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* 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>
* | | ALSA: hdspm - Fix coding style in CTL_ELEM macrosAdrian Knoth2012-10-201-90/+90
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | checkpatch.pl discourages the use of spaces at the beginning of lines. Some of the CTL_ELEM defines were not properly indented. This patch replaces the leading spaces by tabs. No functionality is changed, the commit is purely cosmetic. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hdspm - Fix typo in kcontrol element on RME MADI cardsAdrian Knoth2012-10-201-1/+1
| | | | | | | | | | | | | | | Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hdspm - Fix sync_in detection on AES/AES32Adrian Knoth2012-10-201-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | According to the documentation, AES32 cards use a different bit position for reporting the sync_in status. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hdspm - Fix sync_in reporting on RME MADI cardsAdrian Knoth2012-10-201-0/+5
| | | | | | | | | | | | | | | | | | | | | | | | In contrast to AES32, MADI uses the first status register to report the sync_in status. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hdspm - Also report autosync_sample_rate on MADI and MADIfaceAdrian Knoth2012-10-201-0/+16
| | | | | | | | | | | | | | | | | | | | | | | | | | | MADI and MADIface used to report the autosync_sample_rate. This functionality was lost in commit 0dca1793063c28dde8f6c49c9c72203fe5cb6efc, this commit now adds it back. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hdspm - Fix reported autosync_sample_rateAdrian Knoth2012-10-201-1/+3
| | | | | | | | | | | | | | | | | | | | | | | | Missing breaks lead to a fall-through, thus causing the wrong autosync_sample_rate to be reported. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hdspm - Fix sync check reporting on all RME HDSPM cardsAdrian Knoth2012-10-201-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | Due to missing breaks and the resulting fall-through, card subtype selection was effectively missing, thus causing the wrong sync check functions to be called. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hdspm - Report external rate in slave mode on PCI MADIAdrian Knoth2012-10-201-4/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As a follow-up to a97bda7d29d02a2e9c6609d0947b15e55f5200e5, report the external sample rate as system_sample_rate when in slave mode. For PCIe MADI cards, the DDS value automatically contains the external sample rate, but the PCI version needs this manual workaround. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hdspm - Allow DDS/Varispeed to be set from userspaceAdrian Knoth2012-10-201-1/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The DDS value is the actual physical sample rate. We set it indirectly when selecting 44100, 48000 and so on via snd_hdspm_hw_params or hdspm_set_clock_source. This commit now allows the DDS value to be altered at runtime, thus speeding up or slowing down the physical sample rate. This is required for MADI's varispeed that allows for ±12.5% speed adjustment from the "selected" rate (32kHz, 44100kHz, 48kHz and so on). Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - add dock support for Thinkpad T430Stefán Freyr2012-10-191-0/+1
| |/ |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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> Cc: stable@vger.kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: emu10k1: add chip details for E-mu 1010 PCIe cardMaxim Kachur2012-10-171-0/+9
| | | | | | | | | | | | | | | | 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>
* | Merge tag 'asoc-3.7' of ↵Takashi Iwai2012-10-1711-62/+78
|\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v3.7 Nothing too exciting except for the ams-delta change which is relatively lerge due to the fact that the driver loading had been totally broken as the driver needed a newer API to function.
| * | ASoC: bells: Correct typo in sub speaker DAI name for WM5110Mark Brown2012-10-171-1/+1
| | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: codecs: da9055: Minor improvement in ALC calibration processAshish Chavan2012-10-111-1/+21
| | | | | | | | | | | | | | | | | | | | | | | | | | | This patch slightly improves ALC calibration process of da9055 codec driver by muting Mic PGAs during calibration. Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen@diasemi.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: dmaengine: Correct Makefile when sound is built as modulePeter Ujfalusi2012-10-091-2/+3
| | | | | | | | | | | | | | | | | | | | | | | | soc-dmaengine-pcm library need to be part of the snd-soc-core in order to be able to compile ASoC as modules when dmaengine is enabled on the platform. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: fsi: don't reschedule DMA from an atomic contextGuennadi Liakhovetski2012-10-091-7/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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> Cc: stable@vger.kernel.org
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