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* ALSA: usb-audio: Fix mutex deadlock at disconnectionTakashi Iwai2012-11-141-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | The recent change for USB-audio disconnection race fixes introduced a mutex deadlock again. There is a circular dependency between chip->shutdown_rwsem and pcm->open_mutex, depicted like below, when a device is opened during the disconnection operation: A. snd_usb_audio_disconnect() -> card.c::register_mutex -> chip->shutdown_rwsem (write) -> snd_card_disconnect() -> pcm.c::register_mutex -> pcm->open_mutex B. snd_pcm_open() -> pcm->open_mutex -> snd_usb_pcm_open() -> chip->shutdown_rwsem (read) Since the chip->shutdown_rwsem protection in the case A is required only for turning on the chip->shutdown flag and it doesn't have to be taken for the whole operation, we can reduce its window in snd_usb_audio_disconnect(). Reported-by: Jiri Slaby <jslaby@suse.cz> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: fm801: precedence bug in snd_fm801_tea575x_get_pins()Dan Carpenter2012-11-141-3/+8
| | | | | | | | There is a precedence bug because | has higher precedence than ?:. This code was cut and pasted and I fixed a similar bug a few days ago. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: es1968: precedence bug in snd_es1968_tea575x_get_pins()Dan Carpenter2012-11-131-3/+8
| | | | | | | | | | I don't think this works as intended. '|' higher precedence than ?: so the bitwize OR "0 | (val & STR_MOST)" is a no-op. I have re-written it to be more clear. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge tag 'asoc-3.7' of ↵Takashi Iwai2012-11-138-12/+573
|\ | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v3.7 A few small fixes plus a large but simple change for WM5102 which writes out a bunch of register updates to the device when we enable the clock as recommended following chip evaluation.
| *---------. Merge branches 'fix/arizona', 'fix/core', 'fix/cs42l52', 'fix/mxs', ↵Mark Brown2012-11-138-12/+573
| |\ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | 'fix/samsung' and 'fix/wm8978' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into tmp
| | | | | | | * ASoC: wm8978: pll incorrectly configured when codec is masterEric Millbrandt2012-11-061-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When MCLK is supplied externally and BCLK and LRC are configured as outputs (codec is master), the PLL values are only calculated correctly on the first transmission. On subsequent transmissions, at differenct sample rates, the wrong PLL values are used. Test for f_opclk instead of f_pllout to determine if the PLL values are needed. Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | | | | | * | ASoC: bells: Correct type in sub speaker DAI name for WM5102Charles Keepax2012-11-071-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | | | | * | ASoC: bells: Select WM1250-EV1 Springbank audio I/O moduleDimitris Papastamos2012-11-021-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Ensure we select the WM1250-EV1 as the current software system configuration demands it. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | | | | * | ASoC: bells: Add missing select of WM0010Dimitris Papastamos2012-11-021-0/+1
| | | | | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | | | * | ASoC: mxs-saif: Fix channel swap for 24-bit formatFabio Estevam2012-11-021-4/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Playing 24-bit format file leads to channel swap on mx28 and the reason is that the current driver performs one write/read to/from the SAIF_DATA register to trigger the transfer. This approach works fine for S16_LE case because SAIF_DATA is a 32-bit register and thus is capable of storing the 16-bit left and right channels, but for the S24_LE case it can only store one channel, so in order to not lose the FIFO sync an extra read/write is needed. Reported-by: Dan Winner <DWinner@tc-helicon.com> Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Tested-by: Dan Winner <DWinner@tc-helicon.com> Acked-by: Dong Aisheng <dong.aisheng@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | | | * | ASoC: mxs-saif: Add MODULE_ALIASFabio Estevam2012-11-011-0/+1
| | | | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add MODULE_ALIAS information. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Acked-by: Dong Aisheng <dong.aisheng@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | | * | ASoC: cs42l52: fix the return value of cs42l52_set_fmt()Wei Yongjun2012-11-071-2/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fix the return value of cs42l52_set_fmt() when clock inversion is not allowed and also remove the useless variable ret. dpatch engine is used to auto generate this patch. (https://github.com/weiyj/dpatch) [We had been assigning to ret but then ignoring the value we assgined -- broonie] Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | | * | | ASoC: core: Double control update err for snd_soc_put_volsw_sxMukund Navada2012-11-091-2/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_soc_put_volsw_sx function fails to update second control if first control is updated by snd_soc_update_bits_locked. Signed-off-by: Mukund Navada <navada@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | | * | | ASoC: dapm: Use card_list during DAPM shutdownMisael Lopez Cruz2012-11-091-1/+1
| | | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | DAPM shutdown incorrectly uses "list" field of codec struct while iterating over probed components (codec_dev_list). "list" field refers to codecs registered in the system, "card_list" field is used for probed components. Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | * | / ASoC: wm5102: Write register value corrections after SYSCLK is enabledMark Brown2012-10-301-1/+551
| | | |/ | | |/| | | | | | | | | | | | | | | | | | | | | | | | | Evalation of the WM5102 has identified a number of register values which should be written after SYSCLK is enabled on revision A in order to improve performance. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | / | ALSA: hda - Add a missing quirk entry for iMac 9,1Takashi Iwai2012-11-121-0/+1
|/ / / | | | | | | | | | | | | | | | | | | | | | This is another variant of iMac 9,1 with a different codec SSID. Reported-and-tested-by: Everaldo Canuto <everaldo.canuto@gmail.com> Cc: <stable@vger.kernel.org> [v3.3+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: Fix card refcount unbalanceTakashi Iwai2012-11-085-4/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Add new codec ALC668 and ALC900 (default name ALC1150)Kailang Yang2012-11-081-0/+2
| | | | | | | | | | | | | | | | | | Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Improve HP depop when system enter to S3Kailang Yang2012-11-081-13/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | alc269_toggle_power_output() was only use in ALC269VB. I rename it to alc269vb_toggle_power_output(). Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: usb-audio: Fix crash at re-preparing the PCM streamTakashi Iwai2012-11-083-0/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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=49181 Reported-and-tested-by: Artem S. Tashkinov <t.artem@lycos.com> Cc: <stable@vger.kernel.org> [v3.6] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hdspm - Fix sync check reporting on RME RayDATAdrian Knoth2012-11-071-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Add pin fixups for ASUS G75Takashi Iwai2012-11-071-0/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Fix invalid connections in VT1802 codecTakashi Iwai2012-11-071-0/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Massimo Del Fedele <max@veneto.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Fix empty DAC filling in patch_via.cTakashi Iwai2012-11-071-7/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Massimo Del Fedele <max@veneto.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Force to reset IEC958 status bits for AD codecsTakashi Iwai2012-11-051-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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/359361 Reported-by: Robin Kreis <r.kreis@uni-bremen.de> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: es1968: Add ESS vendor ID to pm_whitelistOndrej Zary2012-11-051-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Ondrej Zary <linux@rainbow-software.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: HDA: Mark CS260x immutable structures constDaniel J Blueman2012-11-051-3/+2
| | | | | | | | | | | | | | | | | | | | | Mark structures that won't change const. Signed-off-by: Daniel J Blueman <daniel@quora.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: HDA: Fix digital microphone on CS420xDaniel J Blueman2012-11-051-5/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Daniel J Blueman <daniel@quora.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda: Cirrus: Fix coefficient index for beep configurationAlexander Stein2012-11-051-1/+1
| | | | | | | | | | | | | | | | | | Signed-off-by: Alexander Stein <alexander.stein@systec-electronic.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - support Teradici 2200 host card audioLars R. Damerow2012-11-041-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
* | | ALSA: Fix typo in drivers soundMasanari Iida2012-11-046-6/+6
| |/ |/| | | | | | | | | | | Correct spelling typo in debug messages within drivers/sound Signed-off-by: Masanari Iida <standby24x7@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: ice1724: Fix rate setup after resumeTakashi Iwai2012-10-311-1/+6
| | | | | | | | | | | | | | | | | | 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: Ondrej Zary <linux@rainbow-software.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: Avoid endless sleep after disconnectTakashi Iwai2012-10-306-1/+46
| | | | | | | | | | | | | | | | | | 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 <tiwai@suse.de>
* | ALSA: Add a reference counter to card instanceTakashi Iwai2012-10-3010-32/+83
| | | | | | | | | | | | | | | | | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Fix races at disconnection in mixer_quirks.cTakashi Iwai2012-10-301-7/+51
| | | | | | | | | | | | | | | | | | Similar like the previous commit, cover with chip->shutdown_rwsem and chip->shutdown checks. Reported-by: Matthieu CASTET <matthieu.castet@parrot.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Use rwsem for disconnect protectionTakashi Iwai2012-10-304-17/+21
| | | | | | | | | | | | | | | | | | | | | | | | 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: Matthieu CASTET <matthieu.castet@parrot.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: usb-audio: Fix races at disconnectionTakashi Iwai2012-10-305-41/+79
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Matthieu CASTET <matthieu.castet@parrot.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: PCM: Fix some races at disconnectionTakashi Iwai2012-10-302-5/+18
| | | | | | | | | | | | | | | | | | | | | | 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: Matthieu CASTET <matthieu.castet@parrot.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge tag 'asoc-3.7' of ↵Takashi Iwai2012-10-282-5/+4
|\ \ | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v3.7 Clean up some fallout from the OMAP header reorganisation and a minor fix for DMIC which has no practical effect but is neater.
| * | ASoC: omap-dmic: Correct functional clock namePeter Ujfalusi2012-10-271-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | We should really use "fck" when asking for the functional clock and not "dmic_fck". This way we can ensure that multiple dmic modules can exist in the system. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: zoom2: Fix compile error by including correct header filesTony Lindgren2012-10-271-3/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Also drop the includes that are no longer needed and just cause problems for the ARM common zImage. Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Tim Gardner <tim.gardner@canonical.com> [tony@atomide.com: updated to drop unneeded headers] Signed-off-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ALSA: hda - Fix mute-LED setup for HP dv5 laptopGustavo Maciel Dias Vieira2012-10-261-0/+2
| |/ |/| | | | | | | | | | | | | | | | | The BIOS on HP dv5 doesn't have the DMI string to guide the setup of mute led GPIO and polarity. Associate this laptop with the hp-inv-led model. Signed-off-by: Gustavo Maciel Dias Vieira <gustavo@sagui.org> Tested-by: Vinícius Angiolucci <angiolucci@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge tag 'asoc-3.7' of ↵Takashi Iwai2012-10-254-6/+38
|\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v3.7 A couple of driver fixes, one that improves the interoperability of WM8994 with controllers that are sensitive to extra BCLK cycles and some build break fixes for ux500.
| * \ Merge remote-tracking branches 'asoc/fix/ux500' and 'asoc/fix/wm8994' into ↵Mark Brown2012-10-2517-80/+136
| |\ \ | | | | | | | | | | | | for-3.7
| | * | ASoC: wm8994: Only enable extra BCLK cycles when requiredMark Brown2012-10-242-1/+18
| | |/ | | | | | | | | | | | | | | | | | | | | | | | | Rather than always assuming the maximum possible BCLK rate will be required generate BCLKs for stereo if either one or two channels is enabled. In order to support this we also need to ensure that only the relevant channels are enabled. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: ux500_msp_i2s: Fix devm_* and return code merge errorLee Jones2012-10-161-5/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some ux500_msp_i2s patches clashed with: b18e93a493626c1446f9788ebd5844d008bbf71c ASoC: ux500_msp_i2s: better use devm functions and fix error return code ... leaving the driver uncompilable. This patch fixes the issues encountered. Acked-by: Linus Walleij <linus.walleij@linaro.org> Signed-off-by: Lee Jones <lee.jones@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: Ux500: Dispose of device nodes correctlyLee Jones2012-10-161-0/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When of_parse_phandle() is used to find a device node, its reference count is incremented by the helper. Once we're finished with them, it's our responsibly to ensure they are freed in the correct manor. Acked-by: Linus Walleij <linus.walleij@linaro.org> Signed-off-by: Lee Jones <lee.jones@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ALSA: als3000: check for the kzalloc return valueDenis Kirjanov2012-10-221-0/+4
| | | | | | | | | | | | | | | Signed-off-by: Denis Kirjanov <kirjanov@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: sound/isa/opti9xx/miro.c: eliminate possible double freeJulia Lawall2012-10-211-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_miro_probe is a static function that is only called twice in the file that defines it. At each call site, its argument is freed using snd_card_free. Thus, there is no need for snd_miro_probe to call snd_card_free on its argument on any of its error exit paths. Because snd_card_free both reads the fields of its argument and kfrees its argments, the results of the second snd_card_free should be unpredictable. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r@ identifier f,free,a; parameter list[n] ps; type T; expression e; @@ f(ps,T a,...) { ... when any when != a = e if(...) { ... free(a); ... return ...; } ... when any } @@ identifier r.f,r.free; expression x,a; expression list[r.n] xs; @@ * x = f(xs,a,...); if (...) { ... free(a); ... return ...; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Fix silent headphone output from Toshiba P200Takashi Iwai2012-10-201-1/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | By some reason, Toshiba laptop doesn't like the EAPD turned up for the headphone pin. Add a fix up code to force to turn down EAPD for NID 0x15. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=569991 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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