February 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/7] ASoC: neo1973_wm8753: Remove scenario management code.
by Lars-Peter Clausen 08 Feb '11

08 Feb '11

It has been proven to be inflexible to do scenario management in kernel space. Since actual neo1973 board support has not been merged in mainline and this patch has been in the neo1973 tree for some time now it should be safe to remove this functionality without breaking existing userspace. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/neo1973_wm8753.c | 142 +---------------------------------- 1 files changed, 5 insertions(+), 137 deletions(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index d3cd688..1ea6e61 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -37,17 +37,6 @@ #include "dma.h" #include "s3c24xx-i2s.h" -/* define the scenarios */ -#define NEO_AUDIO_OFF 0 -#define NEO_GSM_CALL_AUDIO_HANDSET 1 -#define NEO_GSM_CALL_AUDIO_HEADSET 2 -#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3 -#define NEO_STEREO_TO_SPEAKERS 4 -#define NEO_STEREO_TO_HEADPHONES 5 -#define NEO_CAPTURE_HANDSET 6 -#define NEO_CAPTURE_HEADSET 7 -#define NEO_CAPTURE_BLUETOOTH 8 - static struct snd_soc_card neo1973; static struct i2c_client *i2c; @@ -224,113 +213,6 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; -static int neo1973_scenario; - -static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = neo1973_scenario; - return 0; -} - -static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - pr_debug("Entered %s\n", __func__); - - switch (neo1973_scenario) { - case NEO_AUDIO_OFF: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_HANDSET: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_HEADSET: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - default: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - } - - snd_soc_dapm_sync(dapm); - - return 0; -} - -static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - pr_debug("Entered %s\n", __func__); - - if (neo1973_scenario == ucontrol->value.integer.value[0]) - return 0; - - neo1973_scenario = ucontrol->value.integer.value[0]; - set_scenario_endpoints(codec, neo1973_scenario); - return 1; -} - static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; static void lm4857_write_regs(void) @@ -454,22 +336,6 @@ static const struct soc_enum lm4857_mode_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), }; -static const char *neo_scenarios[] = { - "Off", - "GSM Handset", - "GSM Headset", - "GSM Bluetooth", - "Speakers", - "Headphones", - "Capture Handset", - "Capture Headset", - "Capture Bluetooth" -}; - -static const struct soc_enum neo_scenario_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), -}; - static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); @@ -482,8 +348,6 @@ static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { lm4857_get_reg, lm4857_set_reg, mono_tlv), SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], lm4857_get_mode, lm4857_set_mode), - SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], - neo1973_get_scenario, neo1973_set_scenario), SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, lm4857_get_reg, lm4857_set_reg), SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, @@ -520,7 +384,11 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(wm8753_dapm_widgets)); /* set endpoints to default mode */ - set_scenario_endpoints(codec, NEO_AUDIO_OFF); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); /* add neo1973 specific controls */ err = snd_soc_add_controls(codec, wm8753_neo1973_controls, -- 1.7.2.3

3 19

Re: [alsa-devel] [PATCH v8 05/12] media: Entity use count
by Hans Verkuil 08 Feb '11

08 Feb '11

On Friday, February 04, 2011 13:34:02 Sakari Ailus wrote: > Hi, > > And many thanks for the comments! > > Hans Verkuil wrote: > ... > >> diff --git a/include/media/media-entity.h b/include/media/media-entity.h > >> index b82f824..114541a 100644 > >> --- a/include/media/media-entity.h > >> +++ b/include/media/media-entity.h > >> @@ -81,6 +81,8 @@ struct media_entity { > >> struct media_pad *pads; /* Pads array (num_pads elements) */ > >> struct media_link *links; /* Links array (max_links elements)*/ > >> > >> + int use_count; /* Use count for the entity. */ > > > > Isn't unsigned better? > > Could be. The result, though, will be slightly more difficult checking > for bad use count --- which always is a driver bug. > > me->use_count += change; > WARN_ON(me->use_count < 0); > > we must do something like this: > > if (change < 0) > WARN_ON(me->use_count < (unsigned)-change); > me->use_count += change; > > I'd perhaps also go with unsigned int; the choice for signed was made > mainly since the above check and with signed int the check was more trivial. I saw this WARN_ON as well. I think there is a good reason for that WARN_ON, but I think a comment in the header explaining why it is an int will be useful. If I trip over it, then others will as well :-) Regards, Hans > > Regards, > > -- Hans Verkuil - video4linux developer - sponsored by Cisco

2 1

[alsa-devel] [patch] ALSA: soc-cache: dereferencing before checking
by Dan Carpenter 08 Feb '11

08 Feb '11

The patch c358e640a66 "ASoC: soc-cache: Add trace event for snd_soc_cache_sync()" introduced a dereference of "codec->cache_ops" before we had checked it for NULL. I pulled the check forward, and then pulled everything in an indent level. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index db66dc4..5d76da4 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1609,24 +1609,23 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return 0; } + if (!codec->cache_ops || !codec->cache_ops->sync) + return -EINVAL; + if (codec->cache_ops->name) name = codec->cache_ops->name; else name = "unknown"; - if (codec->cache_ops && codec->cache_ops->sync) { - if (codec->cache_ops->name) - dev_dbg(codec->dev, "Syncing %s cache for %s codec\n", - codec->cache_ops->name, codec->name); - trace_snd_soc_cache_sync(codec, name, "start"); - ret = codec->cache_ops->sync(codec); - if (!ret) - codec->cache_sync = 0; - trace_snd_soc_cache_sync(codec, name, "end"); - return ret; - } - - return -EINVAL; + if (codec->cache_ops->name) + dev_dbg(codec->dev, "Syncing %s cache for %s codec\n", + codec->cache_ops->name, codec->name); + trace_snd_soc_cache_sync(codec, name, "start"); + ret = codec->cache_ops->sync(codec); + if (!ret) + codec->cache_sync = 0; + trace_snd_soc_cache_sync(codec, name, "end"); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync);

4 3

[alsa-devel] [PATCH] ASoC: CS4271: Move Chip Select control out of the CODEC code.
by Alexander Sverdlin 08 Feb '11

08 Feb '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Move Chip Select control out of the CODEC code for CS4271. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- include/sound/cs4271.h | 1 - sound/soc/codecs/cs4271.c | 22 +++------------------- 2 files changed, 3 insertions(+), 20 deletions(-) diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index 16f8d32..50a059e 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -19,7 +19,6 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ - int gpio_disable; /* GPIO that disable serial bus, if any */ }; #endif /* __CS4271_H */ diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 9c5b7db..1791796 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -441,22 +441,11 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; - int gpio_disable = -EINVAL; codec->control_data = cs4271->control_data; - if (cs4271plat) { - if (gpio_is_valid(cs4271plat->gpio_nreset)) - gpio_nreset = cs4271plat->gpio_nreset; - if (gpio_is_valid(cs4271plat->gpio_disable)) - gpio_disable = cs4271plat->gpio_disable; - } - - if (gpio_disable >= 0) - if (gpio_request(gpio_disable, "CS4271 Disable")) - gpio_disable = -EINVAL; - if (gpio_disable >= 0) - gpio_direction_output(gpio_disable, 0); + if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) + gpio_nreset = cs4271plat->gpio_nreset; if (gpio_nreset >= 0) if (gpio_request(gpio_nreset, "CS4271 Reset")) @@ -471,7 +460,6 @@ static int cs4271_probe(struct snd_soc_codec *codec) } cs4271->gpio_nreset = gpio_nreset; - cs4271->gpio_disable = gpio_disable; /* * In case of I2C, chip address specified in board data. @@ -509,10 +497,9 @@ static int cs4271_probe(struct snd_soc_codec *codec) static int cs4271_remove(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - int gpio_nreset, gpio_disable; + int gpio_nreset; gpio_nreset = cs4271->gpio_nreset; - gpio_disable = cs4271->gpio_disable; if (gpio_is_valid(gpio_nreset)) { /* Set codec to the reset state */ @@ -520,9 +507,6 @@ static int cs4271_remove(struct snd_soc_codec *codec) gpio_free(gpio_nreset); } - if (gpio_is_valid(gpio_disable)) - gpio_free(gpio_disable); - return 0; };

4 3

Re: [alsa-devel] issue with AD1989A HDA codec and alsamixer
by Raymond Yau 08 Feb '11

08 Feb '11

2011/2/7 Vincent Meserette <vmeserette(a)aldebaran-robotics.com> > > I found from where came my problem with the capture gains by adding some > logs in the hda_codec.c file. > When I set one of the capture gain, the put_vol_mute function well write in > the AMP_GAIN_MUTE verb of the corresponding ADC selector (0x0c, 0x0d or 0x0e > widget) but the mute bit is always set to 1. > Indeed, the value associated to the AMP_GAIN_MUTE verb is formated as folow > : > bits 0 to 6 : gain > bit 7 : mute > bits 8 to 11 : index > bit 12 : set right amp > bit 13 : set left amp > bit 14 : set input amp > bit 15 : set output amp > > and alsa sets the value of this register from a080 / b080 to a0b6 / b0b6 > when i change the gain from -58.5 dB to 22.5dB > > Do you know how alsa and thus the alsamixer determine the minimum and > maximum values for the capture gain, and how to say to alsa the the gain is > only a 7bits step value ? > Take a look at HDA Specification 7.3.4.10 Amplifier Capabilities StepSize (7 bits) indicates the size of each step in the gain range. Each individual step may be 0-32 dB specified in 0.25-dB steps. A value of 0 indicates 0.25-dB steps, while a value of 127d indicates a 32-dB step. NumSteps (7 bits) indicates the number of steps in the gain range. There may be from 1 to 128 steps in the amplifier gain range. (0d means 1 step, 127d means 128 steps). A value of 0 (1 step) means that the gain is fixed and may not be changed. Offset (7 bits) indicates which step is 0 dB. If there are two or more steps, one of the step values must correspond to a value of 0 dB. The “Offset” value reflects the value which, if programmed in to the Amplifier Gain control, would result in a gain of 0 dB. Mute Capable (1 bit) reports if the respective amplifier is capable of muting. Muting implies a –infinity gain (no sound passes), but the actual performance is determined by the hardware. Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Input Source", index=0, device=0 Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0x25 0x25] Connection: 10 0x38 0x39* 0x3a 0x3b 0x3c 0x18 0x24 0x25 0x3d 0x20 Node 0x0d [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Capture Volume", index=1, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Capture Switch", index=1, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Input Source", index=1, device=0 Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0xa7 0xa7] Connection: 10 0x38 0x39* 0x3a 0x3b 0x3c 0x18 0x24 0x25 0x3d 0x20 Node 0x0e [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Capture Volume", index=2, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Capture Switch", index=2, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Input Source", index=2, device=0 Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0xa7 0xa7] Connection: 10 0x38 0x39* 0x3a 0x3b 0x3c 0x18 0x24 0x25 0x3d 0x20

1 0

Re: [alsa-devel] [PATCH] [ALSA] Enable the E-MU 0204 USB
by Takashi Iwai 08 Feb '11

08 Feb '11

At Tue, 8 Feb 2011 01:22:36 -0500, jteich3(a)gmail.com wrote: > > From: Joseph Teichman <josteich(a)gmail.com> > > > Signed-off-by: Joseph Teichman <josteich(a)gmail.com> Applied now. Thanks! Takashi > diff --git a/usb/mixer.c b/usb/mixer.c > index 7df89b3..85af605 100644 > --- a/usb/mixer.c > +++ b/usb/mixer.c > @@ -95,7 +95,7 @@ enum { > }; > > > -/*E-mu 0202(0404) eXtension Unit(XU) control*/ > +/*E-mu 0202/0404/0204 eXtension Unit(XU) control*/ > enum { > USB_XU_CLOCK_RATE = 0xe301, > USB_XU_CLOCK_SOURCE = 0xe302, > @@ -1566,7 +1566,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw > cval->initialized = 1; > } else { > if (type == USB_XU_CLOCK_RATE) { > - /* E-Mu USB 0404/0202/TrackerPre > + /* E-Mu USB 0404/0202/TrackerPre/0204 > * samplerate control quirk > */ > cval->min = 0; > diff --git a/usb/quirks-table.h b/usb/quirks-table.h > index 3599987..921a86f 100644 > --- a/usb/quirks-table.h > +++ b/usb/quirks-table.h > @@ -79,6 +79,13 @@ > .idProduct = 0x3f0a, > .bInterfaceClass = USB_CLASS_AUDIO, > }, > +{ > + /* E-Mu 0204 USB */ > + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, > + .idVendor = 0x041e, > + .idProduct = 0x3f19, > + .bInterfaceClass = USB_CLASS_AUDIO, > +}, > > /* > * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface > diff --git a/usb/quirks.c b/usb/quirks.c > index cf8bf08..e314cdb 100644 > --- a/usb/quirks.c > +++ b/usb/quirks.c > @@ -532,7 +532,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat > } > > /* > - * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, > + * For E-Mu 0404USB/0202USB/TrackerPre/0204 sample rate should be set for device, > * not for interface. > */ > > @@ -589,6 +589,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, > case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ > case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ > case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ > + case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ > set_format_emu_quirk(subs, fmt); > break; > } > -- > 1.7.0.4 >

1 0

[alsa-devel] [PATCH] [ALSA] Enable the E-MU 0204 USB
by josteich＠gmail.com 08 Feb '11

08 Feb '11

From: Joseph Teichman <josteich(a)gmail.com> Signed-off-by: Joseph Teichman <josteich(a)gmail.com> diff --git a/usb/mixer.c b/usb/mixer.c index 7df89b3..85af605 100644 --- a/usb/mixer.c +++ b/usb/mixer.c @@ -95,7 +95,7 @@ enum { }; -/*E-mu 0202(0404) eXtension Unit(XU) control*/ +/*E-mu 0202/0404/0204 eXtension Unit(XU) control*/ enum { USB_XU_CLOCK_RATE = 0xe301, USB_XU_CLOCK_SOURCE = 0xe302, @@ -1566,7 +1566,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw cval->initialized = 1; } else { if (type == USB_XU_CLOCK_RATE) { - /* E-Mu USB 0404/0202/TrackerPre + /* E-Mu USB 0404/0202/TrackerPre/0204 * samplerate control quirk */ cval->min = 0; diff --git a/usb/quirks-table.h b/usb/quirks-table.h index 3599987..921a86f 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -79,6 +79,13 @@ .idProduct = 0x3f0a, .bInterfaceClass = USB_CLASS_AUDIO, }, +{ + /* E-Mu 0204 USB */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x041e, + .idProduct = 0x3f19, + .bInterfaceClass = USB_CLASS_AUDIO, +}, /* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface diff --git a/usb/quirks.c b/usb/quirks.c index cf8bf08..e314cdb 100644 --- a/usb/quirks.c +++ b/usb/quirks.c @@ -532,7 +532,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat } /* - * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, + * For E-Mu 0404USB/0202USB/TrackerPre/0204 sample rate should be set for device, * not for interface. */ @@ -589,6 +589,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; } -- 1.7.0.4

1 0

[PATCH 1/2] ALSA: HDA: Add a new Conexant codec 506e (20590)
by David Henningsson 08 Feb '11

08 Feb '11

Conexant 506e/20590 has the same graph as the rest of the 5066 family. BugLink: http://bugs.launchpad.net/bugs/723672 Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index dd7c5c1..909ce9e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3937,6 +3937,8 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5066 }, { .id = 0x14f15069, .name = "CX20585", .patch = patch_cxt5066 }, + { .id = 0x14f1506e, .name = "CX20590", + .patch = patch_cxt5066 }, { .id = 0x14f15097, .name = "CX20631", .patch = patch_conexant_auto }, { .id = 0x14f15098, .name = "CX20632", @@ -3963,6 +3965,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066"); MODULE_ALIAS("snd-hda-codec-id:14f15067"); MODULE_ALIAS("snd-hda-codec-id:14f15068"); MODULE_ALIAS("snd-hda-codec-id:14f15069"); +MODULE_ALIAS("snd-hda-codec-id:14f1506e"); MODULE_ALIAS("snd-hda-codec-id:14f15097"); MODULE_ALIAS("snd-hda-codec-id:14f15098"); MODULE_ALIAS("snd-hda-codec-id:14f150a1"); -- 1.7.1 --------------020500020206030309030703--

1 0

[alsa-devel] ANNOUNCE: NVIDIA HDMI audio documentation
by Stephen Warren 07 Feb '11

07 Feb '11

I have written a document re: how to understand and trouble-shoot issues with audio-over-HDMI on NVIDIA GPUs. It's available at: ftp://download.nvidia.com/XFree86/gpu-hdmi-audio-document/gpu-hdmi-audio.ht… The intent is to provide all the support information I've given various people on support forums in a centralized place. Hope this helps! -- nvpublic

2 4

[alsa-devel] ALSA: HDA ad1989 on Intel SCH interrupt issues
by Aleksandar Mihaylov 07 Feb '11

07 Feb '11

Hello, I have been working on a custom audio interface, utilizing the Intel SCH and an ad1989 codec. the goal is to provide 1 stereo headphone output and 3 stereo microphone inputs. As the Intel SCH only provides two DMA engines per direction my first question will be: How do i setup interleaving of the 3 stereo ADCs provided by the ad1989, through the two DMA engines. Additionally the more intriguing problem involves interrupt handling during capture(using APIC). Upon boot (or module reload) playback works just fine, however recording fails with "ALSA pcm_lib.c:1787: capture write error (DMA or IRQ trouble?)". This behaviour is consistent if one tries to continuously record form one stereo channel and is applicable to all 3 PCM-substreams as long as they are accessed one at a time. However if i try to record from 2 or more PCM substreams simultaneously, after a few failed attempts(same issue as above: DMA/IRQ trouble) one of the channels will engage, after which any subsequent recording request will start without any problems. The above issues arise when i force the HDA controller in APIC mode. If the controller is left in the default MSI interrupt mode then the kernel receives no interrupts on that IRQ, resulting in playback only playing the first buffer repetitively, until timing out with "DMA or IRQ trouble?". However recording exhibits a similar behaviour to above, where after a few attempts it successfully begins to record, inspection of /proc/interrupts still returns 0 for the assigned IRQ line. The same behaviour is exhibited when i use the default patch_analog.c module (snd-hda-codec-analog). Playback and recording through other media such as usb seem to work fine. Attached i am sending the alsa-info outputs immediately upon boot, and after the capture streams have begun recording properly, i have included detailed lspci output at the end of each file. Additionally i am sending the patch for alsa-driver 1.0.24 that contains the code for the new module as well as the PCI_QUIRKS i have added to hda_intel.c. My working directory is available through at: git://github.com/sashkosnail/AD1989_HDA.git I will much appreciate any input on the problem. -- Aleksandar Mihaylov, EIT

1 0