aboutsummaryrefslogtreecommitdiffstats
Commit message (Collapse)AuthorAge
* ALSA: hda - Always allow basic audio irrespective of ELD infoAnssi Hannula2010-12-07
| | | | | | | | | | | | | | | | | | | | | | | Commit bbbe33900d1f3c 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: Stephen Warren <swarren@nvidia.com> Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi> Cc: stable@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Do not wrongly restrict min_channels based on ELDAnssi Hannula2010-12-07
| | | | | | | | | | | | | | | | | | | | | | | Commit bbbe33900d1f3c 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: Anssi Hannula <anssi.hannula@iki.fi> Reported-by: Jean-Yves Avenard <jyavenard@gmail.com> Tested-by: Jean-Yves Avenard <jyavenard@gmail.com> Cc: stable@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda: Use position_fix=1 for Acer Aspire 5538 to enable capture on ↵Daniel T Chen2010-12-06
| | | | | | | | | | | | | | | internal mic 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: Ralph Wabel <rwabel@gmx.net> Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Enable jack sense for Thinkpad Edge 13Manoj Iyer2010-12-04
| | | | | | | | | Added a quirk to cxt5066_cfg_tbl to enable jack sense for ThinkPad Edge 13. Reference: http://launchpad.net/bugs/685015 Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix ThinkPad T410[s] docking station line-outJohn Baboval2010-12-03
| | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
* ALSA: hda: Use model=lg quirk for LG P1 Express to enable playback and captureDaniel T Chen2010-12-03
| | | | | | | | | | | | | | | | | 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: Philip Peitsch <philip.peitsch@gmail.com> Tested-by: nikhov Cc: <stable@kernel.org> [2.6.32+] Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda: Use "alienware" model quirk for another SSIDDaniel T Chen2010-12-02
| | | | | | | | | | | | | 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: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: Fix SNDCTL_DSP_RESET ioctl for OSS emulationTakashi Iwai2010-11-30
| | | | | | | | | | 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: Amit Nagal <helloin.amit@gmail.com> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda: Use BIOS auto-parsing instead of existing model quirk for MEDION MD2Daniel T Chen2010-11-29
| | | | | | | | | | | | | | | | | | | 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 272a527c04 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: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Use ALC_INIT_DEFAULT for really default initializationTakashi Iwai2010-11-26
| | | | | | | | | | | | | | | | | | | 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=657388 Reported-and-tested-by: Kyle McMartin <kyle@redhat.com> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix ALC660-VD/ALC861-VD capture/playback mixersHerton Ronaldo Krzesinski2010-11-25
| | | | | | | | | | | | | | 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=61159 Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: HDA: Add an extra DAC for Realtek ALC887-VDDavid Henningsson2010-11-24
| | | | | | | | | 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: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix Acer 7730G supportDenis Kuplyakov2010-11-24
| | | | | | | Fixes automatic EAPD configuration on Acer 7730G laptop. Signed-off-by: Denis Kuplyakov <dener.kup@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-linus' of ↵Linus Torvalds2010-11-23
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (41 commits) ALSA: hda - Identify more variants for ALC269 ALSA: hda - Fix wrong ALC269 variant check ALSA: hda - Enable jack sense for Thinkpad Edge 11 ALSA: Revert "ALSA: hda - Fix switching between dmic and mic using the same mux on IDT/STAC" ALSA: hda - Fixed ALC887-VD initial error ALSA: atmel - Fix the return value in error path ALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus A52J ALSA: snd-atmel-abdac: test wrong variable ALSA: azt3328: period bug fix (for PA), add missing ACK on stop timer ALSA: hda: Add Samsung R720 SSID for subwoofer pin fixup ALSA: sound/pci/asihpi/hpioctl.c: Remove unnecessary casts of pci_get_drvdata ALSA: sound/core/pcm_lib.c: Remove unnecessary semicolons ALSA: sound/ppc: Use printf extension %pR for struct resource ALSA: ac97: Apply quirk for Dell Latitude D610 binding Master and Headphone controls ASoC: uda134x - set reg_cache_default to uda134x_reg ASoC: Add support for MAX98089 CODEC ASoC: davinci: fixes for multi-component ASoC: Fix register cache setup WM8994 for multi-component ASoC: Fix dapm_seq_compare() for multi-component ASoC: RX1950: Fix hw_params function ...
| * Merge branch 'fix/asoc' into for-linusTakashi Iwai2010-11-23
| |\
| | * Merge branch 'for-2.6.37' of ↵Takashi Iwai2010-11-23
| | |\ | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into fix/asoc
| | | * ASoC: OMAP: fix OMAP1 compilation problemJanusz Krzysztofik2010-11-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the new code introduced with commit cf4c87abe238ec17cd0255b4e21abd949d7f811e, "OMAP: McBSP: implement McBSP CLKR and FSR signal muxing via mach-omap2/mcbsp.c", the way omap1 build is supposed to bypass omap2 specific functionality doesn't optimize out all omap2 specific stuff. This breaks linking phase for omap1 machines, giving "undefined reference to `omap2_mcbsp1_mux_clkr_src'" and "undefined reference to `omap2_mcbsp1_mux_fsr_src'" errors. Fix it. Created and tested against linux-2.6.37-rc1. Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Paul Walmsley <paul@pwsan.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | * Merge commit 'v2.6.37-rc1' into for-2.6.37Liam Girdwood2010-11-03
| | | |\
| | * | | ASoC: uda134x - set reg_cache_default to uda134x_regAxel Lin2010-11-19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | After checking the code in 2.6.36, I found this is missing during multi-component conversion. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: Add support for MAX98089 CODECJesse Marroquin2010-11-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds initial support for the MAX98089 CODEC. Signed-off-by: Jesse Marroquin <jesse.marroquin@maxim-ic.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: davinci: fixes for multi-componentChris Paulson-Ellis2010-11-17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Chris Paulson-Ellis <chris@edesix.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: Fix register cache setup WM8994 for multi-componentMark Brown2010-11-17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | During the multi-component conversion the WM8994 register cache init got lost. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | ASoC: Fix dapm_seq_compare() for multi-componentMark Brown2010-11-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: RX1950: Fix hw_params functionVasily Khoruzhick2010-11-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Unfortunatelly, I misunderstood datasheet, and on s3c244x-iis when MPLLin source for master clock is selected, prescaler has no effect. Remove dividor calculation for 44100 rate; remove 88200 rate at all, rx1950 can't do it. Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | Fix Atmel soc audio boards Kconfig dependencyRyan Mallon2010-11-11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add Kconfig dependency on AT91_PROGRAMMABLE_CLOCKS for the Atmel SoC audio SAM9G20-EK and PlayPaq boards. Fixes link errors on missing clk_set_parent and clk_set_rate when building without AT91_PROGRAMMABLE_CLOCKS. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> Acked-by: Geoffrey Wossum <gwossum@acm.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: Ensure sane WM835x AIF configuration by defaultMark Brown2010-11-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Ensure that whatever ran before us leaves the WM835x with a sane default audio interface configuration as we do not override the companding, loopback or tristate settings and do not reset the chip at startup (as it is a PMIC). Reported-by: Keiji Mitsuhisa <Keiji.Mitsuhisa@wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | ASoC: Remove broken WM8350 direction constantsMark Brown2010-11-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM8350 driver was using some custom constants to interpret the direction of the MCLK signal which had the opposite values to those used as standard by the ASoC core, causing confusion in machine drivers such as the 1133-EV1 board. Reported-by: Tommy Zhu <Tommy.Zhu@wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | ASoC: s3c24xx: Fix compilation problem for mini2440Marek Belisko2010-11-08
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When make mini2440_defconfig compilation end with undefined references to DMA functions. There was missing selection for S3C2410_DMA when compile ASoC audio for S3C24xx CPU. Tested on mini2440 board. Signed-off-by: Marek Belisko <marek.belisko@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: Return proper error if snd_soc_register_dais fails in psc_i2s_of_probeAxel Lin2010-11-08
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: WM8776: Removed unneeded struct memberDimitris Papastamos2010-11-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The member reg_cache is not used at all and therefore it should be removed. This member was usually needed for older versions of ASoC that did not handle caching automatically and had to be done in the driver itself. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: Lock the CODEC in PXA external jack controlsMark Brown2010-11-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When doing anything with the system, especially DAPM, we need to hold the CODEC mutex. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | phycore-ac97: add ac97 to cardnameSascha Hauer2010-11-05
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We have different codecs on the pcm038 (ac97 wm9712 and mc13783). To make alsactl restore work correctly these should have different names. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC i.MX: switch to new DMA apiSascha Hauer2010-11-05
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC i.MX: register dma audio deviceSascha Hauer2010-11-05
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We have two different transfer methods on i.MX: FIQ and DMA. Since the merge of the ASoC multicomponent support the DMA device is lost. Add it again. Also, imx_ssi_dai_probe has to be called for !AC97 aswell. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC i.MX phycore ac97: remove unnecessary includesSascha Hauer2010-11-05
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC i.MX eukrea tlv320: Fix for multicomponentSascha Hauer2010-11-05
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: Check return value of strict_strtoul() in WM8962Mark Brown2010-11-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | strict_strtoul() has been made __must_check so do so. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | Merge remote branch 'takashi/fix/asoc' into for-2.6.37Mark Brown2010-11-03
| | |\ \ \
| | * | | | ASoC: Fix snd_soc_register_dais error handlingAxel Lin2010-11-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | kzalloc for dai may fail at any iteration of the for loop, thus properly unregister already registered DAIs before return error. The error handling code in snd_soc_register_dais() already ensure all the DAIs are unregistered before return error, we can remove the error handling code to unregister DAIs in snd_soc_register_codec(). Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | | ALSA: hda - Identify more variants for ALC269Kailang Yang2010-11-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Give more correct chip names for ALC269-variant codecs. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: hda - Fix wrong ALC269 variant checkKailang Yang2010-11-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The refactoring commit d433a67831ab2c470cc53a3ff9b60f656767be15 ALSA: hda - Optimize the check of ALC269 codec variants introduced a wrong check for ALC269-vb type. This patch corrects it. Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: hda - Enable jack sense for Thinkpad Edge 11Manoj Iyer2010-11-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add a quirk entry for Thinkpad Edge 11 as well as other TP Edge models. Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: Revert "ALSA: hda - Fix switching between dmic and mic using the same ↵Takashi Iwai2010-11-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | mux on IDT/STAC" This reverts commit f41cc2a85d52ac6971299922084ac5ac59dc339d. The patch broke the digital mic pin handling wrongly. Reference: bko#23162 https://bugzilla.kernel.org/show_bug.cgi?id=23162 Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: hda - Fixed ALC887-VD initial errorKailang Yang2010-11-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ALC887-VD is like ALC888-VD. It can not be initialized as ALC882. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: atmel - Fix the return value in error pathTakashi Iwai2010-11-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the commit c0763e687d0283d0db507813ca4462aa4073c5b5 ALSA: snd-atmel-abdac: test wrong variable the return value via PTR_ERR() had to be fixed as well. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus A52JDaniel T Chen2010-11-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: snd-atmel-abdac: test wrong variableVasiliy Kulikov2010-11-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | After clk_get() pclk is checked second time instead of sample_clk check. Signed-off-by: Vasiliy Kulikov <segoon@openwall.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: azt3328: period bug fix (for PA), add missing ACK on stop timerAndreas Mohr2010-11-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | . 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: Andreas Mohr <andi@lisas.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: hda: Add Samsung R720 SSID for subwoofer pin fixupDaniel T Chen2010-11-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | ALSA: sound/pci/asihpi/hpioctl.c: Remove unnecessary casts of pci_get_drvdataJoe Perches2010-11-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Joe Perches <joe@perches.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>