- 19 Nov, 2012 1 commit
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Dylan Reid authored
Introduce the concept of a "gated" jack. The gated jack's pin sense is only valid when the "gating" jack is plugged. This requires checking the gating jack when the gated jack changes and re-checking the gated jack when the gating jack is plugged/unplugged. This allows handling of devices where the mic jack detect floats when the headphone jack is unplugged. [Rewritten for fixing the possible snd_array reallocation, covering the missing callback calls and jack sync operations, as well as some code cleanups -- tiwai] Signed-off-by:
Dylan Reid <dgreid@chromium.org> Signed-off-by:
Takashi Iwai <tiwai@suse.de>
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- 18 Nov, 2012 1 commit
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Dan Carpenter authored
We set "s" before we have capped "speed" so it could be the wrong value. This could lead to a divide by zero bug. Signed-off-by:
Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by:
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- 17 Nov, 2012 1 commit
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Joe Perches authored
Use bitmap_weight to count the total number of bits set in bitmap. Signed-off-by:
Joe Perches <joe@perches.com> Signed-off-by:
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- 14 Nov, 2012 2 commits
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Xi Wang authored
The dereference snd_pcm_plug_stream(plug) should come after the NULL check snd_BUG_ON(!plug). Signed-off-by:
Xi Wang <xi.wang@gmail.com> Signed-off-by:
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Xi Wang authored
The dereference snd_pcm_plug_stream(plug) should come after the NULL check snd_BUG_ON(!plug). Signed-off-by:
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- 13 Nov, 2012 1 commit
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Martin Schwenke authored
Probing this device currently fails in snd_usb_audio_probe() because the call to snd_usb_create_mixer() fails. This is due to unknown or non-standard interface descriptor subtypes in parse_audio_unit(): usbaudio: unit 51: unexpected type 0x09 snd-usb-audio: probe of 1-8:1.0 failed with error -5 Some people are working around this by recompiling usb-audio with the call to snd_usb_create_mixer() commented out. It would be nice to avoid that. While the best idea would be to look into the mixer creation failure, a reasonable short-term solution is to use quirks to only probe the trouble-free interfaces. This allows audio and MIDI interfaces to be used without any obvious issues. Interface 0 is the main one to ignore. It contains lots of control-fu, including the unexpected interface descriptor subtypes. Interface 5 is for firmware updates and I'm not sure how to get support for this. Interface 3 is some sort of control interface that I don't understand: Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 9 bInCollection 1 baInterfaceNr( 0) 1 Signed-off-by:Martin Schwenke <martin@meltin.net> Signed-off-by:
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- 12 Nov, 2012 4 commits
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Clemens Ladisch authored
Add a MIDI driver for the Stanton FireWire DJ controllers. Tested-by:
Sean M. Pappalardo - D.J. Pegasus <spappalardo@mixxx.org> Signed-off-by:
Clemens Ladisch <clemens@ladisch.de>
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Takashi Iwai authored
Bah, forgot this again... Signed-off-by: -
Ondrej Zary authored
Assume that unknown ICE1724-based cards are AC97-only that can suspend without any additional card-specific code. This fixes suspend on Gainward Hollywood@Home 7.1. Signed-off-by:
Ondrej Zary <linux@rainbow-software.org> Signed-off-by:
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- 09 Nov, 2012 1 commit
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David Henningsson authored
A closer look shows that the name is not even used and can be removed. Signed-off-by:
David Henningsson <david.henningsson@canonical.com> Signed-off-by:
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- 08 Nov, 2012 7 commits
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Takashi Iwai authored
When 2.1 speakers are detected, use the corresponding channel map instead of the standard map with front+rear surrounds. Signed-off-by: -
Takashi Iwai authored
When two built-in speakers are found on the machine, we can suppose it's rather a 2.1 speaker system with a bass output instead of front/surround channels. Signed-off-by: -
Takashi Iwai authored
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Takashi Iwai authored
There are uncovered cases whether the card refcount introduced by the commit a0830dbd isn't properly increased or decreased: - OSS PCM and mixer success paths - When lookup function gets NULL This patch fixes these places. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=50251 Cc: <stable@vger.kernel.org> Signed-off-by:
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Kailang Yang authored
Signed-off-by:
Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by:
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Kailang Yang authored
alc269_toggle_power_output() was only use in ALC269VB. I rename it to alc269vb_toggle_power_output(). Signed-off-by:
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Takashi Iwai authored
There are bug reports of a crash with USB-audio devices when PCM prepare is performed immediately after the stream is stopped via trigger callback. It turned out that the problem is that we don't wait until all URBs are killed. This patch adds a new function to synchronize the pending stop operation on an endpoint, and calls in the prepare callback for avoiding the crash above. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181Reported-and-tested-by:
Artem S. Tashkinov <t.artem@lycos.com> Cc: <stable@vger.kernel.org> [v3.6] Signed-off-by:
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- 07 Nov, 2012 5 commits
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Adrian Knoth authored
The RayDAT reports the sync status of its inputs in consecutive bit positions, so all we do in hdspm_s1_sync_check is to iterate over idx: status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); lock = (status & (0x1<<idx)) ? 1 : 0; sync = (status & (0x100<<idx)) ? 1 : 0; The index is given in kcontrol->private_value: HDSPM_SYNC_CHECK("WC SyncCheck", 0), HDSPM_SYNC_CHECK("AES SyncCheck", 1), HDSPM_SYNC_CHECK("SPDIF SyncCheck", 2), HDSPM_SYNC_CHECK("ADAT1 SyncCheck", 3), HDSPM_SYNC_CHECK("ADAT2 SyncCheck", 4), HDSPM_SYNC_CHECK("ADAT3 SyncCheck", 5), HDSPM_SYNC_CHECK("ADAT4 SyncCheck", 6), HDSPM_SYNC_CHECK("TCO SyncCheck", 7), HDSPM_SYNC_CHECK("SYNC IN SyncCheck", 8), The patch corrects the indicated sync flags by passing the proper index value to hdspm_s1_sync_check(). Signed-off-by:Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by:
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Takashi Iwai authored
To parse properly the subwoofer outputs on ASUS G75 laptop with VT1802 codec, correct the default configurations of speaker pins 0x24 and 0x33. Reported-by:
Massimo Del Fedele <max@veneto.com> Signed-off-by:
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Takashi Iwai authored
VT1802 codec provides the invalid connection lists of NID 0x24 and 0x33 containing the routes to a non-exist widget 0x3e. This confuses the auto-parser. Fix it up in the driver by overriding these connections. Reported-by:
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Takashi Iwai authored
In via_auto_fill_adc_nids(), the parser tries to fill dac_nids[] at the point of the current line-out (i). When no valid path is found for this output, this results in dac = 0, thus it creates a hole in dac_nids[]. This confuses is_empty_dac() and trims the detected DAC in later reference. This patch fixes the bug by appending DAC properly to dac_nids[] in via_auto_fill_adc_nids(). Reported-by:
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David Henningsson authored
On some of the PantherPoint HDMI machines we currently enable, we're seeing trouble with unsol events, i e detecting monitor presence, especially when on battery and after suspend/resume. BugLink: https://bugs.launchpad.net/bugs/1075882Tested-by:
Cyrus Lien <cyrus.lien@canonical.com> Signed-off-by:
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- 05 Nov, 2012 6 commits
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Sergiu Giurgiu authored
This patch adds support for ASUS - Xonar DSX sound cards. Tested on openSUSE 12.2 with kernel: Linux 3.4.6-2.10-desktop #1 SMP PREEMPT Thu Jul 26 09:36:26 UTC 2012 (641c197) x86_64 x86_64 x86_64 GNU/Linux Works: - play sounds - adjust volume on master channel. - mute . Since Xonar DS uses the same chip, everything that works for DS should work for DSX as well. Signed-off-by:
Sergiu Giurgiu <sgiurgiu11@gmail.com> Signed-off-by:
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Takashi Iwai authored
Several bug reports suggest that the forcibly resetting IEC958 status bits is required for AD codecs to get the SPDIF output working properly after changing streams. Original fix credit to Javeed Shaikh. BugLink: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/359361Reported-by:
Robin Kreis <r.kreis@uni-bremen.de> Cc: <stable@vger.kernel.org> Signed-off-by:
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Ondrej Zary authored
Add generic ESS vendor ID to pm_whitelist. This should fix suspend on all Maestro-2 and Maestro-2E based PCI cards. Tested on Terratec DMX and SF64-PCE2. Signed-off-by:
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Daniel J Blueman authored
Mark structures that won't change const. Signed-off-by:
Daniel J Blueman <daniel@quora.org> Signed-off-by:
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Daniel J Blueman authored
Correctly enable the digital microphones with the right bits in the right coeffecient registers on Cirrus CS4206/7 codecs. It also prevents misconfiguring ADC1/2. This fixes the digital mic on the Macbook Pro 10,1/Retina. Based-on-patch-by:
Alexander Stein <alexander.stein@systec-electronic.com> Signed-off-by:
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Alexander Stein authored
Signed-off-by:
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- 04 Nov, 2012 3 commits
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Lars R. Damerow authored
The audio chipset used in Teradici's Tera2 host cards is the same as that in the 1200 host cards. This patch allows ALSA to recognize the Tera2 cards. Signed-off-by:
Lars R. Damerow <lars@pixar.com> Signed-off-by:
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Masanari Iida authored
Correct spelling typo in debug messages within drivers/sound Signed-off-by:
Masanari Iida <standby24x7@gmail.com> Signed-off-by:
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Laurence Darby authored
Starting audio or seeking in various music players causes setup_dig_out_stream() to be called, which resets the SPDIF stream, which caused one DAC (but not another) to make a clicking noise every time. This patch ensures the reset only happens when it needs to, which is when the format changes, and makes the code a little more readable. Signed-off-by:
Laurence Darby <ldarby@tuffmail.com> Signed-off-by:
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- 31 Oct, 2012 1 commit
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Takashi Iwai authored
The rate isn't restored properly after resume since it's only set up in hw_params, and not in prepare callback. For fixing it, put the corresponding call to resume callback as well. Reported-and-tested-by:
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- 30 Oct, 2012 7 commits
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Takashi Iwai authored
... for migrating the core changes for USB-audio disconnection fixes
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Takashi Iwai authored
When disconnect callback is called, each component should wake up sleepers and check card->shutdown flag for avoiding the endless sleep blocking the proper resource release. Cc: <stable@vger.kernel.org> Signed-off-by: -
Takashi Iwai authored
For more strict protection for wild disconnections, a refcount is introduced to the card instance, and let it up/down when an object is referred via snd_lookup_*() in the open ops. The free-after-last-close check is also changed to check this refcount instead of the empty list, too. Reported-by:
Matthieu CASTET <matthieu.castet@parrot.com> Cc: <stable@vger.kernel.org> Signed-off-by:
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Takashi Iwai authored
Similar like the previous commit, cover with chip->shutdown_rwsem and chip->shutdown checks. Reported-by:
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Takashi Iwai authored
Replace mutex with rwsem for codec->shutdown protection so that concurrent accesses are allowed. Also add the protection to snd_usb_autosuspend() and snd_usb_autoresume(), too. Reported-by:
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Takashi Iwai authored
Close some races at disconnection of a USB audio device by adding the chip->shutdown_mutex and chip->shutdown check at appropriate places. The spots to put bandaids are: - PCM prepare, hw_params and hw_free - where the usb device is accessed for communication or get speed, in mixer.c and others; the device speed is now cached in subs->speed instead of accessing to chip->dev The accesses in PCM open and close don't need the mutex protection because these are already handled in the core PCM disconnection code. The autosuspend/autoresume codes are still uncovered by this patch because of possible mutex deadlocks. They'll be covered by the upcoming change to rwsem. Also the mixer codes are untouched, too. These will be fixed in another patch, too. Reported-by:
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Takashi Iwai authored
Fix races at PCM disconnection: - while a PCM device is being opened or closed - while the PCM state is being changed without lock in prepare, hw_params, hw_free ops Reported-by:
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