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

[alsa-devel] [PATCH] ALSA: include/sound/asound.h whitespace fixups
by Daniel Mack 11 May '10

11 May '10

This fixes some whitespace/indentation flaws I stumbled over. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> --- include/sound/asound.h | 20 ++++++++++---------- 1 files changed, 10 insertions(+), 10 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 0985955..9f1eecf 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -574,7 +574,7 @@ enum { #define SNDRV_TIMER_FLG_SLAVE (1<<0) /* cannot be controlled */ struct snd_timer_id { - int dev_class; + int dev_class; int dev_sclass; int card; int device; @@ -762,7 +762,7 @@ struct snd_ctl_elem_id { snd_ctl_elem_iface_t iface; /* interface identifier */ unsigned int device; /* device/client number */ unsigned int subdevice; /* subdevice (substream) number */ - unsigned char name[44]; /* ASCII name of item */ + unsigned char name[44]; /* ASCII name of item */ unsigned int index; /* index of item */ }; @@ -809,7 +809,7 @@ struct snd_ctl_elem_info { struct snd_ctl_elem_value { struct snd_ctl_elem_id id; /* W: element ID */ unsigned int indirect: 1; /* W: indirect access - obsoleted */ - union { + union { union { long value[128]; long *value_ptr; /* obsoleted */ @@ -827,15 +827,15 @@ struct snd_ctl_elem_value { unsigned char *data_ptr; /* obsoleted */ } bytes; struct snd_aes_iec958 iec958; - } value; /* RO */ + } value; /* RO */ struct timespec tstamp; - unsigned char reserved[128-sizeof(struct timespec)]; + unsigned char reserved[128-sizeof(struct timespec)]; }; struct snd_ctl_tlv { - unsigned int numid; /* control element numeric identification */ - unsigned int length; /* in bytes aligned to 4 */ - unsigned int tlv[0]; /* first TLV */ + unsigned int numid; /* control element numeric identification */ + unsigned int length; /* in bytes aligned to 4 */ + unsigned int tlv[0]; /* first TLV */ }; #define SNDRV_CTL_IOCTL_PVERSION _IOR('U', 0x00, int) @@ -886,8 +886,8 @@ struct snd_ctl_event { unsigned int mask; struct snd_ctl_elem_id id; } elem; - unsigned char data8[60]; - } data; + unsigned char data8[60]; + } data; }; /* -- 1.7.1

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[alsa-devel] MUTE broken on hda nvhdmi
by VDR User 11 May '10

11 May '10

Hi. Mute seems to be broken since the nvhdmi rewrite. I'm currently using alsa-driver-1.0.23.0.g30ba4.0.g0c8e7.tar.bz2 -- anyone got plans to fix this on their TODO list or could please look into it? Thanks.

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Re: [alsa-devel] USB transfer_buffer allocations on 64bit systems
by Alan Stern 11 May '10

11 May '10

On Tue, 11 May 2010, FUJITA Tomonori wrote: > > > > > At one point we tried an experiment, printing out the buffer and DMA > > > > > addresses. I don't recall seeing anything obviously wrong, but if an > > > > > IOMMU was in use then that might not mean anything. Is it possible > > > > > that the IOMMU mappings sometimes get messed up for addresses above 4 > > > > > GB? > > > > > > > > You mean that an IOMMU could allocate an address above 4GB wrongly? If > > > > so, IIRC, all the IOMMU implementations use dev->dma_mask and > > > > dev->coherent_dma_mask properly. And the DMA address space of the > > > > majority of IOMMUs are limited less than 4GB. > > > > > > The Intel IOMMU code will use dev->dma_mask and dev->coherent_dma_mask > > > properly. It is not limited to 4GiB, but it will tend to give virtual > > > DMA addresses below 4GiB even when a device is capable of more; it'll > > > only give out higher addresses when the address space below 4GiB is > > > exhausted. > > > > What I meant was: Is it possible that the IOMMU code will return a > > virtual DMA address before 4 GB but will somehow forget to actually map > > that address to the data buffer? > > Then, the IOMMU is completely broken. Then, we would get tons of DMA > bugs not only about USB, I guess. So I'm not sure. Yes, you're right about that. > > The problem goes away when Pedro boots with mem=4G. And the dma_mask > > value is set properly (in fact, the ehci-hcd driver currently doesn't > > use 64-bit DMA at all). > > > > If anyone wants to see the debug log entries showing the buffer and DMA > > addresses, they are attached to this email message: > > > > http://marc.info/?l=linux-kernel&m=127076841801054&w=2 > > > > Either the data isn't getting written to the buffer correctly or else > > the buffer isn't getting sent to the device correctly. Can anybody > > suggest a means of determining which is the case? > > I can't say anything about this log that including only DMA addresses. > I'm not familiar with how the USB core does DMA stuff. And the USB > stack design that the USB core does DMA stuff (allocating, mappings, > etc) makes debugging DMA issues really difficult. The DMA stuff is simple enough in this case. The urb->transfer_buffer address is passed to dma_map_single(), and the DMA address it returns is stored in urb->transfer_dma. Those are the two values printed out by the debugging patch. Alan Stern

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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 11 May '10

11 May '10

The following changes since commit 896060c76bdfd8a45eb33b3dd1a8307fe37f6c04: Mark Brown (1): ASoC: Use more idiomatic driver name for WM8731 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 A bunch of minor WM8993/4 enhancements and Peter's invert handling. Mark Brown (4): ASoC: Add register write logging for WM8994 ASoC: Add additional WM hubs DC servo trace ASoC: Reorder power down sequence for WM hubs devices ASoC: Don't restart unconfigured WM8994 FLLs Peter Ujfalusi (1): ASoC: core: Fix for the volume limiting when invert is in use include/sound/soc.h | 23 ++++++++++++----------- sound/soc/codecs/wm8994.c | 5 +++++ sound/soc/codecs/wm_hubs.c | 12 ++++++++---- sound/soc/soc-core.c | 30 +++++++++++++++++++++--------- 4 files changed, 46 insertions(+), 24 deletions(-)

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Re: [alsa-devel] WM8993: amp control, amplifying pops.
by Mark Brown 11 May '10

11 May '10

On Tue, May 11, 2010 at 11:07:38AM +0800, Zhang JieJing wrote: > 2010/5/10 Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Hi, Our kernel version is 2.6.28, but port the wm8993 driver to this > version's kernel. Normally in these situations I would recommend backporting the entire ASoC subsystem - it's fairly well isolated from the rest of the kernel so this tends to be reasonably straightforward. > the amplifying pop is when system resume, the speaker will hear one > short pop noise, > also, this short pop noise will hear the first time driver is opened. When exactly do you hear this noise? The WM8993 driver will not power up the speaker outputs unless they are in use and there are no audio paths configured to the speakers by default. Are you configuring an audio path in the CODEC at system startup? Which mode do you have the WM8993 speaker drivers in - class D or class A/B? It might be helpful to capture all the register writes done when bringing up the WM8993. Also, if you are configuring an audio path from an analogue input you should check that there is no DC offset present in the input signal - if there is one then the WM8993 will reproduce it into the output path, which would result in an audible noise. > In 53242c68333570631a15a69842851b458eca3d99, > there was a line: > > + ret = wm8993_set_fll(codec->dai, 0, wm8993->fll_src, > + fll_fref, fll_fout); > + > but in lastest version of wm8993 driver, wm8993_set_fll will do > nothing we the second parameter(fll_id) is 0, so, I don't know what > this function want to do, > Cound you give me some idea ? This parameter specifies which of the FLLs/PLLs in the device to configure. Since the WM8993 has only one FLL it just ignores this parameter, other devices with multiple FLLs such as the WM8994 need this to select between the FLLs.

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[alsa-devel] snd-hda-intel - wallclk patch
by Jaroslav Kysela 11 May '10

11 May '10

Hi, I would like to ask HDA gurus to check this patch (I will include it to my tree once acked). The patch uses WALLCLK from HDA chips (marked as required in the HDA specification) to check for bogus - too early - interrups which confuses the upper PCM layer (sound skipping issues). More details about patch testing on problematic hardware can be found at: https://bugzilla.kernel.org/show_bug.cgi?id=15912 Thanks, Jaroslav diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 236b4ca..714978f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -174,7 +174,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #define ICH6_GSTS_FSTS (1 << 1) /* flush status */ #define ICH6_REG_INTCTL 0x20 #define ICH6_REG_INTSTS 0x24 -#define ICH6_REG_WALCLK 0x30 +#define ICH6_REG_WALLCLK 0x30 /* 24Mhz source */ #define ICH6_REG_SYNC 0x34 #define ICH6_REG_CORBLBASE 0x40 #define ICH6_REG_CORBUBASE 0x44 @@ -340,8 +340,8 @@ struct azx_dev { unsigned int period_bytes; /* size of the period in bytes */ unsigned int frags; /* number for period in the play buffer */ unsigned int fifo_size; /* FIFO size */ - unsigned long start_jiffies; /* start + minimum jiffies */ - unsigned long min_jiffies; /* minimum jiffies before position is valid */ + unsigned long start_wallclk; /* start + minimum wallclk */ + unsigned long period_wallclk; /* wallclk for period */ void __iomem *sd_addr; /* stream descriptor pointer */ @@ -361,7 +361,6 @@ struct azx_dev { unsigned int opened :1; unsigned int running :1; unsigned int irq_pending :1; - unsigned int start_flag: 1; /* stream full start flag */ /* * For VIA: * A flag to ensure DMA position is 0 @@ -1674,8 +1673,9 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) return err; } - azx_dev->min_jiffies = (runtime->period_size * HZ) / - (runtime->rate * 2); + /* wallclk has 24Mhz clock source */ + azx_dev->period_wallclk = (((runtime->period_size * 24000) / + runtime->rate) * 1000); azx_setup_controller(chip, azx_dev); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1; @@ -1729,14 +1729,15 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) if (s->pcm->card != substream->pcm->card) continue; azx_dev = get_azx_dev(s); - if (rstart) { - azx_dev->start_flag = 1; - azx_dev->start_jiffies = jiffies + azx_dev->min_jiffies; - } - if (start) + if (start) { + azx_dev->start_wallclk = azx_readl(chip, WALLCLK); + if (!rstart) + azx_dev->start_wallclk -= + azx_dev->period_wallclk; azx_stream_start(chip, azx_dev); - else + } else { azx_stream_stop(chip, azx_dev); + } azx_dev->running = start; } spin_unlock(&chip->reg_lock); @@ -1880,12 +1881,13 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) */ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) { + u32 wallclk; unsigned int pos; - if (azx_dev->start_flag && - time_before_eq(jiffies, azx_dev->start_jiffies)) + wallclk = azx_readl(chip, WALLCLK); + if ((wallclk - azx_dev->start_wallclk) < + (azx_dev->period_wallclk * 2) / 3) return -1; /* bogus (too early) interrupt */ - azx_dev->start_flag = 0; pos = azx_get_position(chip, azx_dev); if (chip->position_fix == POS_FIX_AUTO) { @@ -1899,13 +1901,12 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) chip->position_fix = POS_FIX_POSBUF; } - if (!bdl_pos_adj[chip->dev_index]) - return 1; /* no delayed ack */ if (WARN_ONCE(!azx_dev->period_bytes, "hda-intel: zero azx_dev->period_bytes")) return 0; /* this shouldn't happen! */ if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) return 0; /* NG - it's below the period boundary */ + azx_dev->start_wallclk = wallclk; return 1; /* OK, it's fine */ } @@ -1915,7 +1916,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) static void azx_irq_pending_work(struct work_struct *work) { struct azx *chip = container_of(work, struct azx, irq_pending_work); - int i, pending; + int i, pending, ok; if (!chip->irq_pending_warned) { printk(KERN_WARNING @@ -1934,11 +1935,14 @@ static void azx_irq_pending_work(struct work_struct *work) !azx_dev->substream || !azx_dev->running) continue; - if (azx_position_ok(chip, azx_dev)) { + ok = azx_position_ok(chip, azx_dev); + if (ok > 0) { azx_dev->irq_pending = 0; spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(azx_dev->substream); spin_lock(&chip->reg_lock); + } else if (ok < 0) { + pending = 0; /* too early */ } else pending++; } ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [PATCH] ASoC: core: Fix for the volume limiting when invert is in use
by Peter Ujfalusi 11 May '10

11 May '10

If the register for the volume needs invert, than the inversion need to be done from the chip maximum, and not from the platform dependent limit. Introduce soc_mixer_control.platform_max value, which initially equals to chip maximum. The snd_soc_limit_volume function only modify the platform_max, all volsw_info call returns this as well. The .max value holds the chip default (maximum), and it is used for the inversion, if it is needed. Additional check in the volsw_info call has been added to check the validity of the platform_max in case, when custom macros used by codec drivers are not initializing it correctly. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello, Normal, and inverted volume controls now can be limited, and both behave correctly. Has been tested on twl4030, tpa6130a2, and tlv320dac33 codec drivers. The patch is generated against Takashi's topic/asoc branch, but I think it should be fine with Mark's tree as well. Peter include/sound/soc.h | 23 ++++++++++++----------- sound/soc/soc-core.c | 30 +++++++++++++++++++++--------- 2 files changed, 33 insertions(+), 20 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9f306f0..aac97d7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -30,10 +30,10 @@ #define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = xshift, .rshift = xshift, .max = xmax, \ - .invert = xinvert}) + .platform_max = xmax, .invert = xinvert}) #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .max = xmax, .invert = xinvert}) + {.reg = xreg, .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,\ @@ -53,14 +53,14 @@ .put = snd_soc_put_volsw, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .info = snd_soc_info_volsw_2r, \ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -70,7 +70,7 @@ .put = snd_soc_put_volsw, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = shift_left, .rshift = shift_right,\ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -80,7 +80,7 @@ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ @@ -89,7 +89,8 @@ .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ .put = snd_soc_put_volsw_s8, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .min = xmin, .max = xmax} } + {.reg = xreg, .min = xmin, .max = xmax, \ + .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmax, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .max = xmax, .texts = xtexts } @@ -126,7 +127,7 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -146,7 +147,7 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -157,7 +158,7 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .max = xmax, .invert = xinvert} } + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ @@ -569,7 +570,7 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { - int min, max; + int min, max, platform_max; unsigned int reg, rreg, shift, rshift, invert; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4079223..9340dcd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1954,18 +1954,22 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; + int platform_max; unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; - if (max == 1 && !strstr(kcontrol->id.name, " Volume")) + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = shift == rshift ? 1 : 2; uinfo->value.integer.min = 0; - uinfo->value.integer.max = max; + uinfo->value.integer.max = platform_max; return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw); @@ -2063,16 +2067,20 @@ int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; + int platform_max; - if (max == 1 && !strstr(kcontrol->id.name, " Volume")) + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; uinfo->value.integer.min = 0; - uinfo->value.integer.max = max; + uinfo->value.integer.max = platform_max; return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r); @@ -2173,13 +2181,17 @@ int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int max = mc->max; + int platform_max; int min = mc->min; + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; uinfo->value.integer.min = 0; - uinfo->value.integer.max = max-min; + uinfo->value.integer.max = platform_max - min; return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); @@ -2268,7 +2280,7 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec, if (found) { mc = (struct soc_mixer_control *)kctl->private_value; if (max <= mc->max) { - mc->max = max; + mc->platform_max = max; ret = 0; } } -- 1.7.1

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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 11 May '10

11 May '10

The following changes since commit d0bbc24d2ab78d9efc61bc80cfc85f59626ba5af: Jassi Brar (1): ASoC: SMDK64XX: Switch to IISv4 CPU driver are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Mark Brown (8): ASoC: Remove unneeded suspend bias managment from CODEC drivers ASoC: Remove unused DAPM suspend flag ASoC: Refactor DAPM suspend handling ASoC: Support leaving paths enabled over system suspend ASoC: Allow active paths from the GSM modem while the GTA02 is suspended ASoC: Allow DAI links to be kept active over suspend ASoC: Refactor WM8731 regulator management into bias management ASoC: Use more idiomatic driver name for WM8731 drivers/mfd/wm831x-irq.c | 30 ++++++++++ include/sound/soc-dapm.h | 3 +- include/sound/soc.h | 3 + sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/stac9766.c | 3 - sound/soc/codecs/tlv320aic23.c | 1 - sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 3 - sound/soc/codecs/twl4030.c | 1 - sound/soc/codecs/twl6040.c | 1 - sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wm8350.c | 3 - sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 54 +++++++++-------- sound/soc/codecs/wm8903.c | 1 - sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm9712.c | 3 - sound/soc/codecs/wm9713.c | 3 - sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 8 +++ sound/soc/soc-core.c | 73 ++++++++++++++++++++++-- sound/soc/soc-dapm.c | 92 +++++++++++++++++++----------- 27 files changed, 206 insertions(+), 94 deletions(-)

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[alsa-devel] [PATCH] hda_intel: Fixes distorted recording on US15W chipset
by Shahin Ghazinouri 11 May '10

11 May '10

The HDA controller in US15W (Poulsbo) reports inaccurate position values for capture streams when using the LPIB read method, resulting in distorted recordings. However, using the position buffer is broken for playback streams, resulting in a fallback to the LPIB method with the current driver. This patch works around the issue by independently detecting the read position method for capture and playback streams. The patch will not have any effect if the position fix method is explicitly set. Signed-off-by: Shahin Ghazinouri <shahin.ghazinouri(a)pelagicore.com> --- --- linux-2.6.34-rc6/sound/pci/hda/hda_intel.c 2010-04-30 05:02:05.000000000 +0200 +++ linux-2.6.34-rc6_patch/sound/pci/hda/hda_intel.c 2010-04-30 11:31:57.000000000 +0200 @@ -425,7 +425,8 @@ struct snd_dma_buffer posbuf; /* flags */ - int position_fix; + int playback_position_fix; + int capture_position_fix; int poll_count; unsigned int running :1; unsigned int initialized :1; @@ -1302,8 +1303,10 @@ azx_sd_writel(azx_dev, SD_BDLPU, upper_32_bits(azx_dev->bdl.addr)); /* enable the position buffer */ - if (chip->position_fix == POS_FIX_POSBUF || - chip->position_fix == POS_FIX_AUTO || + if (chip->capture_position_fix == POS_FIX_POSBUF || + chip->capture_position_fix == POS_FIX_AUTO || + chip->playback_position_fix == POS_FIX_POSBUF || + chip->playback_position_fix == POS_FIX_AUTO || chip->via_dmapos_patch) { if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE)) azx_writel(chip, DPLBASE, @@ -1843,13 +1846,24 @@ if (chip->via_dmapos_patch) pos = azx_via_get_position(chip, azx_dev); - else if (chip->position_fix == POS_FIX_POSBUF || - chip->position_fix == POS_FIX_AUTO) { - /* use the position buffer */ - pos = le32_to_cpu(*azx_dev->posbuf); + else if (azx_dev->index < chip->playback_index_offset) { + /* Capture stream */ + if (chip->capture_position_fix == POS_FIX_POSBUF || + chip->capture_position_fix == POS_FIX_AUTO) + /* use the position buffer */ + pos = le32_to_cpu(*azx_dev->posbuf); + else + /* read LPIB */ + pos = azx_sd_readl(azx_dev, SD_LPIB); } else { - /* read LPIB */ - pos = azx_sd_readl(azx_dev, SD_LPIB); + /* Playback stream */ + if (chip->playback_position_fix == POS_FIX_POSBUF || + chip->playback_position_fix == POS_FIX_AUTO) + /* use the position buffer */ + pos = le32_to_cpu(*azx_dev->posbuf); + else + /* read LPIB */ + pos = azx_sd_readl(azx_dev, SD_LPIB); } if (pos >= azx_dev->bufsize) pos = 0; @@ -1884,15 +1898,30 @@ azx_dev->start_flag = 0; pos = azx_get_position(chip, azx_dev); - if (chip->position_fix == POS_FIX_AUTO) { - if (!pos) { - printk(KERN_WARNING - "hda-intel: Invalid position buffer, " - "using LPIB read method instead.\n"); - chip->position_fix = POS_FIX_LPIB; - pos = azx_get_position(chip, azx_dev); - } else - chip->position_fix = POS_FIX_POSBUF; + if (azx_dev->index < chip->playback_index_offset) { + /* Capture stream */ + if (chip->capture_position_fix == POS_FIX_AUTO) { + if (!pos) { + printk(KERN_WARNING + "hda-intel: Invalid position buffer, " + "using LPIB read method instead.\n"); + chip->capture_position_fix = POS_FIX_LPIB; + pos = azx_get_position(chip, azx_dev); + } else + chip->capture_position_fix = POS_FIX_POSBUF; + } + } else { + /* Playback stream */ + if (chip->playback_position_fix == POS_FIX_AUTO) { + if (!pos) { + printk(KERN_WARNING + "hda-intel: Invalid position buffer, " + "using LPIB read method instead.\n"); + chip->playback_position_fix = POS_FIX_LPIB; + pos = azx_get_position(chip, azx_dev); + } else + chip->playback_position_fix = POS_FIX_POSBUF; + } } if (!bdl_pos_adj[chip->dev_index]) @@ -2431,7 +2460,10 @@ chip->dev_index = dev; INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work); - chip->position_fix = check_position_fix(chip, position_fix[dev]); + chip->capture_position_fix = + check_position_fix(chip, position_fix[dev]); + chip->playback_position_fix = + check_position_fix(chip, position_fix[dev]); check_probe_mask(chip, dev); chip->single_cmd = single_cmd;

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[alsa-devel] [PATCH] ALSA: hda: Fix 0 dB for Lenovo models using Conexant CX20549 (Venice)
by Daniel Chen 11 May '10

11 May '10

Hi, Attached please find a patch against sound-2.6 (also applicable to stable trees) that fixes the reported distortion for Lenovo models on CX20549. Best, -Dan

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