_be() should be defined on all platforms, though. Takashi > > Thanks! > > > > > If the platform already provides a definition for these accessors > > do not redefine them. The warning was caught on MIPS. > > > > Signed-off-by: Florian Fainelli <florian(a)openwrt.org> > > --- > > diff --git a/sound/pci/mixart/mixart_hwdep.h > > b/sound/pci/mixart/mixart_hwdep.h index a46f508..812e288 100644 > > --- a/sound/pci/mixart/mixart_hwdep.h > > +++ b/sound/pci/mixart/mixart_hwdep.h > > @@ -25,11 +25,21 @@ > > > > #include <sound/hwdep.h> > > > > +#ifndef readl_be > > #define readl_be(x) be32_to_cpu(__raw_readl(x)) > > +#endif > > + > > +#ifndef writel_be > > #define writel_be(data,addr) __raw_writel(cpu_to_be32(data),addr) > > +#endif > > > > +#ifndef readl_le > > #define readl_le(x) le32_to_cpu(__raw_readl(x)) > > +#endif > > + > > +#ifndef writel_le > > #define writel_le(data,addr) __raw_writel(cpu_to_le32(data),addr) > > +#endif > > > > #define MIXART_MEM(mgr,x) ((mgr)->mem[0].virt + (x)) > > #define MIXART_REG(mgr,x) ((mgr)->mem[1].virt + (x)) >

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[alsa-devel] [PATCH 48/49] sound/oss/dev_table.c: Use vzalloc
by Joe Perches 22 Nov '10

22 Nov '10

Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/oss/dev_table.c | 6 ++---- 1 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c index 727bdb9..d8cf3e5 100644 --- a/sound/oss/dev_table.c +++ b/sound/oss/dev_table.c @@ -71,7 +71,7 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver, if (sound_nblocks >= MAX_MEM_BLOCKS) sound_nblocks = MAX_MEM_BLOCKS - 1; - op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_operations))); + op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vzalloc(sizeof(struct audio_operations))); sound_nblocks++; if (sound_nblocks >= MAX_MEM_BLOCKS) sound_nblocks = MAX_MEM_BLOCKS - 1; @@ -81,7 +81,6 @@ int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver, sound_unload_audiodev(num); return -(ENOMEM); } - memset((char *) op, 0, sizeof(struct audio_operations)); init_waitqueue_head(&op->in_sleeper); init_waitqueue_head(&op->out_sleeper); init_waitqueue_head(&op->poll_sleeper); @@ -128,7 +127,7 @@ int sound_install_mixer(int vers, char *name, struct mixer_operations *driver, /* FIXME: This leaks a mixer_operations struct every time its called until you unload sound! */ - op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct mixer_operations))); + op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vzalloc(sizeof(struct mixer_operations))); sound_nblocks++; if (sound_nblocks >= MAX_MEM_BLOCKS) sound_nblocks = MAX_MEM_BLOCKS - 1; @@ -137,7 +136,6 @@ int sound_install_mixer(int vers, char *name, struct mixer_operations *driver, printk(KERN_ERR "Sound: Can't allocate mixer driver for (%s)\n", name); return -ENOMEM; } - memset((char *) op, 0, sizeof(struct mixer_operations)); memcpy((char *) op, (char *) driver, driver_size); strlcpy(op->name, name, sizeof(op->name)); -- 1.7.3.1.g432b3.dirty

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[alsa-devel] [PATCH] ALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus A52J
by Daniel Chen 22 Nov '10

22 Nov '10

Hi, Please apply this patch against current sound-2.6 master HEAD to enable headphones automute for the bug reporter's laptop. Thanks, -Dan

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[alsa-devel] [PATCH] azt3328: period bug fix (PulseAudio testing), add missing ACK on stop timer
by Andreas Mohr 22 Nov '10

22 Nov '10

ChangeLog: . Fix PulseAudio "ALSA driver bug" issue (if we have two alternated areas within a 64k DMA buffer, then max period size should obviously be 32k only). Back references: http://pulseaudio.org/wiki/AlsaIssues http://fedoraproject.org/wiki/Features/GlitchFreeAudio . In stop timer function, need to supply ACK in the timer control byte. . Minor log output correction When I did my first PA testing recently, the period size bug resulted in quite precisely observeable half-period-based playback distortion. PA-based operation is quite a bit more underrun-prone (despite its zero-copy optimizations etc.) than raw ALSA with this rather spartan sound hardware implementation on my puny Athlon. Note that even with this patch, azt3328 still doesn't work for both cases yet, PA tsched=0 and tsched (on tsched=0 it will playback tiny fragments of periods, leading to tiny stuttering sounds with some pauses in between, whereas with timer-scheduled operation playback works fine - minus some quite increased underrun trouble on PA vs. ALSA, that is). Patch runtime-tested on 2.6.36-rc8 git (and run through checkpatch.pl). Since 2.6 git log doesn't show patch activity it should still apply easily. I'd suggest handling this patch whichever way you want. If you already have a couple other -rc fixes, then perhaps simply include it, and I'm quite ambivalent about -stable handling as well. Thanks! Signed-off-by: Andreas Mohr <andi(a)lisas.de> diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 4679ed8..2f3cacb 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1129,10 +1129,11 @@ snd_azf3328_codec_setdmaa(struct snd_azf3328 *chip, count_areas = size/2; addr_area2 = addr+count_areas; - count_areas--; /* max. index */ snd_azf3328_dbgcodec("setdma: buffers %08lx[%u] / %08lx[%u]\n", addr, count_areas, addr_area2, count_areas); + count_areas--; /* max. index */ + /* build combined I/O buffer length word */ lengths = (count_areas << 16) | (count_areas); spin_lock_irqsave(&chip->reg_lock, flags); @@ -1740,11 +1741,15 @@ static const struct snd_pcm_hardware snd_azf3328_hardware = .rate_max = AZF_FREQ_66200, .channels_min = 1, .channels_max = 2, - .buffer_bytes_max = 65536, - .period_bytes_min = 64, - .period_bytes_max = 65536, - .periods_min = 1, - .periods_max = 1024, + .buffer_bytes_max = (64*1024), + .period_bytes_min = 1024, + .period_bytes_max = (32*1024), + /* We simply have two DMA areas (instead of a list of descriptors + such as other cards); I believe that this is a fixed hardware + attribute and there isn't much driver magic to be done to expand it. + Thus indicate that we have at least and at most 2 periods. */ + .periods_min = 2, + .periods_max = 2, /* FIXME: maybe that card actually has a FIFO? * Hmm, it seems newer revisions do have one, but we still don't know * its size... */ @@ -1980,8 +1985,13 @@ snd_azf3328_timer_stop(struct snd_timer *timer) chip = snd_timer_chip(timer); spin_lock_irqsave(&chip->reg_lock, flags); /* disable timer countdown and interrupt */ - /* FIXME: should we write TIMER_IRQ_ACK here? */ - snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0); + /* Hmm, should we write TIMER_IRQ_ACK here? + YES indeed, otherwise a rogue timer operation - which prompts + ALSA(?) to call repeated stop() in vain, but NOT start() - + will never end (value 0x03 is kept shown in control byte). + Simply manually poking 0x04 _once_ immediately successfully stops + the hardware/ALSA interrupt activity. */ + snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x04); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_azf3328_dbgcallleave(); return 0;

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[alsa-devel] [PATCH] ALSA: hda: Add Samsung R720 SSID for subwoofer pin fixup
by Daniel Chen 22 Nov '10

22 Nov '10

Hi, Please apply the following patch against current sound-2.6 master HEAD to enable automuting of the subwoofer on the Samsung R720 when headphones are inserted. Thanks, -Dan

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