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* ASoC: Mark cache as dirty when suspendingMark Brown2011-08-041-0/+1
| | | | | | | | | | | | | | | | | | commit 7be4ba24a3ea53bc8ade841635e4d4a59e98ceb5 upstream. Since quite a few drivers are not managing to flag the cache as needing to be resynced after suspend and it's a reasonable thing to do flag the cache as needing sync automatically when suspending. The expectation is that systems will mainly only keep the CODEC powered when doing audio through the CODEC so we won't actually suspend the device anyway; drivers which want to can override this behaviour when they resume. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ASoC: davinci: fix codec start and stop functionsRajashekhara, Sudhakar2011-08-041-4/+4
| | | | | | | | | | | | | | | | | | commit 3012f43eaf7592d8121426918e43e3b5db013aff upstream. According to DM365 voice codec data sheet at [1], before starting recording or playback, ADC/DAC modules should follow a reset and enable cycle. Writing a 1 to the ADC/DAC bit in the register resets the module and clearing the bit to 0 will enable the module. But the driver seems to be doing the reverse of it. [1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* ASoC: davinci: add missing break statementRajashekhara, Sudhakar2011-08-041-0/+1
| | | | | | | | | | | | | | commit 82d1d521036eb3f5aae48b847f939d99a44c18bb upstream. In davinci_vcif_trigger() function, a break() statement was missing causing the davinci_vcif_stop() function to be called as a fallback after calling davinci_vcif_start(). Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* Merge branch 'fix/asoc' of ↵Linus Torvalds2011-07-184-15/+16
|\ | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: ASoC: Correct WM8994 MICBIAS supply widget hookup ASoC: Fix shift in WM8958 accessory detection default implementation ASoC: sh: fsi-hdmi: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-ak4642: fixup snd_soc_card name
| * ASoC: Correct WM8994 MICBIAS supply widget hookupMark Brown2011-07-171-3/+4
| | | | | | | | | | | | | | | | | | | | | | | | The WM8994 and WM8958 series of devices have two MICBIAS supplies rather than one, the current widget actually manages the microphone detection control register bit (which is managed separately by the relevant API). Fix this, hooking the relevant supplies up to the MICBIAS1 and MICBIAS2 widgets. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| * ASoC: Fix shift in WM8958 accessory detection default implementationMark Brown2011-07-151-1/+1
| | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| * ASoC: sh: fsi-hdmi: fixup snd_soc_card nameKuninori Morimoto2011-07-091-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | it shouldn't contain space letters and special letters like parentheses. aplay will be "Segmentation fault" without this patch special thanks to Takashi. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: sh: fsi-da7210: fixup snd_soc_card nameKuninori Morimoto2011-07-091-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | it shouldn't contain space letters and special letters like parentheses. aplay will be "Segmentation fault" without this patch. special thanks to Takashi. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: sh: fsi-ak4642: fixup snd_soc_card nameKuninori Morimoto2011-07-091-8/+8
| | | | | | | | | | | | | | | | | | | | | | | | it shouldn't contain space letters and special letters like parentheses. aplay will be "Segmentation fault" without this patch. special thanks to Takashi. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | Merge branch 'fix/asoc' into for-linusTakashi Iwai2011-07-098-35/+45
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| * Merge branch 'for-3.0' of ↵Takashi Iwai2011-07-092-27/+4
| |\ | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc
| | * ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2Kuninori Morimoto2011-07-071-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | mask didn't cover update-data Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| | * ASoC: Manage WM8731 ACTIVE bit as a supply widgetMark Brown2011-07-051-26/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | Now we have supply widgets there's no need to open code the handling of the ACTIVE bit. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Tested-by: Nicolas Ferre <nicolas.ferre@atmel.com> Acked-by: Liam Girdwood <lrg@ti.com>
| * | ASoC: Don't set invalid name string to snd_card->driver fieldTakashi Iwai2011-07-051-2/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The snd_card->driver field contains a driver name string, and in general it shouldn't contain space or special letters. The commit 2b39535b9e54888649923beaab443af212b6c0fd changed the string copy from card->name, but the long name string may contain such letters, thus it may still lead to a segfault. A temporary fix is not to copy the long name string but just keep it empty as the earlier version did. Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge branch 'for-3.0' of ↵Takashi Iwai2011-07-053-2/+19
| |\ \ | | |/ | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc
| | * ASoC: Ensure we delay long enough for WM8994 FLL to lock when startingMark Brown2011-07-041-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | This delay is very conservative. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Cc: stable@kernel.org
| | * ASoC: Tegra: I2S: Ensure clock is enabled when writing regsStephen Warren2011-07-041-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The I2S controller needs a clock to respond to register writes. Without this, register writes will at worst hang the CPU. In practice, I've only observed writes being dropped. Luckily, the dropped register writes historically had no effect: TEGRA_I2S_TIMING: The value we wrote was the reset default. TEGRA_I2S_FIFO_SCR: The default was for the FIFOs to request more data when one slot was empty. The requested value was for the FIFOs to request when four slots were empty. The DMA controller in the mainline kernel is configured to burst a single entry at a time into the FIFO, hence there was no issue. The only negative effect was on bus efficiency losses due to an increased number of arbitration attempts. However, in various non-upstream changes, the DMA controller now bursts four entries at a time into the FIFO. If there is only space for one entry, the data is simply dropped. In practice, this resulted in 3/4 of samples being dropped, and playback at 4x the expected rate and pitch. By fixing the clocking issue, this is solved. Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Fix Blackfin I2S _pointer() implementation return in bounds valuesMark Brown2011-06-291-2/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Blackfin DMA controller can report one frame beyond the end of the buffer in the wraparound case but ALSA requires that the pointer always be in the buffer. Do the wraparound to handle this. A similar bug is likely to apply to the other Blackfin PCM drivers but the code is less obvious to inspection and I don't have a user to test. Reported-by: Kieran O'Leary <Kieran.O'Leary@wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| * | Merge branch 'for-3.0' of ↵Takashi Iwai2011-07-052-4/+19
| |\ \ | | |/ | |/| | | | git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into fix/asoc
| | * ASoC: tlv320aic3x: Do soft reset to codec when going to bias off stateJarkko Nikula2011-05-231-0/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | TLV320AIC33, TLV320AIC34 and I believe others too in this family have some hw bugs that cause that analogue and digital VDD supplies remain leaking up to a few mA of current after certain use cases even the hw blocks inside codec are driven to off. Highest leakages occur after using the bypass paths inside codec but it is possible to get smaller leakages just by toggling mute switches in unused audio paths (i.e. no DAPM changes) while codec is on due another active audio path. While some cases are able to workaroud by making sure that e.g. output mixer switches are muted before powering down the output stage this doesn't help all the cases. Therefore use the software reset command to clear possible leakage currents since that works in every cases and affects only this codec instance. Only drawback is that now cache sync is required everytime when codec bias comes out from bias off state, not only when supply regulators were off. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@ti.com>
| | * ASoC: tlv320aic3x: Don't sync first two registers from register cacheJarkko Nikula2011-05-231-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There is no need to sync first two registers from cache to hw after a reset. First one is used to select page for register access and this driver is normally accessing page 0 only. Second one does a software reset which is obviously unneeded after hardware or previous software reset command. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@ti.com>
| | * audio: tlv320aic26: fix PLL register configurationMichael Williamson2011-05-211-3/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The current PLL configuration code for the tlc320aic26 codec appears to assume a hardcoded system clock of 12 MHz. Use the clock value provided by the DAI_OPS API for the calculation. Tested using a MityDSP-L138 platform providing a 24.576 MHz clock. Signed-off-by: Michael Williamson <michael.williamson@criticallink.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@ti.com>
* | | ALSA: hda - Fix a copmile warningTakashi Iwai2011-07-091-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | It's harmless but annyoing. sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’: sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Change all ADCs for dual-adc switching mode for RealtekTakashi Iwai2011-07-071-10/+23
| | | | | | | | | | | | | | | | | | | | | | | | | | | When the dual-adc switching mode is active in Realtek auto-parser, we need to couple all ADCs as a single capture-volume. Currently, the volume control changes only the first ADC, thus others may remain silent. This patch fixes the problem. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hdspm - Fix compile warnings with PPCTakashi Iwai2011-06-301-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The char can be unsigned on some architectures. Since the code checks the negative values, they should be declared as signed char explicitly. sound/pci/rme9652/hdspm.c:5449: warning: comparison is always false due to limited range of data type sound/pci/rme9652/hdspm.c:5462: warning: comparison is always false due to limited range of data type Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: cs5535 - Fix invalid big-endian conversionsTakashi Iwai2011-06-301-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | Fix the wrongly converted short values: sound/pci/cs5535audio/cs5535audio_pcm.c:152: warning: large integer implicitly truncated to unsigned type sound/pci/cs5535audio/cs5535audio_pcm.c:160: warning: large integer implicitly truncated to unsigned type Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: HDMI - fix ELD monitor name lengthWu Fengguang2011-06-291-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | I noticed that the last character of the ELD monitor name is lost, this fixes the issue. This fix should be confirming to the HDA spec, and works together with the DRM part of the ELD patch. The HDA spec does not mention that Monitor_Name_String is an '\0' ending string, and it allows NML to be 1, which is only valid when MNL does not count the possible ending '\0'. Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: atmel - update author email for ABDAC, AC97C and AT73C213Hans-Christian Egtvedt2011-06-283-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | This patch updates the email address of the sound drivers supported by me to an email account I will use on a more regular basis in the future. Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: HDA: Add model=auto quirk for Acer Aspire 3830TGDavid Henningsson2011-06-281-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | Since we're not using the new auto parser as a fallback yet, add it manually as a quirk. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: HDA: Add a new Conexant codec ID (506c)David Henningsson2011-06-281-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Conexant ID 506c was found on Acer Aspire 3830TG. As users report no playback, sending to stable should be safe. Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/783582 Reported-by: andROOM Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | Merge branch 'fix/asoc' into for-linusTakashi Iwai2011-06-245-13/+5
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| * | ASoC: pxa-ssp: Correct check for stream presenceDaniel Mack2011-06-231-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Don't rely on the codec's channels_min information to decide wheter or not allocate a substream's DMA buffer. Rather check if the substream itself was allocated previously. Signed-off-by: Daniel Mack <zonque@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| * | ASoC: imx: add missing module informationsArnaud Patard (Rtp)2011-06-232-1/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | - add some modules aliases - add module license to avoid tainted kernel when loading the imx-pcm-audio driver Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org> Acked-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: imx: Remove unused Kconfig SND_MXC_SOC_SSI entryArnaud Patard (Rtp)2011-06-231-7/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | SND_MXC_SOC_SSI looks to be unused, so kill it. Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org> Acked-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: Remove unused and about to be broken SND_SOC_CUSTOM I/O busMark Brown2011-06-131-3/+0
| | | | | | | | | | | | | | | | | | | | | This will be removed in -next so let's drop it from mainline as soon as we can in order to minimise surprises. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ALSA: Remove unneeded version.h includes from sound/Jesper Juhl2011-06-242-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the sound/ directory there are two files (flagged by 'make versioncheck'); sound/pci/asihpi/asihpi.c and sound/soc/codecs/wm8991.c that include linux/version.h although they don't need it. This patch removes the unneeded includes. Signed-off-by: Jesper Juhl <jj@chaosbits.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: HDA: Pinfix quirk for HP Z200 WorkstationDavid Henningsson2011-06-231-0/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | BIOS lists the internal speaker as an internal line-out. Change to internal speaker + model=auto for better auto-mute capabilities. BugLink: http://bugs.launchpad.net/bugs/754964 Reported-by: Marc Legris <marc.legris@canonical.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: VIA HDA: Create a master amplifier control for VT1718S.Lydia Wang2011-06-211-0/+13
| | | | | | | | | | | | | | | | | | | | | Create a master volume and mute control of playback for VT1718S. Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: VIA HDA: Mute/unmute mixer conncted to Headphone for VT1718S.Lydia Wang2011-06-211-1/+12
| | | | | | | | | | | | | | | | | | | | | | | | When switch HP independent mode, mute/unmute connctions of mixer which is connected to headphone for VT1718S. Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: VIA HDA: Modify initial verbs list for VT1718S.Lydia Wang2011-06-211-7/+2
| | | | | | | | | | | | | | | | | | | | | | | | Remove some invalid initial verbs and correct some wrong initial verbs for VT1718S codec. Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Remove ALC268 model override for CPR2000Tony Vroon2011-06-211-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The "diverse" Quanta ID 0x0763 is overridden to ALC268_ACER. This keeps headphone automute and microphone input from operating on at least one laptop from Opti Systems. Without the override, the BIOS parser does a fine job setting the card up and everything works. Tested-By: Peter Schneider <e.at.chi.kaen@googlemail.com> Signed-off-by: Tony Vroon <tony@linx.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: HDA: Remove quirk for an HP deviceDavid Henningsson2011-06-201-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The reporter, who is running kernel 2.6.38, reports that he needs to set model=auto for the headphone output to work correctly. BugLink: http://bugs.launchpad.net/bugs/761022 Cc: stable@kernel.org (v2.6.38+) Reported-by: Jo Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Fix no NID error with VIA codecsTakashi Iwai2011-06-171-4/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The via driver spews warnigs like hda-codec: no NID for mapping control Independent HP:0:0 with some codecs because snd_hda_add_nid() is called with nid=0. This patch fixes it by skipping the call when no corresponding widget is found. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: isight: adjust for new queueing APIClemens Ladisch2011-06-171-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Since commit 13882a82ee16 (optimize iso queueing by setting wake only after the last packet), drivers are required to call fw_iso_context_queue_flush() after queueing a batch of packets. The missing call would have an effect only if the controller queue underruns, but then the DMA would stop completely. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: 6fire - Fix signedness bugTorsten Schenk2011-06-161-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fixed remaining issues of the signedness bug discovered by Dan Carpenter. A check was remaining that tests if unsigned rt->rate is >= 0. Changed that so that rt->rate now consistently uses ARRAY_SIZE(rates) as invalid rate value and not -1. Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: lola - Fix section mismatchTakashi Iwai2011-06-151-1/+1
| | | | | | | | | | | | | | | | | | Add missing __devinit. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Fix missing static inline to beep dummy functionTakashi Iwai2011-06-141-1/+1
| | | | | | | | | | | | | | | | | | | | | The commit 2308f4add3de9f6c9c9f02e49461e94d84bb200a missed static inline thus it resulted in multiple-definitions error at linking. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: 6fire: Fix double-free bug in usb6fire_fw_ezusb_upload()Jesper Juhl2011-06-141-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We have a double-free bug in sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload(). We already call release_firmware(fw) on line 258, so when we then do it again after usb6fire_fw_ezusb_write() returns <0, we have a double-free. Easily fixed by just removing the last call to release_firmware(). Signed-off-by: Jesper Juhl <jj@chaosbits.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Fix beep_device compilation warningsJoe Perches2011-06-131-2/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Using static inline functions can reduce compilation messages and macro misuse. sound/pci/hda/patch_conexant.c: In function ‘patch_cxt5045’: sound/pci/hda/patch_conexant.c:1232:3: warning: statement with no effect Signed-off-by: Joe Perches <joe@perches.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Fix wrong auto-mute type for Acer Aspire-oneTakashi Iwai2011-06-131-2/+1
| | | | | | | | | | | | | | | | | | | | | | | | The auto-mute setup for Acer Aspire-one with ALC268 was set wrongly during the clean-up of auto-mute function. Fixed now. Tested-by: Borislav Petkov <bp@alien8.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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