summaryrefslogtreecommitdiffstats
path: root/sound
Commit message (Collapse)AuthorAgeFilesLines
* ALSA: hda - adding a new NV HDMI/DP codec ID in the driverHui Wang2017-02-091-0/+1
| | | | | | | | | | | | Without this change, the HDMI/DP codec will be recognised as a generic codec, and there is no sound when playing through this codec. As suggested by NVidia side, after adding the new ID in the driver, the sound playing works well. Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: seq: Fix race at creating a queueTakashi Iwai2017-02-081-13/+20
| | | | | | | | | | | | | When a sequencer queue is created in snd_seq_queue_alloc(),it adds the new queue element to the public list before referencing it. Thus the queue might be deleted before the call of snd_seq_queue_use(), and it results in the use-after-free error, as spotted by syzkaller. The fix is to reference the queue object at the right time. Reported-by: Dmitry Vyukov <dvyukov@google.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Revert "ALSA: line6: Only determine control port properties if needed"Takashi Iwai2017-02-071-1/+2
| | | | | | | | | | | | | | | This reverts commit f6a0dd107ad0c8b59d1c9735eea4b8cb9f460949. The commit caused a regression on LINE6 Transport that has no control caps. Although reverting the commit may result back in a spurious error message for some device again, it's the simplest regression fix, hence it's taken as is at first. The further code fix will follow later. Fixes: f6a0dd107ad0 ("ALSA: line6: Only determine control port properties if needed") Reported-by: Igor Zinovev <zinigor@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: seq: Don't handle loop timeout at snd_seq_pool_done()Takashi Iwai2017-02-061-8/+1
| | | | | | | | | | | | | | | | | snd_seq_pool_done() syncs with closing of all opened threads, but it aborts the wait loop with a timeout, and proceeds to the release resource even if not all threads have been closed. The timeout was 5 seconds, and if you run a crazy stuff, it can exceed easily, and may result in the access of the invalid memory address -- this is what syzkaller detected in a bug report. As a fix, let the code graduate from naiveness, simply remove the loop timeout. BugLink: http://lkml.kernel.org/r/CACT4Y+YdhDV2H5LLzDTJDVF-qiYHUHhtRaW4rbb4gUhTCQB81w@mail.gmail.com Reported-by: Dmitry Vyukov <dvyukov@google.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge tag 'asoc-fix-v4.10-rc3' of ↵Takashi Iwai2017-01-1114-58/+143
|\ | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v4.10 As well as the usual smattering of driver specific fixes collected since the merge window this has one particularly important fix to the core for handling of aux_devs which was broken during the merge window by some of the componentization refactoring.
| *-----. Merge remote-tracking branches 'asoc/fix/nau8825', 'asoc/fix/rt5645', ↵Mark Brown2017-01-105-9/+26
| |\ \ \ \ | | | | | | | | | | | | | | | | | | 'asoc/fix/tlv320aic3x' and 'asoc/fix/topology' into asoc-linus
| | | | | * ASoC: topology: kfree kcontrol->private_value before freeing kcontrolColin Ian King2016-12-151-2/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | kcontrol->private_value is being kfree'd after kcontrol has been freed (in previous call to snd_ctl_remove). Instead, fix this by kfreeing the private_value before kcontrol. CoverityScan CID#1388311 "Read from pointer after free" Fixes: eea3dd4f1247a ("ASoC: topology: Only free TLV for volume mixers of a widget") Signed-off-by: Colin Ian King <colin.king@canonical.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | | * | ASoC: tlv320aic3x: Mark the RESET register as volatilePeter Ujfalusi2016-12-311-0/+13
| | | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The RESET register only have one self clearing bit and it should not be cached. If it is cached, when we sync the registers back to the chip we will initiate a software reset as well, which is not desirable. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Reviewed-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * / ASoC: rt5645: set sel_i2s_pre_div1 to 2Bard Liao2017-01-091-0/+3
| | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The i2s clock pre-divider 1 is used for both i2s1 and sysclk. The i2s1 is usually used for the main i2s and the pre-divider will be set in hw_params function. However, if i2s2 is used, the pre-divider is not set in the hw_params function and the default value of i2s clock pre-divider 1 is too high for sysclk and DMIC usage. Fix by overriding default divider value to 2. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=95681 Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Bard Liao <bardliao@realtek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | * | ASoC: nau8825: fix invalid configuration in Pre-Scalar of FLLJohn Hsu2016-12-312-2/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The clk_ref_div is not configured in the correct position of the register. The patch fixes that clk_ref_div, Pre-Scalar, is assigned the wrong value. Signed-off-by: John Hsu <KCHSU0@nuvoton.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | * | ASoC: nau8825: correct the function name of registerJohn Hsu2016-12-312-5/+5
| | |/ | | | | | | | | | | | | | | | | | | Change to correct name of the register function. Signed-off-by: John Hsu <KCHSU0@nuvoton.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | |
| | \
| | \
| | \
| | \
| | \
| | \
| | \
| *-------. | Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/dpcm', ↵Mark Brown2017-01-104-37/+91
| |\ \ \ \ \| | | | | | | | | | | | | | | | | | | | | | 'asoc/fix/dwc', 'asoc/fix/fsl-ssi' and 'asoc/fix/hdmi-codec' into asoc-linus
| | | | | * | ASoC: fsl_ssi: set fifo watermark to more reliable valueCaleb Crome2017-01-041-21/+53
| | | | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The fsl_ssi fifo watermark is by default set to 2 free spaces (i.e. activate DMA on FIFO when only 2 spaces are left.) This means the DMA must service the fifo within 2 audio samples, which is just not enough time for many use cases with high data rate. In many configurations the audio channel slips (causing l/r swap in stereo configurations, or channel slipping in multi-channel configurations). This patch gives more breathing room and allows the SSI to operate reliably by changing the fifio refill watermark to 8. There is no change in behavior for older chips (with an 8-deep fifo). Only the newer chips with a 15-deep fifo get the new behavior. I suspect a new fifo depth setting could be optimized on the older chips too, but I have not tested. Signed-off-by: Caleb Crome <caleb@crome.org> Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | | * / ASoC: dwc: Fix PIO mode initializationJose Abreu2016-12-141-14/+11
| | | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We can no longer rely on the return value of devm_snd_dmaengine_pcm_register(...) to check if the DMA handle is declared in the DT. Previously this check activated PIO mode but currently dma_request_chan returns either a valid channel or -EPROBE_DEFER. In order to activate PIO mode check instead if the interrupt line is declared. This reflects better what is documented in the DT bindings (see Documentation/devicetree/bindings/sound/ designware-i2s.txt). Also, initialize use_pio variable which was never being set causing PIO mode to never work. Signed-off-by: Jose Abreu <joabreu@synopsys.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * / ASoC: dpcm: Avoid putting stream state to STOP when FE stream is pausedPatrick Lai2017-01-061-1/+3
| | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When multiple front-ends are using the same back-end, putting state of a front-end to STOP state upon receiving pause command will result in backend stream getting released by DPCM framework unintentionally. In order to avoid backend to be released when another active front-end stream is present, put the stream state to PAUSED state instead of STOP state. Signed-off-by: Patrick Lai <plai@codeaurora.org> Signed-off-by: Mark Brown <broonie@kernel.org>
| | * | ASoC: wm_adsp: Don't overrun firmware file buffer when reading region dataRichard Fitzgerald2016-12-201-1/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Protect against corrupt firmware files by ensuring that the length we get for the data in a region actually lies within the available firmware file data buffer. Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * | | Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linusMark Brown2017-01-101-3/+1
| |\ \ \
| | * | | ASoC: rsnd: don't double free kctrlColin Ian King2016-12-151-3/+1
| | | |/ | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | On an error, snd_ctl_add already free's kctrl, so calling snd_ctl_free_one to free it again leads to a double free error. Fix this by removing the extraneous snd_ctl_free_one call. Issue found using static analysis with CoverityScan, CID 1372908 Signed-off-by: Colin Ian King <colin.king@canonical.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * | | Merge remote-tracking branch 'asoc/fix/intel' into asoc-linusMark Brown2017-01-103-4/+20
| |\ \ \
| | * | | ASoC: Intel: Skylake: Release FW ctx in cleanupJeeja KP2017-01-061-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Saved firmware ctx was not never released, so release Firmware ctx in cleanup routine. Signed-off-by: Jeeja KP <jeeja.kp@intel.com> Acked-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | * | | ASoC: Intel: bytcr-rt5640: fix settings in internal clock modePierre-Louis Bossart2017-01-051-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Frequency value of zero did not make sense, use same 24.576MHz setting and only change the clock source in idle mode Suggested-by: Bard Liao <bardliao@realtek.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | * | | ASoC: Intel: Skylake: Fix to fail safely if module not available in pathG Kranthi2016-12-311-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If a module is not available in a pipeline, fail safely rather than causing oops. Signed-off-by: G Kranthi <gudishax.kranthikumar@intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | * | | ASoC: Intel: bytcr_rt5640: fallback mechanism if MCLK is not enabledPierre-Louis Bossart2016-12-191-3/+13
| | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit df1a2776a795 ("ASoC: Intel: bytcr_rt5640: add MCLK support") was merged but the corresponding clock framework patches have not, after being bumped from audio to clock to x86 domains. The missing clock-related patches result in a regression starting with 4.9 with the audio card not being created. Rather than reverting this commit and all following updates already queued up for 4.10, handle run-time dependency on MCLK and fall back to the previous bit-clock mode. This provides the same functionality as in 4.8 for Baytrail devices. On Baytrail-CR most devices remain silent with this fallback but additional patches are needed anyway. As suggested by Mark Brown, the fallback is only allowed with -ENOENT, all other run-time errors, including -EPROBE_DEFER, will stop the probe with no sound card registered. This patch should be applied to -stable as well as ASoC 4.10 fixes Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| * | | Merge remote-tracking branch 'asoc/fix/component' into asoc-linusMark Brown2017-01-101-5/+5
| |\ \ \
| | * | | ASoC: Fix binding and probing of auxiliary componentsSylwester Nawrocki2016-12-311-5/+5
| | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently binding of auxiliary devices doesn't work as in soc_bind_aux_dev() function a bound component is not being added to any list and in soc_probe_aux_devices() we are trying to walk the component_dev_list list to probe auxiliary components but at that time this list doesn't contain any auxiliary components since they are being added to the card only in soc_probe_component(). This patch adds a list to the card where are stored bound but not probed auxiliary devices, so that all aux devices can be probed. Fixes: 1a653aa44725 "ASoC: core: replace aux_comp_list to component_dev_list" Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com> Signed-off-by: Mark Brown <broonie@kernel.org>
* | / / ALSA: usb-audio: Add a quirk for Plantronics BT600Dennis Kadioglu2017-01-101-0/+1
|/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Plantronics BT600 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x1" and "cannot get freq at ep 0x82". This patch adds the USB ID of the BT600 to quirks.c and avoids those error messages. Signed-off-by: Dennis Kadioglu <denk@post.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | Revert "ALSA: firewire-lib: change structure member with proper type"Takashi Sakamoto2017-01-052-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This reverts commit 6b7e95d1336b9eb0d4c6db190ce756480496bd13. This commit is based on a concern about value of the given parameter. It's expected to be ORed value with some enumeration-constants, thus often it can not be one of the enumeration-constants. I understood that this is out of specification and causes implementation-dependent issues. In C language specification, enumerated type can be interpreted as an integer type, in which all of enumeration-constants in corresponding enumerator-list can be stored. Implementations can select one of char, signed int and unsigned int as its type, and this selection is implementation-dependent. In GCC, a signed integer is selected when at least one of enumeration-constants has negative value, else an unsigned integer is selected. This behaviour can be switched by -fshort-enums to short type. Anyway, the type can be decided after scanning all of enumeration-constants. Totally, there's no rules to constrain the value of enumerated type to be one of enumeration-constants. In short, in enumerated type, decision of actual type for the type is the most important and enumeration-constants are just used for the decision, thus it's permitted to have an integer value in a range of enumeration-constants. In our case, actual type for the type is currently deterministic to be either char or unsigned int. Under GCC, it's unsigned int. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: usb-audio: test EP_FLAG_RUNNING at urb completionIoan-Adrian Ratiu2017-01-051-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Testing EP_FLAG_RUNNING in snd_complete_urb() before running the completion logic allows us to save a few cpu cycles by returning early, skipping the pending urb in case the stream was stopped; the stop logic handles the urb and sets the completion callbacks to NULL. Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: usb-audio: Fix irq/process data synchronizationIoan-Adrian Ratiu2017-01-053-16/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit 16200948d83 ("ALSA: usb-audio: Fix race at stopping the stream") was incomplete causing another more severe kernel panic, so it got reverted. This fixes both the original problem and its fallout kernel race/crash. The original fix is to move the endpoint member NULL clearing logic inside wait_clear_urbs() so the irq triggering the urb completion doesn't call retire_capture/playback_urb() after the NULL clearing and generate a panic. However this creates a new race between snd_usb_endpoint_start()'s call to wait_clear_urbs() and the irq urb completion handler which again calls retire_capture/playback_urb() leading to a new NULL dereference. We keep the EP deactivation code in snd_usb_endpoint_start() because removing it will break the EP reference counting (see [1] [2] for info), however we don't need the "can_sleep" mechanism anymore because a new function was introduced (snd_usb_endpoint_sync_pending_stop()) which synchronizes pending stops and gets called inside the pcm prepare callback. It also makes sense to remove can_sleep because it was also removed from deactivate_urbs() signature in [3] so we benefit from more simplification. [1] commit 015618b90 ("ALSA: snd-usb: Fix URB cancellation at stream start") [2] commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream") [3] commit ccc1696d5 ("ALSA: usb-audio: simplify endpoint deactivation code") Fixes: f8114f8583bb ("Revert "ALSA: usb-audio: Fix race at stopping the stream"") Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Apply asus-mode8 fixup to ASUS X71SLTakashi Iwai2017-01-041-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Although the old quirk table showed ASUS X71SL with ALC663 codec being compatible with asus-mode3 fixup, the bugzilla reporter explained that asus-model8 fits better for the dual headphone controls. So be it. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=191781 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: hda - Fix up GPIO for ASUS ROG RangerTakashi Iwai2017-01-041-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ASUS ROG Ranger VIII with ALC1150 codec requires the extra GPIO pin to up for the front panel. Just use the existing fixup for setting up the GPIO pins. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=189411 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: firewire-lib: change structure member with proper typeTakashi Sakamoto2017-01-032-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The 'amdtp_stream' structure is initialized by a call of 'amdtp_stream_init()'. Although a parameter of this function is for bit flags of packet attributes, its type is enumerator. This commit changes the type so that it's proper for a bit flags. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: firewire-tascam: Fix to handle error from initialization of stream dataTakashi Sakamoto2017-01-031-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This module has a bug not to return error code in a case that data structure for transmitted packets fails to be initialized. This commit fixes the bug. Fixes: 35efa5c489de ("ALSA: firewire-tascam: add streaming functionality") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: fireworks: fix asymmetric API call at unit removalTakashi Sakamoto2017-01-031-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ALSA fireworks driver has a bug not to call an API to destroy 'cmp_connection' structure for input direction. Currently this causes no issues because it just destroys 'mutex' structure, while it's better to fix it for future work. Fix: d23c2cc4485d ("ALSA: fireworks/bebob/dice/oxfw: allow stream destructor after releasing runtime") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ktime: Cleanup ktime_set() usageThomas Gleixner2016-12-253-5/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ktime_set(S,N) was required for the timespec storage type and is still useful for situations where a Seconds and Nanoseconds part of a time value needs to be converted. For anything where the Seconds argument is 0, this is pointless and can be replaced with a simple assignment. Signed-off-by: Thomas Gleixner <tglx@linutronix.de> Cc: Peter Zijlstra <peterz@infradead.org>
* | | ktime: Get rid of the unionThomas Gleixner2016-12-251-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ktime is a union because the initial implementation stored the time in scalar nanoseconds on 64 bit machine and in a endianess optimized timespec variant for 32bit machines. The Y2038 cleanup removed the timespec variant and switched everything to scalar nanoseconds. The union remained, but become completely pointless. Get rid of the union and just keep ktime_t as simple typedef of type s64. The conversion was done with coccinelle and some manual mopping up. Signed-off-by: Thomas Gleixner <tglx@linutronix.de> Cc: Peter Zijlstra <peterz@infradead.org>
* | | clocksource: Use a plain u64 instead of cycle_tThomas Gleixner2016-12-251-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There is no point in having an extra type for extra confusion. u64 is unambiguous. Conversion was done with the following coccinelle script: @rem@ @@ -typedef u64 cycle_t; @fix@ typedef cycle_t; @@ -cycle_t +u64 Signed-off-by: Thomas Gleixner <tglx@linutronix.de> Cc: Peter Zijlstra <peterz@infradead.org> Cc: John Stultz <john.stultz@linaro.org>
* | | Replace <asm/uaccess.h> with <linux/uaccess.h> globallyLinus Torvalds2016-12-247-7/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This was entirely automated, using the script by Al: PATT='^[[:blank:]]*#[[:blank:]]*include[[:blank:]]*<asm/uaccess.h>' sed -i -e "s!$PATT!#include <linux/uaccess.h>!" \ $(git grep -l "$PATT"|grep -v ^include/linux/uaccess.h) to do the replacement at the end of the merge window. Requested-by: Al Viro <viro@zeniv.linux.org.uk> Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
* | | Revert "ALSA: usb-audio: Fix race at stopping the stream"Takashi Iwai2016-12-211-8/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This reverts commit 16200948d8353fe29a473a394d7d26790deae0e7. The commit was intended to cover the race condition, but it introduced yet another regression for devices with the implicit feedback, leading to a kernel panic due to NULL-dereference in an irq context. As the race condition that was addressed by the commit is very rare and the regression is much worse, let's revert the commit for rc1, and fix the issue properly in a later patch. Fixes: 16200948d835 ("ALSA: usb-audio: Fix race at stopping the stream") Reported-by: Ioan-Adrian Ratiu <adi@adirat.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
* | | Merge tag 'sound-4.10-rc1' of ↵Linus Torvalds2016-12-14234-4475/+21271
|\ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "No dramatic changes are found in this development cycle, but as usual, many commits are applied in a wide range of drivers. Most of big changes are in ASoC, where a few bits of framework work and quite a lot of cleanups and improvements to existing code have been done. The rest are usual stuff, a few HD-audio and USB-audio quirks and fixes, as well as the drop of kthread usages in the whole subsystem. Below are some highlights: ASoC: - support for stereo DAPM controls - some initial work on the of-graph sound card - regmap conversions of the remaining AC'97 drivers - a new version of the topology ABI; this should be backward compatible - updates / cleanups of rsnd, sunxi, sti, nau8825, samsung, arizona, Intel skylake, atom-sst - new drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and Realtek RT5665 USB-audio: - yet another race fix at disconnection - tolerated packet size calculation for some Android devices - quirks for Axe-Fx II, QuickCam, TEAC 501/503 HD-audio: - improvement of Dell pin fixup mapping - quirks for HP Z1 Gen3, Alienware 15 R2 2016 and ALC622 headset mic Misc: - replace all kthread usages with simple works" * tag 'sound-4.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (296 commits) ALSA: hiface: Fix M2Tech hiFace driver sampling rate change ALSA: usb-audio: Eliminate noise at the start of DSD playback. ALSA: usb-audio: Add native DSD support for TEAC 501/503 DAC ASoC: wm_adsp: wm_adsp_buf_alloc should use kfree in error path ASoC: topology: avoid uninitialized kcontrol_type ALSA: usb-audio: Add QuickCam Communicate Deluxe/S7500 to volume_control_quirks ALSA: usb-audio: add implicit fb quirk for Axe-Fx II ASoC: zte: spdif: correct ZX_SPDIF_CLK_RAT define ASoC: zte: spdif and i2s drivers are not zx296702 specific ASoC: rsnd: setup BRGCKR/BRRA/BRRB when starting ASoC: rsnd: enable/disable ADG when suspend/resume timing ASoC: rsnd: tidyup ssi->usrcnt counter check in hw_params ALSA: cs46xx: add a new line ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture ASoC: nau8825: disable sinc filter for high THD of ADC ALSA: usb-audio: more tolerant packetsize ALSA: usb-audio: avoid setting of sample rate multiple times on bus ASoC: cs35l34: Simplify the logic to set CS35L34_MCLK_CTL setting ALSA: hda - Gate the mic jack on HP Z1 Gen3 AiO ALSA: hda: when comparing pin configurations, ignore assoc in addition to seq ...
| * | | ALSA: hiface: Fix M2Tech hiFace driver sampling rate changeJussi Laako2016-12-121-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Sampling rate changes after first set one are not reflected to the hardware, while driver and ALSA think the rate has been changed. Fix the problem by properly stopping the interface at the beginning of prepare call, allowing new rate to be set to the hardware. This keeps the hardware in sync with the driver. Signed-off-by: Jussi Laako <jussi@sonarnerd.net> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: usb-audio: Eliminate noise at the start of DSD playback.Nobutaka Okabe2016-12-121-1/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | [Problem] In some USB DACs, a terrible pop noise comes to be heard at the start of DSD playback (in the following situations). - play first DSD track - change from PCM track to DSD track - change from DSD64 track to DSD128 track (and etc...) - seek DSD track - Fast-Forward/Rewind DSD track [Cause] At the start of playback, there is a little silence. The silence bit pattern "0x69" is required on DSD mode, but it is not like that. [Solution] This patch adds DSD silence pattern to the endpoint settings. Signed-off-by: Nobutaka Okabe <nob77413@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: usb-audio: Add native DSD support for TEAC 501/503 DACNobutaka Okabe2016-12-121-0/+38
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds native DSD support for the following devices. - TEAC NT-503 - TEAC UD-503 - TEAC UD-501 (1) Add quirks for native DSD support for TEAC devices. (2) A specific vendor command is needed to switch between PCM/DOP and DSD mode, same as Denon/Marantz devices. Signed-off-by: Nobutaka Okabe <nob77413@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | Merge tag 'asoc-v4.10' of ↵Takashi Iwai2016-12-12200-4301/+20977
| |\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Updates for v4.10 There's been a few bits of framework work this time around and quite a lot of cleanups and improvements to existing code: - Support for stereo DAPM controls from Chen-yu Tsai. - Some initial work on the of-graph sound card from Morimoto-san, the main bulk of this is currently in binding review. - Lots of Renesas cleanups from Morimoto-san and sunxi work from Chen-yu Tsai. - regmap conversions of the remaining AC'97 drivers from Lars-Peter Clausen. - A new version of the topology ABI from Mengdong Lin. - New drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and Realtek RT5665.
| | | \
| | | \
| | | \
| | | \
| | *---. \ Merge remote-tracking branches 'asoc/topic/wm9712', 'asoc/topic/wm9713' and ↵Mark Brown2016-12-1212-120/+97
| | |\ \ \ \ | | | |_|_|/ | | |/| | | | | | | | | 'asoc/topic/zte' into asoc-next
| | | | | * ASoC: zte: spdif: correct ZX_SPDIF_CLK_RAT defineShawn Guo2016-12-081-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The macro ZX_SPDIF_CLK_RAT should be 2 instead of 4. With this fix, we can get correct audio output on HDMI through SPDIF interface. Signed-off-by: Shawn Guo <shawn.guo@linaro.org> Acked-by: Jun Nie <jun.nie@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | | | * ASoC: zte: spdif and i2s drivers are not zx296702 specificShawn Guo2016-12-084-10/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ZTE ZX SPDIF and I2S drivers can work on not only ZX296702 but also other ZTE ZX family SoCs like ZX296718, which is an arm64 platform. Let's make a few renaming and tweak the Kconfig a bit to get the drivers available for other ZTE ZX platforms. Signed-off-by: Shawn Guo <shawn.guo@linaro.org> Reviewed-by: Jun Nie <jun.nie@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | | * | ASoC: wm9713: Remove unused DAI ID definesLars-Peter Clausen2016-10-241-4/+0
| | | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used for matching instead and the defines are unused. The last user of these defines was removed in commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support"). So remove the defines as well. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * | ASoC: wm9712: Remove unused DAI ID definesLars-Peter Clausen2016-10-246-16/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used for matching instead and the defines are unused. The last user of these defines was removed in commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support"). So remove the defines as well. This also means the wm9712.h file no longer has any content and can be removed. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
| | | * | ASoC: wm9712: Remove ac97_read/ac97_write wrappersLars-Peter Clausen2016-10-211-26/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Since the regmap conversion ac97_read/ac97_write are just simple wrappers around snd_soc_read/snd_soc_write. Use those instead directly and remove the wrappers. Also use snd_soc_update_bits() were appropriate. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Tested-by: Marek Vasut <marex@denx.de> Acked-by: Marek Vasut <marex@denx.de> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-29 07:40:46 +0000