- 08 12月, 2010 3 次提交
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由 Anssi Hannula 提交于
When a new HDMI/DP device is plugged in, hdmi_update_short_audio_desc() is called for every SAD (Short Audio Descriptor) in the ELD data. For LPCM coding type SAD defines the supported sample sizes. For several other coding types (such as AC-3), a maximum bitrate is defined. The maximum bitrate and sample size fields are not always cleared. Therefore, if a device is unplugged and a different one is plugged in, and the coding types of some SAD positions differ between the devices, the old max_bitrate or sample_bits values will persist if the new SADs do not define those values. The leftover max_bitrate and sample_bits do not cause any issues other than wrongly showing up in eld#X.Y procfs file and kernel log. Fix that by always clearing sample_bits and max_bitrate when reading SADs. Signed-off-by: NAnssi Hannula <anssi.hannula@iki.fi> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Anssi Hannula 提交于
Commit bbbe3390 added functionality to restrict PCM parameters based on ELD info (derived from EDID data) of the audio sink. However, according to CEA-861-D no SAD is needed for basic audio (32/44.1/48kHz stereo 16-bit audio), which is instead indicated with a basic audio flag in the CEA EDID Extension. The flag is not present in ELD. However, as all audio capable sinks are required to support basic audio, we can assume it to be always available. Fix allowed audio formats with sinks that have SADs (Short Audio Descriptors) which do not completely overlap with the basic audio formats (there are no reports of affected devices so far) by always assuming that basic audio is supported. Reported-by: NStephen Warren <swarren@nvidia.com> Signed-off-by: NAnssi Hannula <anssi.hannula@iki.fi> Cc: stable@kernel.org Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Anssi Hannula 提交于
Commit bbbe3390 added functionality to restrict PCM parameters based on ELD info (derived from EDID data) of the audio sink. However, it wrongly assumes that the bits 0-2 of the first byte of CEA Short Audio Descriptors mean a supported number of channels. In reality, they mean the maximum number of channels (as per CEA-861-D 7.5.2). This means that the channel count can only be used to restrict max_channels, not min_channels. Restricting min_channels causes us to deny opening the device in stereo mode if the sink only has SADs that declare larger numbers of channels (like Primare SP32 AV Processor does). Fix that by not restricting min_channels based on ELD information. Signed-off-by: NAnssi Hannula <anssi.hannula@iki.fi> Reported-by: NJean-Yves Avenard <jyavenard@gmail.com> Tested-by: NJean-Yves Avenard <jyavenard@gmail.com> Cc: stable@kernel.org Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 06 12月, 2010 1 次提交
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由 Daniel T Chen 提交于
BugLink: https://launchpad.net/bugs/685161 The reporter of the bug states that he must use position_fix=1 to enable capture for the internal microphone, so set it for his machine's PCI SSID. Verified using 2.6.35 and the 2010-12-04 alsa-driver build. Reported-and-tested-by: NRalph Wabel <rwabel@gmx.net> Cc: <stable@kernel.org> Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 05 12月, 2010 1 次提交
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由 Anssi Hannula 提交于
Switch to the generic hdmi parser for codec id 1002:aa01 (ATI R6xx HDMI), as the codec appears to work fine with it. Note that the codec is still limited to stereo output only, despite it reportedly being multichannel capable. Some as of yet unknown quirks will be needed to get that working. Testing was done on 2.6.36 by John Ettedgui. Signed-off-by: NAnssi Hannula <anssi.hannula@iki.fi> Tested-by: NJohn Ettedgui <john.ettedgui@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 04 12月, 2010 1 次提交
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由 Manoj Iyer 提交于
Added a quirk to cxt5066_cfg_tbl to enable jack sense for ThinkPad Edge 13. Reference: http://launchpad.net/bugs/685015Signed-off-by: NManoj Iyer <manoj.iyer@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 03 12月, 2010 5 次提交
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由 Takashi Iwai 提交于
It sounds like a non-linear beep tone on my test machines... Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
These codecs have the digital beep widget in NID 0x21. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Instead of hard-coded magic numbers, properly define and use macros for improve the readability. Also, dell_automute is handled samely as thinkpad, since it also sets port_d_mode, too. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 John Baboval 提交于
On the docking station for the Lenovo T410 and T410s, the line-out doesn't work. The trouble seems to be that it generates a plug event, but then doesn't report that the jack is connected. So automute mutes the jack when you plug something into it. The following patch (next message) fixes it. Signed-off-by: John Baboval <john.baboval at virtualcomputer.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel T Chen 提交于
BugLink: https://launchpad.net/bugs/595482 The original reporter states that audible playback from the internal speaker is inaudible despite the hardware being properly detected. To work around this symptom, he uses the model=lg quirk to properly enable both playback, capture, and jack sense. Another user corroborates this workaround on separate hardware. Add this PCI SSID to the quirk table to enable it for further LG P1 Expresses. Reported-and-tested-by: NPhilip Peitsch <philip.peitsch@gmail.com> Tested-by: nikhov Cc: <stable@kernel.org> [2.6.32+] Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 02 12月, 2010 1 次提交
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由 Daniel T Chen 提交于
BugLink: https://launchpad.net/bugs/683695 The original reporter states that headphone jacks do not appear to work. Upon inspecting his codec dump, and upon further testing, it is confirmed that the "alienware" model quirk is correct. Reported-and-tested-by: Cody Thierauf Cc: <stable@kernel.org> [2.6.32+] Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 30 11月, 2010 1 次提交
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由 Takashi Iwai 提交于
In OSS emulation, SNDCTL_DSP_RESET ioctl needs the reset of the internal buffer state in addition to drop of the running streams. Otherwise the succeeding access becomes inconsistent. Tested-by: NAmit Nagal <helloin.amit@gmail.com> Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 29 11月, 2010 2 次提交
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由 Takashi Iwai 提交于
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel T Chen 提交于
BugLink: https://launchpad.net/bugs/682199 A 2.6.35 (Ubuntu Maverick) user, burningphantom1, reported a regression in audio: playback was inaudible through both speakers and headphones. In commit 272a527c of sound-2.6.git, a new model was added with this machine's PCI SSID. Fortunately, it is now sufficient to use the auto model for BIOS auto-parsing instead of the existing quirk. Playback, capture, and jack sense were verified working for both 2.6.35 and the alsa-driver snapshot from 2010-11-27 when model=auto is used. Reported-and-tested-by: burningphantom1 Cc: <stable@kernel.org> [2.6.35+] Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 27 11月, 2010 1 次提交
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由 Takashi Iwai 提交于
When SKU assid gives no valid bits for 0x38, the driver didn't take any action, so far. This resulted in the missing initialization for external amps, etc, thus the silent output in the end. Especially users hit this problem on ALC888 newly since 2.6.35, where the driver doesn't force to use ALC_INIT_DEFAULT any more. This patch sets the default initialization scheme to use ALC_INIT_DEFAULT when no valid bits are set for SKU assid. Reference: https://bugzilla.redhat.com/show_bug.cgi?id=657388Reported-and-tested-by: NKyle McMartin <kyle@redhat.com> Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 25 11月, 2010 1 次提交
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The mixer nids passed to alc_auto_create_input_ctls are wrong: 0x15 is a pin, and 0x09 is the ADC on both ALC660-VD/ALC861-VD. Thus with current code, input playback volume/switches and input source mixer controls are not created, and recording doesn't work. Select correct mixers, 0x0b (input playback mixer) and 0x22 (capture source mixer). Reference: https://qa.mandriva.com/show_bug.cgi?id=61159Signed-off-by: NHerton Ronaldo Krzesinski <herton@mandriva.com.br> Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 24 11月, 2010 2 次提交
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由 David Henningsson 提交于
The patch enables ALC887-VD to use the DAC at nid 0x26, which makes it possible to use this DAC for e g Headphone volume. Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Denis Kuplyakov 提交于
Fixes automatic EAPD configuration on Acer 7730G laptop. Signed-off-by: NDenis Kuplyakov <dener.kup@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 23 11月, 2010 4 次提交
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由 Kailang Yang 提交于
Give more correct chip names for ALC269-variant codecs. Signed-off-by: NKailang Yang <kailang@realtek.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Kailang Yang 提交于
The refactoring commit d433a678 ALSA: hda - Optimize the check of ALC269 codec variants introduced a wrong check for ALC269-vb type. This patch corrects it. Signed-off-by: NKailang Yang <kailang@realtek.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Manoj Iyer 提交于
Add a quirk entry for Thinkpad Edge 11 as well as other TP Edge models. Signed-off-by: NManoj Iyer <manoj.iyer@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
This reverts commit f41cc2a8. The patch broke the digital mic pin handling wrongly. Reference: bko#23162 https://bugzilla.kernel.org/show_bug.cgi?id=23162Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 22 11月, 2010 11 次提交
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由 Takashi Iwai 提交于
Add a generic callback function for fixup elements. This can be used to do some unusual things like overriding the AMP cache, etc. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Kailang Yang 提交于
ALC887-VD is like ALC888-VD. It can not be initialized as ALC882. Signed-off-by: NKailang Yang <kailang@realtek.com> Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
In the commit c0763e68 ALSA: snd-atmel-abdac: test wrong variable the return value via PTR_ERR() had to be fixed as well. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel T Chen 提交于
BugLink: https://launchpad.net/bugs/677652 The original reporter states that, in 2.6.35, headphones do not appear to work, nor does inserting them mute the A52J's onboard speakers. Upon inspecting the codec dump, it appears that the newly committed hp-laptop quirk will suffice to enable this basic functionality. Testing was done with an alsa-driver build from 2010-11-21. Reported-and-tested-by: Joan Creus Cc: <stable@kernel.org> [2.6.35+] Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Vasiliy Kulikov 提交于
After clk_get() pclk is checked second time instead of sample_clk check. Signed-off-by: NVasiliy Kulikov <segoon@openwall.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Andreas Mohr 提交于
. 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). Signed-off-by: NAndreas Mohr <andi@lisas.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel T Chen 提交于
BugLink: https://launchpad.net/bugs/677830 The original reporter states that the subwoofer does not mute when inserting headphones. We need an entry for his machine's SSID in the subwoofer pin fixup list, so add it there (verified using hda_analyzer). Reported-and-tested-by: i-NoD Cc: <stable@kernel.org> Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Joe Perches 提交于
Signed-off-by: NJoe Perches <joe@perches.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Joe Perches 提交于
Signed-off-by: NJoe Perches <joe@perches.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Joe Perches 提交于
Using %pR standardizes the struct resource output. Signed-off-by: NJoe Perches <joe@perches.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Daniel T Chen 提交于
BugLink: https://launchpad.net/bugs/669279 The original reporter states: "The Master mixer does not change the volume from the headphone output (which is affected by the headphone mixer). Instead it only seems to control the on-board speaker volume. This confuses PulseAudio greatly as the Master channel is merged into the volume mix." Fix this symptom by applying the hp_only quirk for the reporter's SSID. The fix is applicable to all stable kernels. Reported-and-tested-by: NBen Gamari <bgamari@gmail.com> Cc: <stable@kernel.org> [2.6.32+] Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 19 11月, 2010 1 次提交
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由 Axel Lin 提交于
After checking the code in 2.6.36, I found this is missing during multi-component conversion. Signed-off-by: NAxel Lin <axel.lin@gmail.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 18 11月, 2010 4 次提交
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由 Jesse Marroquin 提交于
This patch adds initial support for the MAX98089 CODEC. Signed-off-by: NJesse Marroquin <jesse.marroquin@maxim-ic.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Chris Paulson-Ellis 提交于
Multi-component commit f0fba2ad broke a few things which this patch should fix. Tested on the DM355 EVM. I've been as careful as I can, but it would be good if those with access to other Davinci boards could test. -- The multi-component commit put the initialisation of snd_soc_dai.[capture|playback]_dma_data into snd_soc_dai_ops.hw_params of the McBSP, McASP & VCIF drivers (davinci-i2s.c, davinci-mcasp.c & davinci-vcif.c). The initialisation had to be moved from the probe function in these drivers because davinci_*_dai changed from snd_soc_dai to snd_soc_dai_driver. Unfortunately, the DMA params pointer is needed by davinci_pcm_open (in davinci-pcm.c) before hw_params is called. I have moved the initialisation to a new snd_soc_dai_ops.startup function in each of these drivers. This fix indicates that all platforms that use davinci-pcm must have been broken and need to test with this fix. -- The multi-component commit also changed the McBSP driver name from "davinci-asp" to "davinci-i2s" in davinci-i2s.c without updating the board level references to the driver name. This change is understandable, as there is a similarly named "davinci-mcasp" driver in davinci-mcasp.c. There is probably no 'correct' name for this driver. The DM6446 datasheet calls it the "ASP" and describes it as a "specialised McBSP". The DM355 datasheet calls it the "ASP" and describes it as a "specialised ASP". The DM365 datasheet calls it the "McBSP". Rather than fix this problem by reverting to "davinci-asp", I've elected to avoid future confusion with the "davinci-mcasp" driver by changing it to "davinci-mcbsp", which is also consistent with the names of the functions in the driver. There are other fixes required, so it was never going to be as simple as a revert anyway. -- The DM365 only has one McBSP port (of the McBSP platforms, only the DM355 has 2 ports), so I've changed the the id of the platform_device from 0 to -1. -- In davinci-evm.c, the DM6446 EVM can no longer share a snd_soc_dai_link structure with the DM355 EVM as they use different cpu DAI names (the DM355 has 2 ports and the EVM uses the second port, but the DM6446 only has 1 port). This also means that the 2 boards need different snd_soc_card structures. -- The codec_name entries in davinci-evm.c didn't match the i2c ids in the board files. I have only checked and fixed the details of the names used for the McBSP based platforms. Someone with a McASP based platform (eg DA8xx) should check the others. Signed-off-by: NChris Paulson-Ellis <chris@edesix.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
During the multi-component conversion the WM8994 register cache init got lost. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
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由 Arnd Bergmann 提交于
The big kernel lock has been removed from all these files at some point, leaving only the #include. Remove this too as a cleanup. Signed-off-by: NArnd Bergmann <arnd@arndb.de> Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>
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- 15 11月, 2010 1 次提交
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由 Mark Brown 提交于
Ensure that we keep all widget powerups in DAPM sequence by making the CODEC the last thing we compare on rather than the first thing. Also fix the fact that we're currently comparing the widget pointers rather than the CODEC pointers when we do the substraction so we won't get stable results. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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