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* ALSA: hda - Fix silent headphone output from Toshiba P200Takashi Iwai2012-10-311-1/+18
| | | | | | | | | | | | | | commit cb766404e6b8c566569eb9ada02ea45d28729864 upstream. 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 Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - add dock support for Thinkpad T430Stefán Freyr2012-10-311-0/+1
| | | | | | | | | | | | | | | | | | | | commit 84f98fdf7865fbd35b312eb39ea91e5618c514c7 upstream. I have a Lenovo ThinkPad T430 and an UltraBase Series 3 docking station. Without this patch, if I plug my headphones into the jack on the computer, everything works fine. The computer speakers mute and the audio is played in the headphones. However, if I plug into the docking station headphone jack the computer speakers are muted but there is no audio in the headphones. Addresses https://bugs.launchpad.net/bugs/1060372 Signed-off-by: Joseph Salisbury <joseph.salisbury@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: emu10k1: add chip details for E-mu 1010 PCIe cardMaxim Kachur2012-10-211-0/+9
| | | | | | | | | | | | commit 10f571d09106c3eb85951896522c9650596eff2e upstream. Add chip details for E-mu 1010 PCIe card. It has the same chip as found in E-mu 1010b but it uses different PCI id. Signed-off-by: Maxim Kachur <mcdebugger@duganet.ru> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: ac97 - Fix missing NULL check in snd_ac97_cvol_new()Takashi Iwai2012-10-211-0/+2
| | | | | | | | | | commit 733a48e5ae5bf28b046fad984d458c747cbb8c21 upstream. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=44721 Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: omap-abe-twl6040: Fix typo of VibratorPeter Ujfalusi2012-10-211-1/+1
| | | | | | | | | | | commit 034940a6b3afbe79022ab6922dd9d2982b78e6d5 upstream. It is not Vinrator but Vibrator. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: wm2200: Fix non-inverted OUT2 mute controlMark Brown2012-10-211-1/+1
| | | | | | | | commit a1b98e12b7f8fad2f0aa3c08a3302bcac7ae1ec7 upstream. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: wm2200: Use rev A register patches on rev BMark Brown2012-10-211-0/+1
| | | | | | | | commit 5ae9eb4cbdfd640269dbd66aa3c92ea8e11cc838 upstream. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: fsi: don't reschedule DMA from an atomic contextGuennadi Liakhovetski2012-10-211-7/+8
| | | | | | | | | | | | | | | commit 57451e437796548d658d03c2c4aab659eafcd799 upstream. shdma doesn't support transfer re-scheduling or triggering from callbacks or from atomic context. The fsi driver issues DMA transfers from a tasklet context, which is a bug. To fix it convert tasklet to a work. Reported-by: Do Q.Thang <dq-thang@jinso.co.jp> Tested-by: Do Q.Thang <dq-thang@jinso.co.jp> Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Always check array bounds in alc_get_line_out_pfxDavid Henningsson2012-10-211-1/+3
| | | | | | | | | | | | | commit 71aa5ebe36a4e936eff281b375a4707b6a8320f2 upstream. Even when CONFIG_SND_DEBUG is not enabled, we don't want to return an arbitrary memory location when the channel count is larger than we expected. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Stop LPIB delay counting on broken hardwareTakashi Iwai2012-10-211-3/+6
| | | | | | | | | | | | | | commit 1f04661fde9deda4a2cd5845258715a22d8af197 upstream. If LPIB reports a pretty bad value, we can't trust such hardware for calculating the PCM delay. Automatically turn off the delay counting when such a problem is encountered. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=48911 Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Fix registration race of VGA switcherooTakashi Iwai2012-10-211-11/+20
| | | | | | | | | | | | | | | commit 128960a9ad67e2d119738f5211956e0304517551 upstream. Delay the registration of VGA switcheroo client to the end of the probing. Otherwise a too quick switching may result in Oops during probing. Also add the check of the return value from snd_hda_lock_devices(). Reported-and-tested-by: Daniel J Blueman <daniel@quora.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Fix memory leaks at error path in patch_cirrus.cTakashi Iwai2012-10-211-3/+3
| | | | | | | | | | commit c5e0b6dbad9b4d18c561af90b384d02373f1c994 upstream. The proper destructor should be called at the error path. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - do not detect jack on internal speakers for RealtekDavid Henningsson2012-10-211-0/+2
| | | | | | | | | | | | | | commit f7f4b2322bf7b8c5929b7eb5a667091f32592580 upstream. This caused the internal speaker to mute itself because it was present, which happened after powersave. It was found on Dell XPS 15 (L502x), ALC665. Reported-by: Da Fox <da.fox.mail@gmail.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Add missing hda_gen_spec to struct via_specTakashi Iwai2012-10-211-0/+4
| | | | | | | | | | | | | | | | commit 7819d1c70eb6a57e43554d86e10b39d1e106ed65 upstream. The commit [4b527b65 ALSA: hda - limit internal mic boost for Asus X202E] introduced the use of auto-parser code, but it forgot to add struct hda_gen_spec at the head of codec->spec which the auto-parser assumes silently. Without this record, it may result in memory corruption. This patch adds the missing piece. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Fix hang caused by race during suspend.Dylan Reid2012-10-131-4/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit d17344b3547669f5b6ee4fda993d03737a141bd6 upstream. There was a race condition when the system suspends while hda_power_work is running in the work queue. If system suspend (snd_hda_suspend) happens after the work queue releases power_lock but before it calls hda_call_codec_suspend, codec_suspend runs with power_on=0, causing the codec to power up for register reads, and hanging when it calls cancel_delayed_work_sync from the running work queue. The call chain from the work queue will look like this: hda_power_work <<- power_on = 1, unlock, then power_on cleard by suspend hda_call_codec_suspend hda_set_power_state snd_hda_codec_read codec_exec_verb snd_hda_power_up snd_hda_power_save __snd_hda_power_up cancel_delayed_work_sync <<-- cancelling executing wq Fix this by waiting for the work queue to finish before starting suspend if suspend is not happening on the work queue. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - use LPIB for delay estimationPierre-Louis Bossart2012-10-131-7/+29
| | | | | | | | | | | | | | | | | | | | | | | commit 90accc58a6946e7245993da6079f88d8c29cb731 upstream. DMA Position in Buffer (DPIB) should be used for ring buffer management, while LPIB register provides information on the number of samples transfered on the link. The difference between the two pieces of information corresponds to hardware/DMA buffering. This patch reports this difference in runtime->delay, and removes the use of the COMBO mode on recent Intel hardware. Credits to Takashi Iwai for an initial patch. [rebased to for-next branch and replaced snd_printk() with snd_printdd() by tiwai] Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Add another pci id for Haswell boardWang Xingchao2012-10-131-0/+3
| | | | | | | | | | | | | | | commit d279fae8a41690ec1b20c07be8c6f42f8af27a17 upstream. A new PCI id 0x0d0c for Haswell HDA Controller. [root@SKBM04SDP ~]# lspci |grep Audio 00:03.0 Audio device: Intel Corporation Device 0d0c (rev 02) 00:1b.0 Audio device: Intel Corporation Lynx Point HD Audio Controller Signed-off-by: Wang Xingchao <xingchao.wang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: USB: Support for (original) Xbox CommunicatorMarko Friedemann2012-10-131-0/+53
| | | | | | | | | | | commit c05fce586d4da2dfe0309bef3795a8586e967bc3 upstream. Added support for Xbox Communicator to USB quirks. Signed-off-by: Marko Friedemann <mfr@bmx-chemnitz.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: snd-usb: Add quirks for Playback Designs devicesDaniel Mack2012-10-135-0/+49
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 2b58fd5b3193fd3af3d15114d95706087d25a7fe upstream. Playback Designs' USB devices have some hardware limitations on their USB interface. In particular: - They need a 20ms delay after each class compliant request as the hardware ACKs the USB packets before the device is actually ready for the next command. Sending data immediately will result in buffer overflows in the hardware. - The devices send bogus feedback data at the start of each stream which confuse the feedback format auto-detection. This patch introduces a new quirks hook that is called after each control packet and which adds a delay for all devices that match Playback Designs' USB VID for now. In addition, it adds a counter to snd_usb_endpoint to drop received packets on the floor. Another new quirks function that is called once an endpoint is started initializes that counter for these devices on their sync endpoint. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Andreas Koch <andreas@akdesigninc.com> Supported-by: Demian Martin <demianm_1@yahoo.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: usb - disable broken hw volume for Tenx TP6911David Henningsson2012-10-131-0/+7
| | | | | | | | | | | | | | | | | | commit c10514394ef9e8de93a4ad8c8904d71dcd82c122 upstream. While going through Ubuntu bugs, I discovered this patch being posted and a confirmation that the patch works as expected. Finding out how the hw volume really works would be preferrable to just disabling the broken one, but this would be better than nothing. Credit: sndfnsdfin (qawsnews) BugLink: https://bugs.launchpad.net/bugs/559939 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - limit internal mic boost for Asus X202EDavid Henningsson2012-10-131-0/+29
| | | | | | | | | | | | | | commit 4b527b6516ab1f0af8aaedd02dbf71ee2c1180f4 upstream. When the input gain for the internal mic is set to its maximum level, the background noise becomes so high - and any relevant signal clipped - that the setting becomes unusable. It is better to limit the amplification. BugLink: https://bugs.launchpad.net/bugs/1052460 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/realtek - Fix detection of ALC271X codecHerton Ronaldo Krzesinski2012-10-131-6/+6
| | | | | | | | | | | | | | | | | | | | | commit 9f720bb9409ea5923361fbd3fdbc505ca36cf012 upstream. In commit af741c1 ("ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup"), alc_auto_parse_customize_define was moved after detection of ALC271X. The problem is that detection of ALC271X relies on spec->cdefine.platform_type, and it's set on alc_auto_parse_customize_define. Move the alc_auto_parse_customize_define and its required fixup setup before the block doing the ALC271X and other codec setup. BugLink: https://bugs.launchpad.net/bugs/1006690 Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com> Reviewed-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda/via - don't report presence on HPs with no presence supportHerton Ronaldo Krzesinski2012-10-131-1/+2
| | | | | | | | | | | | | | | | | | | | | | commit cf55e904516947597d75fd3844acc24891a95772 upstream. If headphone jack can't detect plug presence, and we have the jack in the jack table, snd_hda_jack_detect will return the plug as always present (as it'll be considered as a phantom jack). The problem is that when this happens, line out pins will always be disabled, resulting in no sound if there are no headphones connected. This was reported as a no sound problem after suspend on http://bugs.launchpad.net/bugs/1052499, since the bug doesn't manifests on first initialization before the phantom jack is added, but on resume we reexecute the initialization code, and via_hp_automute starts reporting HP always present with the jack now on the table. BugLink: https://bugs.launchpad.net/bugs/1052499 Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: aloop - add locking to timer accessOmair Mohammed Abdullah2012-10-131-0/+6
| | | | | | | | | | | | | | | | | | | commit d4f1e48bd11e3df6a26811f7a1f06c4225d92f7d upstream. When the loopback timer handler is running, calling del_timer() (for STOP trigger) will not wait for the handler to complete before deactivating the timer. The timer gets rescheduled in the handler as usual. Then a subsequent START trigger will try to start the timer using add_timer() with a timer pending leading to a kernel panic. Serialize the calls to add_timer() and del_timer() using a spin lock to avoid this. Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah@linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310Felix Kaechele2012-10-131-0/+1
| | | | | | | | | | | | commit e4db0952e542090c605fd41d31d761f1b4624f4a upstream. The Lenovo IdeaPad U310 has an internal mic where the right channel is phase inverted. Signed-off-by: Felix Kaechele <felix@fetzig.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: wm9712: Fix name of Capture SwitchMark Brown2012-10-131-1/+1
| | | | | | | | | | commit 689185b78ba6fbe0042f662a468b5565909dff7a upstream. Help UIs associate it with the matching gain control. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: wm5110: Adding missing volume update bitsCharles Keepax2012-10-131-0/+4
| | | | | | | | | | | commit ae60503741991a36ed6b2a8f53b249b2a72af52b upstream. The volume update bits were being set on all but one input and one output. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ASoC: wm_hubs: Ensure volume updates are handled during class W startupMark Brown2012-10-131-0/+5
| | | | | | | | | | | | commit eb4d5fc1f0ce89e3d5b072c594a1e213a0e05881 upstream. In some circumstances we may need to flush volume updates to the device after switching to class W mode. Do this unconditionally to ensure that these situations are handled. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
* ALSA: snd-usb: fix next_packet_size calls for pause caseDaniel Mack2012-09-271-1/+7
| | | | | | | | | | | | Also fix the calls to next_packet_size() for the pause case. This was missed in 245baf983 ("ALSA: snd-usb: fix calls to next_packet_size"). Signed-off-by: Daniel Mack <zonque@gmail.com> Reviewed-by: Takashi Iwai <tiwai@suse.de> Reported-and-tested-by: Christian Tefzer <ctrefzer@gmx.de> Cc: stable@kernel.org [ Taking directly because Takashi is on vacation - Linus ] Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
* ASoC: wm2000: Correct register sizeMark Brown2012-09-261-1/+1
| | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
* Merge tag 'asoc-3.6' of ↵Takashi Iwai2012-09-1511-40/+21
|\ | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Updates for 3.6 A bigger set of updates than I'm entirely comfortable with - things backed up a bit due to travel. As ever the majority of these are small, focused updates for specific drivers though there are a couple of core changes. There's been good exposure in -next. The AT91 patch fixes a build break.
| * ASoC: wm8904: correct the indexBo Shen2012-09-141-1/+1
| | | | | | | | | | Signed-off-by: Bo Shen <voice.shen@atmel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: tegra: fix maxburst settings in dmaengine codeStephen Warren2012-09-071-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | The I2S controllers are programmed with an "attention" level of 4 DWORDs. This must match the configuration passed to the DMA driver, so that when they burst in data, they don't overflow the available FIFO space. Also, the burst size is relevant to the destination for playback, and source for capture, not vice-versa as originally written. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * ASoC: samsung dma - Don't indicate support for pause/resume.Dylan Reid2012-09-061-7/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The pause and resume operations indicate that the stream can be un-paused/resumed from the exact location they were paused/suspended. This is not true for this driver, the pause and suspend triggers share the same code path with stop, they flush all pending DMA transfers. This drops all pending samples. The pause_release/resume triggers are the same as start, except that prepare won't be called beforehand, nothing will be enqueued to the DMA engine and nothing will happen (no audio). Removing the pause flag will let apps know that it isn't supported. Removing the resume flag will cause user space to call prepare and start instead of resume, so audio will continue playing when the system wakes up. Before removing the pause and resume flags, I tested this on an exynos 5250, using 'aplay -i'. Pause/un-pause leads to silence followed by a write error. Suspend/resume testing led to the same result. Removing the two flags fixes suspend/resume (since snd_pcm_prepare is called again). And leads to a proper reporting of pause not supported. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * ASoC: mc13783: Remove mono supportFabio Estevam2012-09-061-4/+4
| | | | | | | | | | | | | | | | | | | | Playing a mono track on a mc13783 codec results in incorrect playback rate. Remove mono support so that a mono track can be played correctly. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Tested-by: Gaëtan Carlier <gcembed@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: arizona: Fix typo in 44.1kHz ratesHeather Lomond2012-09-061-1/+1
| | | | | | | | | | Signed-off-by: Heather Lomond <hlomond@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: spear: correct the check for NULL dma_buffer pointerPrasad Joshi2012-08-311-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | The if condition if (!buf && !buf->area) checks if the buf pointer is NULL and then dereferences it again to check if the buffer area is NULL, resulting in possible NULL dereference. Signed-off-by: Prasad Joshi <prasadjoshi.linux@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * sound: tegra_alc5632: remove HP detect GPIO inversionStephen Warren2012-08-281-1/+0
| | | | | | | | | | | | | | | | | | | | | | Both the schematics and practical testing show that the HP detect GPIO is high when the headphones are plugged in. Hence, the snd_soc_jack_gpio should not specify to invert the signal. Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Andrey Danin <danindrey@mail.ru> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: <stable@vger.kernel.org> # v3.4 v3.5
| * ASoC: dapm: Don't force card bias level to be updatedMark Brown2012-08-251-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) means that any DAPM context being updated will have the bias level automatically set, including the card. We can't safely do this as the card callbacks are called for each device context and so the management of the card bias is more complex. Several multi-component cards rely on this behaviour. Skip updates during the asynchronous run entirely. We should really do them in the synchronous section but it's not 100% clear which values to pick as the different DAPM contexts may have different bias levels. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: dapm: Make sure we update the bias level for CODECs with no opMark Brown2012-08-251-0/+2
| | | | | | | | | | | | | | | | | | Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) ensures that we update non-CODEC DAPM contexts but means that if a CODEC has no set_bias_level() operation it'll not be updated. Fix that. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: am3517evm: fix error return codeJulia Lawall2012-08-201-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | It was forgotten to initialize ret to the result of calling snd_soc_dai_set_sysclk, unlike at the other calls in the same function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: ux500_msp_i2s: better use devm functions and fix error return codeJulia Lawall2012-08-201-20/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Remove unnecessary calls to devm_kfree and replace iounmap by devm_iounmap (and use resource_size for the third argument). These changes make it possible to remove the error-handling code at the end of ux500_msp_i2s_init_msp, and all of the gotos become direct returns. In the case of the second call to devm_kzalloc, the return variable ret was not initialized. Here it is changed to a direct return of -ENOMEM. A simplified version of the semantic match that finds the second problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: imx-sgtl5000: fix error return codeJulia Lawall2012-08-201-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Initialize ret on the second call to imx_audmux_v2_configure_port so that the subsequent test checks that result and not the previous one. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * Merge tag 'sound-3.6' of ↵Linus Torvalds2012-08-086-16/+21
| |\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "Containing only a few really small/trivial fixes. The only urgent fix is a regression fix of HDMI codec probing, introduced in 3.6-rc1. The rest are HD-audio specific fixes and a copule of minor bug fixes in PCM core and the old emu10k1." * tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: hda - Fix double quirk for Quanta FL1 / Lenovo Ideapad ALSA: hda - Fix ugly debug prints with CONFIG_SND_VERBOSE_PRINTK=y ALSA: hda - remove redundant auto quirks for conexant 506x ALSA: hda - remove quirk for Dell Vostro 1015 ALSA: hda - add dock support for Thinkpad X230 ALSA: hda - Fix regression of HDMI codec probing ALSA: hda - add dock support for Thinkpad T430s ALSA: emu10k1: Avoid access to invalid pages when period=1 ALSA: PCM: Fix possible memory leaks in the error path
| * \ Merge tag 'sound-3.6' of ↵Linus Torvalds2012-08-0319-13/+65
| |\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "A bunch of small fixes for ASoC, mainly against regressions due to the defaulting regmap i/o, in addition to a HD-audio fixup." * tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ASoC: core: Fix check before defaulting to regmap ALSA: hda - Support dock on Lenovo Thinkpad T530 with ALC269VC ASoC: wm8962: Allow VMID time to fully ramp ASoC: AC97 doesn't use regmap by default ASoC: sgtl5000: enable VAG_POWER for LINE_IN ASoC: ab8500: Inform SoC Core that we have our own I/O arrangements ASoC: omap: Add missing modules aliases to get sound working on omap devices sound: tegra_alc5632: Adjust to of_get_named_gpio() change sound: tegra_wm8903: Adjust to of_get_named_gpio() change ASoC: mc13783: Provide codec->control_data ASoC: ux500: Include the correct header files ASoC: wm8994: Hold runtime PM reference while handling mic and jack IRQs ASoC: sgtl5000: remove unneeded snd_soc_dapm_new_widgets in probe ASoC: mxs-saif: set a base clock rate for EXTMASTER mode work ASoC: mxs-saif: fix clock prepare and enable unbalance issue ASoC: wm8994: Ensure there are enough BCLKs for four channels
* | | | ALSA: hda - Yet another position_fix quirk for ASUS machinesTakashi Iwai2012-09-131-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ASUS X53S also suffers from the same issue as in commit c302d6133. Use POS_FIX_POSBUF for this hardware, too. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=47461 Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: ice1724: Use linear scale for AK4396 volume control.Matteo Frigo2012-09-121-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The AK4396 DAC has a linear-scale attentuator, but sound/pci/ice1712/prodigy_hifi.c used a log scale instead, which is not quite right. This patch restores the correct scale, borrowing from the ak4396 code in sound/pci/oxygen/oxygen.c. Signed-off-by: Matteo Frigo <athena@fftw.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: hda_intel: add position_fix quirk for Asus K53ECatalin Iacob2012-09-111-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit c20c5a841cbe47f5b7812b57bd25397497e5fbc0 changed some chipsets to default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including repeated sounds on my Asus laptop. My hardware is Cougar Point which the commit log of c20c5a841cbe47f5b7812b57bd25397497e5fbc0 mentions as tested so POS_FIX_COMBO probably works in general but apparently it doesn't on Asus K53E therefore the need for the quirk. Signed-off-by: Catalin Iacob <iacobcatalin@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: compress_core: fix open flags test in snd_compr_open()Dan Carpenter2012-09-111-5/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | O_RDONLY is zero so the original test (f->f_flags & O_RDONLY) is always false and it will never do compress capture. The test for O_WRONLY is also slightly off. The original test would consider "->flags = (O_WRONLY | O_RDWR)" as write only instead of rejecting it as invalid. I've also removed the pr_err() because that could flood dmesg. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: hda - Fix Oops at codec reset/reconfigTakashi Iwai2012-09-101-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_hda_codec_reset() calls restore_pincfgs() where the codec is powered up again, which eventually tries to resume and initialize via the callbacks of the codec. However, it's the place just after codec free callback, thus no codec callbacks should be called after that. On a codec like CS4206, it results in Oops due to the access in init callback. This patch fixes the issue by clearing the codec callbacks properly after freeing codec. Reported-by: Daniel J Blueman <daniel@quora.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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