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* ALSA: ASoC: add DT bindings for CS4271Daniel Mack2012-09-281-3/+21
| | | | | | | | | Apart from pure matching, the bindings also support setting the the reset gpio line. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: Alexander Sverdlin <subaparts@yandex.ru> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm_hubs: Ensure volume updates are handled during class W startupMark Brown2012-09-281-0/+5
| | | | | | | | | 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> Cc: stable@vger.kernel.org
* ASoC: wm5110: Adding missing volume update bitsCharles Keepax2012-09-271-0/+4
| | | | | | | | 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> Cc: stable@vger.kernel.org
* ASoC: wm5110: Add OUT3R supportMark Brown2012-09-261-0/+4
| | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm5110: Add AEC loopback supportMark Brown2012-09-261-0/+24
| | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm5110: Rename EPOUT to HPOUT3Mark Brown2012-09-261-15/+30
| | | | | | | The third output on WM5110 is a general purpose headphone output which can be used to drive an earpice rather than a dedicated earpiece driver. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: arizona: Add more clock ratesMark Brown2012-09-261-0/+6
| | | | | | Some devices support additional clock rates. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: arizona: Add more DSP options for mixer input muxesMark Brown2012-09-262-1/+37
| | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm0010: Initialise chip state before we register the interruptMark Brown2012-09-261-5/+2
| | | | | | The interrupt handler uses the chip state. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm0010: Don't check if reset GPIO is defined when removingMark Brown2012-09-261-5/+2
| | | | | | We will fail to probe without one. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm0010: Allow slow GPIO for resetMark Brown2012-09-261-3/+5
| | | | | | | We never set the GPIO from atomic context so there's no reason why we can't support a GPIO that needs to sleep when configuring. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm5110: Enable bypass mode for MICVDDMark Brown2012-09-261-1/+1
| | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm5102: Enable bypass mode for MICVDDMark Brown2012-09-261-1/+1
| | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: dapm: Allow regulators to bypass as well as disable when idleMark Brown2012-09-261-2/+21
| | | | | | | | | | Allow regulators managed via DAPM to make use of the bypass support that has recently been added to the regulator API by setting a flag SND_SOC_DAPM_REGULATOR_BYPASS. When this flag is set the regulator will be put into bypass mode before being disabled, allowing the regulator to fall into bypass mode if it can't be disabled due to other users. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: cs4270: Remove mono supportFabio Estevam2012-09-251-2/+2
| | | | | | | | According to cs4270 datasheet, there is no reference to mono mode. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Acked-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Merge remote-tracking branch 'asoc/topic/ux500' into for-3.7Mark Brown2012-09-225-26/+205
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| * ASoC: Ux500: Minor coding layout changesLee Jones2012-09-201-5/+2
| | | | | | | | | | | | | | | | | | Includes removal of duplicate debug print affirming entry into the probe function, an unnecessary line break of a coding line <80 chars and a white space change (unintentional tab). Acked-by: Ola Lilja <ola.o.lilja@stericsson.com> Signed-off-by: Lee Jones <lee.jones@linaro.org>
| * ASoC: codecs: Enable AB8500 CODEC for Device TreeLee Jones2012-09-201-0/+81
| | | | | | | | | | | | | | | | | | | | We continue to allow the AB8500 CODEC to be registered via the AB8500 Multi Functional Device API, only this time we extract its configuration from the Device Tree binary. Acked-by: Ola Lilja <ola.o.lilja@stericsson.com> Acked-by: Linus Walleij <linus.walleij@linaro.org> Signed-off-by: Lee Jones <lee.jones@linaro.org>
| * ASoC: Ux500: Enable ux500 MSP driver for Device TreeLee Jones2012-09-202-3/+25
| | | | | | | | | | | | | | | | | | | | | | | | Register both parts of the MSP driver from Device Tree so that they are probed when Device Tree is enabled. Also, as there is platform data involved, we ensure that there is allocated memory to place the configuration into and that the correct information is extracted from the DT binary. Acked-by: Ola Lilja <ola.o.lilja@stericsson.com> Acked-by: Linus Walleij <linus.walleij@linaro.org> Signed-off-by: Lee Jones <lee.jones@linaro.org>
| * ASoC: Ux500: Enable MOP500 driver for Device TreeLee Jones2012-09-201-0/+40
| | | | | | | | | | | | | | | | | | | | Here we ensure that the MOP500 audio driver will be probed during a Device Tree boot. We also parse the sound node to link together the codec, dma and the CPU-side Digital Audio Interface. Acked-by: Ola Lilja <ola.o.lilja@stericsson.com> Acked-by: Linus Walleij <linus.walleij@linaro.org> Signed-off-by: Lee Jones <lee.jones@linaro.org>
| * ASoC: Ux500: Move MSP pinctrl setup into the MSP driverLee Jones2012-09-202-18/+57
| | | | | | | | | | | | | | | | | | | | | | | | | | In the initial submission of the MSP driver msp1 and msp3's associated pinctrl mechanism was passed back to platform code using a plat_init() call-back routine, but it has no place in platform code. The MSP driver should set this up for the appropriate ports. Instead we use a use_pinctrl identifier which is passed from platform_data/Device Tree which indicates which ports should use pinctrl. Acked-by: Ola Lilja <ola.o.lilja@stericsson.com> Acked-by: Linus Walleij <linus.walleij@linaro.org> Signed-off-by: Lee Jones <lee.jones@linaro.org>
* | ASoC: wm2000: Add regulator supportMark Brown2012-09-221-6/+47
| | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: wm2000: Convert to devm_regmap_init_i2c()Mark Brown2012-09-221-9/+4
| | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | Merge tag 'v3.6-rc6' into for-3.7Mark Brown2012-09-2220-78/+114
|\ \ | |/ | | | | | | | | | | Linux 3.6-rc6 has all our bug fixes. Conflicts (trivial overlap): sound/soc/omap/am3517evm.c
| * 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>
| * | 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>
| * | ALSA: usb-audio: Fix bogus error messages for delay accountingTakashi Iwai2012-09-061-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The recent fix for the missing fine delayed time adjustment gives strange error messages at each start of the playback stream, such as delay: estimated 0, actual 352 delay: estimated 353, actual 705 These come from the sanity check in retire_playback_urb(). Before the stream is activated via start_endpoints(), a few silent packets have been already sent. And at this point the delay account is still in the state as if the new packets are just queued, so the driver gets confused and spews the bogus error messages. For fixing the issue, we just need to check whether the received packet is valid, whether it's zero sized or not. Reported-by: Markus Trippelsdorf <markus@trippelsdorf.de> Cc: <stable@vger.kernel.org> [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - Fix missing Master volume for STAC9200/925xTakashi Iwai2012-09-061-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | With the commit [2faa3bf: ALSA: hda - Rewrite the mute-LED hook with vmaster hook in patch_sigmatel.c], the former Master volume control was converted to PCM. This was supposed to be covered by the vmaster control. But due to the lack of "PCM" slave definition, this didn't happen properly. The patch fixes the missing entry. Reported-by: Andrew Shadura <bugzilla@tut.by> Cc: <stable@vger.kernel.org> [v3.4+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: snd-usb: fix cross-interface streaming devicesDaniel Mack2012-08-311-0/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit 68e67f40b ("ALSA: snd-usb: move calls to usb_set_interface") saved us some unnecessary calls to snd_usb_set_interface() but ignored the fact that there is at least one device out there which operates on two endpoint in different interfaces simultaniously. Take care for this by catching the case where data and sync endpoints are located on different interfaces and calling snd_usb_set_interface() between the start of the two endpoints. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Robert M. Albrecht <linux@romal.de> Cc: stable@kernel.org [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: snd-usb: fix calls to next_packet_sizeDaniel Mack2012-08-313-13/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In order to support devices with implicit feedback streaming models, packet sizes are now stored with each individual urb, and the PCM handling code which fills the buffers purely relies on the size fields now. However, calling snd_usb_audio_next_packet_size() for all possible packets in an URB at once, prior to letting the PCM code do its job does in fact not lead to the same behaviour than what the old code did: The PCM code will break its loop once a period boundary is reached, consequently using up less packets that it really could. As snd_usb_audio_next_packet_size() implements a feedback mechanism to the endpoints phase accumulator, the number of calls to that function matters, and when called too often, the data rate runs out of bounds. Fix this by making the next_packet function public, and call it from the PCM code as before if the packet data sizes are not defined. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: stable@kernel.org [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: snd-usb: restore delay informationDaniel Mack2012-08-311-3/+26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Parts of commit 294c4fb8 ("ALSA: usb: refine delay information with USB frame counter") were unfortunately lost during the refactoring of the snd-usb driver in 3.5. This patch adds them back, restoring the correct delay information behaviour. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: stable@kernel.org [3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: snd-usb: use list_for_each_safe for endpoint resourcesPavel Roskin2012-08-311-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | snd_usb_endpoint_free() frees the structure that contains its argument. Signed-off-by: Pavel Roskin <proski@gnu.org> Cc: stable@vger.kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: snd-usb: Fix URB cancellation at stream startDaniel Mack2012-08-303-11/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream") fixed a scheduling-while-atomic bug that happened when snd_usb_endpoint_start was called from the trigger callback, which is an atmic context. However, the patch breaks the idea of the endpoints reference counting, which is the reason why the driver has been refactored lately. Revert that commit and let snd_usb_endpoint_start() take care of the URB cancellation again. As this function is called from both atomic and non-atomic context, add a flag to denote whether the function may sleep. Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: stable@kernel.org [3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - Don't trust codec EPSS bit for IDT 92HD83xx & coTakashi Iwai2012-08-281-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | These codecs seem reporting EPSS but require longer delay for the proper D3 transition. For example, D3_STOP_CLOCK_OK bit won't be set correctly even after D3. In this patch, codec->epss flag is overridden for avoid the misbehavior. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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