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

[alsa-devel] [PATCH 0/2] RFC: support for audio wall clock
by Pierre-Louis Bossart 15 Jun '12

15 Jun '12

Attached a set 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. 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_lib.c | 15 ++++++- sound/core/pcm_native.c | 2 + sound/pci/hda/hda_intel.c | 106 ++++++++++++++++++++++++++++++++++++++++++++- 5 files changed, 128 insertions(+), 4 deletions(-) -- 1.7.6.5

4 11

[alsa-devel] [PATCH] pcm: support for audio wallclock
by Pierre-Louis Bossart 15 Jun '12

15 Jun '12

add new snd_pcm_status_get_htstamp_wallclk() routine to query the audio timestamps read from the wallclock. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/pcm.h | 1 + include/sound/asound.h | 7 ++++++- src/pcm/pcm.c | 16 ++++++++++++++++ 3 files changed, 23 insertions(+), 1 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 4997557..325c0ea 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -858,6 +858,7 @@ 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..c186fd6 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -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 +427,10 @@ 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 */ + union { + unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* audio wall clock timestamp */ + } ext; }; struct sndrv_pcm_mmap_status { @@ -434,6 +438,7 @@ struct sndrv_pcm_mmap_status { int pad1; /* Needed for 64 bit alignment */ sndrv_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ + struct timespec audio_tstamp; /* audio wall clock timestamp */ int suspended_state; /* RO: suspended stream state */ }; diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 18b43b3..2221865 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -6213,6 +6213,22 @@ 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 + */ +#ifndef DOXYGEN +void INTERNAL(snd_pcm_status_get_htstamp_wallclk)(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +#else +void snd_pcm_status_get_htstamp_wallclk(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +#endif +{ + assert(obj && ptr); + *ptr = obj->ext.audio_tstamp; +} +use_default_symbol_version(__snd_pcm_status_get_htstamp_wallclk, snd_pcm_status_get_htstamp_wallclk, ALSA_0.9.0rc8); /* FIXME, no idea why this is needed */ + +/** * \brief Get delay from a PCM status container (see #snd_pcm_delay) * \return Delay in frames * -- 1.7.6.5

2 2

[alsa-devel] [PATCH] ASoC: omap-mcpdm: Convert to use devm_*
by Peter Ujfalusi 15 Jun '12

15 Jun '12

Switch to use devm_* te make the probe/remove code more cleaner. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-mcpdm.c | 52 ++++++++++-------------------------------- 1 files changed, 13 insertions(+), 39 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 59d47ab..b5e23af 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -445,9 +445,8 @@ static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) { struct omap_mcpdm *mcpdm; struct resource *res; - int ret = 0; - mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + mcpdm = devm_kzalloc(&pdev->dev, sizeof(struct omap_mcpdm), GFP_KERNEL); if (!mcpdm) return -ENOMEM; @@ -456,55 +455,30 @@ static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) mutex_init(&mcpdm->mutex); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) { - dev_err(&pdev->dev, "no resource\n"); - goto err_res; - } + if (res == NULL) + return -ENOMEM; - if (!request_mem_region(res->start, resource_size(res), "McPDM")) { - ret = -EBUSY; - goto err_res; - } + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), "McPDM")) + return -EBUSY; - mcpdm->io_base = ioremap(res->start, resource_size(res)); - if (!mcpdm->io_base) { - ret = -ENOMEM; - goto err_iomap; - } + mcpdm->io_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!mcpdm->io_base) + return -ENOMEM; mcpdm->irq = platform_get_irq(pdev, 0); - if (mcpdm->irq < 0) { - ret = mcpdm->irq; - goto err_irq; - } + if (mcpdm->irq < 0) + return mcpdm->irq; mcpdm->dev = &pdev->dev; - ret = snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); - if (!ret) - return 0; - -err_irq: - iounmap(mcpdm->io_base); -err_iomap: - release_mem_region(res->start, resource_size(res)); -err_res: - kfree(mcpdm); - return ret; + return snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); } static int __devexit asoc_mcpdm_remove(struct platform_device *pdev) { - struct omap_mcpdm *mcpdm = platform_get_drvdata(pdev); - struct resource *res; - snd_soc_unregister_dai(&pdev->dev); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - iounmap(mcpdm->io_base); - release_mem_region(res->start, resource_size(res)); - - kfree(mcpdm); return 0; } -- 1.7.8.6

2 1

Re: [alsa-devel] [PATCH 1/1] hda_intel: activate COMBO mode for Intel client chipsets
by Takashi Iwai 15 Jun '12

15 Jun '12

At Thu, 14 Jun 2012 14:23:53 -0700, Seth Heasley wrote: > > 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. > > Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> Applied now. Thanks. Takashi > --- > 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 >

1 0

[alsa-devel] McBSP in TDM for audio codec
by learnom learnom 15 Jun '12

15 Jun '12

Hi All, I have to support 4 channels in I2S format in TDM mode. Processor: OMAP 3703 If any one have any idea on how to go about this. Please let me know. Thanks, learnom

2 3

[alsa-devel] Anyone using alsactl preinit / postinit scripts?
by Takashi Iwai 14 Jun '12

14 Jun '12

Hi, is anyone already using preinit / postinit scripts of alsactl init? I'm thinking of appliny the patch below, but before that, I'd like to make sure that *.conf is the appropriate extension. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] alsactl: Read only *.conf files when a directory is passed via INCLUDE When alsactl init is invoked and a directory path is passed to INCLUDE command in the config file, read only *.conf files in that directory. This will avoid reading backup files or invalid files that have been created accidentally. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/init_parse.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/alsactl/init_parse.c b/alsactl/init_parse.c index 51b515c..83d12ba 100644 --- a/alsactl/init_parse.c +++ b/alsactl/init_parse.c @@ -1502,12 +1502,16 @@ static int parse_line(struct space *space, char *line, size_t linesize) if (dir) { count = strlen(string); while ((dirent = readdir(dir)) != NULL) { - if (strcmp(dirent->d_name, ".") == 0 || - strcmp(dirent->d_name, "..") == 0) + const char *name = dirent->d_name; + const char *ext; + if (*name == '.') + continue; + ext = strrchr(name, ','); + if (!ext || strcmp(ext, ".conf")) continue; string[count] = '\0'; strlcat(string, "/", sizeof(string)); - strlcat(string, dirent->d_name, sizeof(string)); + strlcat(string, name, sizeof(string)); space->go_to = NULL; space->rootdir = new_root_dir(string); if (space->rootdir) { -- 1.7.10.3

2 2

[alsa-devel] [PATCH] ALSA: HDA: Create phantom jacks for fixed inputs and outputs
by David Henningsson 14 Jun '12

14 Jun '12

PulseAudio sometimes have difficulties knowing that there is a "Speaker" or "Internal Mic", if they have no individual volume controls or selectors. As a result, only e g "Headphone" might be created for a laptop, but no "Speaker". To help out, create phantom jacks (that are always present, at least for now) for "Speaker", "Internal Mic" etc, in case we detect them. The naming convention is e g "Speaker Phantom Jack". In order not to pollute the /dev/input namespace with even more devices, these are added to the kcontrols only, not the input devices. An implementation note: The possibility to override how to actually do jack sensing opens up for more advanced usage of the jack infrastructure in the future (e g letting the headphone jack detection set presence status for both headphone and mic in case of a headset jack, among other things). Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_jack.c | 64 ++++++++++++++++++++++++++++++++++------------ sound/pci/hda/hda_jack.h | 5 ++++ 2 files changed, 52 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 2dd1c11..c5569f1 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -133,6 +133,13 @@ static void jack_detect_update(struct hda_codec *codec, } } +static void jack_detect_always_present(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + jack->pin_sense = AC_PINSENSE_PRESENCE; + jack->jack_dirty = 0; +} + /** * snd_hda_set_dirty_all - Mark all the cached as dirty * @@ -162,7 +169,8 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) { struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); if (jack) { - jack_detect_update(codec, jack); + if (jack->update_func) + jack->update_func(codec, jack); return jack->pin_sense; } return read_pin_sense(codec, nid); @@ -215,7 +223,8 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) for (i = 0; i < codec->jacktbl.used; i++, jack++) if (jack->nid) { - jack_detect_update(codec, jack); + if (jack->update_func) + jack->update_func(codec, jack); if (!jack->kctl) continue; state = get_jack_plug_state(jack->pin_sense); @@ -264,8 +273,8 @@ static void hda_free_jack_priv(struct snd_jack *jack) * This assigns a jack-detection kctl to the given pin. The kcontrol * will have the given name and index. */ -int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, int idx) +static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, + const char *name, int idx, bool phantom_jack) { struct hda_jack_tbl *jack; struct snd_kcontrol *kctl; @@ -283,19 +292,33 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, if (err < 0) return err; jack->kctl = kctl; + if (phantom_jack) + jack->update_func = jack_detect_always_present; + else + jack->update_func = jack_detect_update; + state = snd_hda_jack_detect(codec, nid); snd_kctl_jack_report(codec->bus->card, kctl, state); #ifdef CONFIG_SND_HDA_INPUT_JACK - jack->type = get_input_jack_type(codec, nid); - err = snd_jack_new(codec->bus->card, name, jack->type, &jack->jack); - if (err < 0) - return err; - jack->jack->private_data = jack; - jack->jack->private_free = hda_free_jack_priv; - snd_jack_report(jack->jack, state ? jack->type : 0); + if (!phantom_jack) { + jack->type = get_input_jack_type(codec, nid); + err = snd_jack_new(codec->bus->card, name, jack->type, + &jack->jack); + if (err < 0) + return err; + jack->jack->private_data = jack; + jack->jack->private_free = hda_free_jack_priv; + snd_jack_report(jack->jack, state ? jack->type : 0); + } #endif return 0; } + +int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, + const char *name, int idx) +{ + return __snd_hda_jack_add_kctl(codec, nid, name, idx, false); +} EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, @@ -305,25 +328,32 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, unsigned int def_conf, conn; char name[44]; int idx, err; + bool phantom_jack; if (!nid) return 0; - if (!is_jack_detectable(codec, nid)) - return 0; def_conf = snd_hda_codec_get_pincfg(codec, nid); conn = get_defcfg_connect(def_conf); - if (conn != AC_JACK_PORT_COMPLEX) + if (conn == AC_JACK_PORT_NONE) return 0; + phantom_jack = (conn != AC_JACK_PORT_COMPLEX) || + !is_jack_detectable(codec, nid); snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx); + if (phantom_jack) + /* Example final name: "Internal Mic Phantom Jack" */ + strncat(name, " Phantom", sizeof(name) - strlen(name) - 1); if (!strcmp(name, lastname) && idx == *lastidx) idx++; - strncpy(lastname, name, 44); + strncpy(lastname, name, sizeof(name)); *lastidx = idx; - err = snd_hda_jack_add_kctl(codec, nid, name, idx); + err = __snd_hda_jack_add_kctl(codec, nid, name, idx, phantom_jack); if (err < 0) return err; - return snd_hda_jack_detect_enable(codec, nid, 0); + + if (!phantom_jack) + return snd_hda_jack_detect_enable(codec, nid, 0); + return 0; } /** diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 8ae5246..4055236 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -13,6 +13,10 @@ #define __SOUND_HDA_JACK_H struct auto_pin_cfg; +struct hda_jack_tbl; + +typedef void (*snd_hda_jack_detect_update)(struct hda_codec *codec, + struct hda_jack_tbl *jack); struct hda_jack_tbl { hda_nid_t nid; @@ -24,6 +28,7 @@ struct hda_jack_tbl { unsigned int jack_detect:1; /* capable of jack-detection? */ unsigned int jack_dirty:1; /* needs to update? */ struct snd_kcontrol *kctl; /* assigned kctl for jack-detection */ + snd_hda_jack_detect_update update_func; /* function to call to perform jack detection */ #ifdef CONFIG_SND_HDA_INPUT_JACK int type; struct snd_jack *jack; -- 1.7.9.5

2 5

Re: [alsa-devel] [PATCH 1/1] hda_intel: switch Intel chipsets to COMBO position_fix
by Takashi Iwai 14 Jun '12

14 Jun '12

At Wed, 13 Jun 2012 17:58:57 +0000, Heasley, Seth wrote: > > >> This patch implements the COMBO position_fix for recent Intel client > >chipsets. > >> > >> Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> > > > >Thanks for the patch. > >But could you give a bit more context why this change is needed? > > The COMBO method was recommended for Intel chipsets by a certain audio maintainer. ;) It was also tested on the most recent Intel chips because we were seeing warning messages when using DMA positioning ("Invalid position buffer, using LPIB read method instead."). Could you resend the patch with such some reasonable comments in the changelog? thanks, Takashi > >> --- > >> 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 > >> >

1 0

[alsa-devel] Does amixer have commands to turn on / off Codec TWL4030 analog microphone bias voltage output?
by Philip Chu 14 Jun '12

14 Jun '12

I am using Linux kernel 3.0 and a Logic PD DM3530 SOM LV, looking for the amixer commands to turn on / off analog microphone bias supply. Following lines do not have the bias needed for Headset MIC input: amixer set 'Analog Left AUXL' nocap amixer set 'Analog Right AUXR' nocap amixer set 'Analog Left Main Mic' nocap amixer set 'Analog Left Headset Mic' cap Anybody can help? Thanks, Philip Chu principal software engineer LogicPD www.logicpd.com 5 Clock Tower Place, Suite 215 Maynard, MA 01754 Phone: 978-243-2450 Fax: 978-461-1598

1 0

Re: [alsa-devel] [PATCH v2 0/5] mmp audio support
by Arnd Bergmann 14 Jun '12

14 Jun '12

On Monday 11 June 2012, Zhangfei Gao wrote: > > v1->v2: > modify accroingly with suggestion from Vinod and Mark > remove deprecated chan->private and dma.h > remove prep_slave_sg > remove module_init and move operation to probe > misc change It seems that this driver is missing DT bindings, which really need to get added. The brownstone platform is currently supported using both DT and ATAG probing, but we want to remove the ATAG variant eventually, so adding a new driver for that platform without support for DT booting is a step in the wrong direction. Arnd -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1