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* switch SNDRV_PCM_IOCTL_LINK to fget_light()Al Viro2012-09-271-6/+7
| | | | Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
* Merge tag 'asoc-3.6' of ↵Takashi Iwai2012-09-1511-40/+21
|\ | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Updates for 3.6 A bigger set of updates than I'm entirely comfortable with - things backed up a bit due to travel. As ever the majority of these are small, focused updates for specific drivers though there are a couple of core changes. There's been good exposure in -next. The AT91 patch fixes a build break.
| * ASoC: wm8904: correct the indexBo Shen2012-09-141-1/+1
| | | | | | | | | | Signed-off-by: Bo Shen <voice.shen@atmel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: tegra: fix maxburst settings in dmaengine codeStephen Warren2012-09-071-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | The I2S controllers are programmed with an "attention" level of 4 DWORDs. This must match the configuration passed to the DMA driver, so that when they burst in data, they don't overflow the available FIFO space. Also, the burst size is relevant to the destination for playback, and source for capture, not vice-versa as originally written. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * ASoC: samsung dma - Don't indicate support for pause/resume.Dylan Reid2012-09-061-7/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The pause and resume operations indicate that the stream can be un-paused/resumed from the exact location they were paused/suspended. This is not true for this driver, the pause and suspend triggers share the same code path with stop, they flush all pending DMA transfers. This drops all pending samples. The pause_release/resume triggers are the same as start, except that prepare won't be called beforehand, nothing will be enqueued to the DMA engine and nothing will happen (no audio). Removing the pause flag will let apps know that it isn't supported. Removing the resume flag will cause user space to call prepare and start instead of resume, so audio will continue playing when the system wakes up. Before removing the pause and resume flags, I tested this on an exynos 5250, using 'aplay -i'. Pause/un-pause leads to silence followed by a write error. Suspend/resume testing led to the same result. Removing the two flags fixes suspend/resume (since snd_pcm_prepare is called again). And leads to a proper reporting of pause not supported. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * ASoC: mc13783: Remove mono supportFabio Estevam2012-09-061-4/+4
| | | | | | | | | | | | | | | | | | | | Playing a mono track on a mc13783 codec results in incorrect playback rate. Remove mono support so that a mono track can be played correctly. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Tested-by: Gaëtan Carlier <gcembed@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: arizona: Fix typo in 44.1kHz ratesHeather Lomond2012-09-061-1/+1
| | | | | | | | | | Signed-off-by: Heather Lomond <hlomond@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: spear: correct the check for NULL dma_buffer pointerPrasad Joshi2012-08-311-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | The if condition if (!buf && !buf->area) checks if the buf pointer is NULL and then dereferences it again to check if the buffer area is NULL, resulting in possible NULL dereference. Signed-off-by: Prasad Joshi <prasadjoshi.linux@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * sound: tegra_alc5632: remove HP detect GPIO inversionStephen Warren2012-08-281-1/+0
| | | | | | | | | | | | | | | | | | | | | | Both the schematics and practical testing show that the HP detect GPIO is high when the headphones are plugged in. Hence, the snd_soc_jack_gpio should not specify to invert the signal. Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Andrey Danin <danindrey@mail.ru> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: <stable@vger.kernel.org> # v3.4 v3.5
| * ASoC: dapm: Don't force card bias level to be updatedMark Brown2012-08-251-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) means that any DAPM context being updated will have the bias level automatically set, including the card. We can't safely do this as the card callbacks are called for each device context and so the management of the card bias is more complex. Several multi-component cards rely on this behaviour. Skip updates during the asynchronous run entirely. We should really do them in the synchronous section but it's not 100% clear which values to pick as the different DAPM contexts may have different bias levels. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: dapm: Make sure we update the bias level for CODECs with no opMark Brown2012-08-251-0/+2
| | | | | | | | | | | | | | | | | | Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) ensures that we update non-CODEC DAPM contexts but means that if a CODEC has no set_bias_level() operation it'll not be updated. Fix that. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: am3517evm: fix error return codeJulia Lawall2012-08-201-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | It was forgotten to initialize ret to the result of calling snd_soc_dai_set_sysclk, unlike at the other calls in the same function. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: ux500_msp_i2s: better use devm functions and fix error return codeJulia Lawall2012-08-201-20/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Remove unnecessary calls to devm_kfree and replace iounmap by devm_iounmap (and use resource_size for the third argument). These changes make it possible to remove the error-handling code at the end of ux500_msp_i2s_init_msp, and all of the gotos become direct returns. In the case of the second call to devm_kzalloc, the return variable ret was not initialized. Here it is changed to a direct return of -ENOMEM. A simplified version of the semantic match that finds the second problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: imx-sgtl5000: fix error return codeJulia Lawall2012-08-201-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Initialize ret on the second call to imx_audmux_v2_configure_port so that the subsequent test checks that result and not the previous one. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> ( if@p1 (\(ret < 0\|ret != 0\)) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret *if(...) { ... when != ret = e2 when forall return ret; } // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * Merge tag 'sound-3.6' of ↵Linus Torvalds2012-08-086-16/+21
| |\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "Containing only a few really small/trivial fixes. The only urgent fix is a regression fix of HDMI codec probing, introduced in 3.6-rc1. The rest are HD-audio specific fixes and a copule of minor bug fixes in PCM core and the old emu10k1." * tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: hda - Fix double quirk for Quanta FL1 / Lenovo Ideapad ALSA: hda - Fix ugly debug prints with CONFIG_SND_VERBOSE_PRINTK=y ALSA: hda - remove redundant auto quirks for conexant 506x ALSA: hda - remove quirk for Dell Vostro 1015 ALSA: hda - add dock support for Thinkpad X230 ALSA: hda - Fix regression of HDMI codec probing ALSA: hda - add dock support for Thinkpad T430s ALSA: emu10k1: Avoid access to invalid pages when period=1 ALSA: PCM: Fix possible memory leaks in the error path
| * \ Merge tag 'sound-3.6' of ↵Linus Torvalds2012-08-0319-13/+65
| |\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "A bunch of small fixes for ASoC, mainly against regressions due to the defaulting regmap i/o, in addition to a HD-audio fixup." * tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ASoC: core: Fix check before defaulting to regmap ALSA: hda - Support dock on Lenovo Thinkpad T530 with ALC269VC ASoC: wm8962: Allow VMID time to fully ramp ASoC: AC97 doesn't use regmap by default ASoC: sgtl5000: enable VAG_POWER for LINE_IN ASoC: ab8500: Inform SoC Core that we have our own I/O arrangements ASoC: omap: Add missing modules aliases to get sound working on omap devices sound: tegra_alc5632: Adjust to of_get_named_gpio() change sound: tegra_wm8903: Adjust to of_get_named_gpio() change ASoC: mc13783: Provide codec->control_data ASoC: ux500: Include the correct header files ASoC: wm8994: Hold runtime PM reference while handling mic and jack IRQs ASoC: sgtl5000: remove unneeded snd_soc_dapm_new_widgets in probe ASoC: mxs-saif: set a base clock rate for EXTMASTER mode work ASoC: mxs-saif: fix clock prepare and enable unbalance issue ASoC: wm8994: Ensure there are enough BCLKs for four channels
* | | | ALSA: hda - Yet another position_fix quirk for ASUS machinesTakashi Iwai2012-09-131-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ASUS X53S also suffers from the same issue as in commit c302d6133. Use POS_FIX_POSBUF for this hardware, too. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=47461 Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: ice1724: Use linear scale for AK4396 volume control.Matteo Frigo2012-09-121-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The AK4396 DAC has a linear-scale attentuator, but sound/pci/ice1712/prodigy_hifi.c used a log scale instead, which is not quite right. This patch restores the correct scale, borrowing from the ak4396 code in sound/pci/oxygen/oxygen.c. Signed-off-by: Matteo Frigo <athena@fftw.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: hda_intel: add position_fix quirk for Asus K53ECatalin Iacob2012-09-111-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit c20c5a841cbe47f5b7812b57bd25397497e5fbc0 changed some chipsets to default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including repeated sounds on my Asus laptop. My hardware is Cougar Point which the commit log of c20c5a841cbe47f5b7812b57bd25397497e5fbc0 mentions as tested so POS_FIX_COMBO probably works in general but apparently it doesn't on Asus K53E therefore the need for the quirk. Signed-off-by: Catalin Iacob <iacobcatalin@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: compress_core: fix open flags test in snd_compr_open()Dan Carpenter2012-09-111-5/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | O_RDONLY is zero so the original test (f->f_flags & O_RDONLY) is always false and it will never do compress capture. The test for O_WRONLY is also slightly off. The original test would consider "->flags = (O_WRONLY | O_RDWR)" as write only instead of rejecting it as invalid. I've also removed the pr_err() because that could flood dmesg. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: hda - Fix Oops at codec reset/reconfigTakashi Iwai2012-09-101-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_hda_codec_reset() calls restore_pincfgs() where the codec is powered up again, which eventually tries to resume and initialize via the callbacks of the codec. However, it's the place just after codec free callback, thus no codec callbacks should be called after that. On a codec like CS4206, it results in Oops due to the access in init callback. This patch fixes the issue by clearing the codec callbacks properly after freeing codec. Reported-by: Daniel J Blueman <daniel@quora.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | 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>
* | | | ALSA: hda - Avoid unnecessary parameter read for EPSSTakashi Iwai2012-08-282-2/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | EPSS parameter should be static, so we can read it once and remember. This also allows more easily to override the wrong EPSS capability reported from a codec by changing the flag in the codec initialization step. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: hda - Do not set GPIOs for speakers on IDT if there are no speakersDavid Henningsson2012-08-221-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This fixes an issue with a machine where there were no speakers, but GPIO0 had to be data=1 for the headphone to be functioning. I'm not sure if we need a more advanced patch to solve all possible cases, but if so, this patch would still provide a minor optimisation. BugLink: https://bugs.launchpad.net/bugs/1040077 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: snd-als100: fix suspend/resumeOndrej Zary2012-08-211-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_card_als100_probe() does not set pcm field in struct snd_sb. As a result, PCM is not suspended and applications don't know that they need to resume the playback. Tested with Labway A381-F20 card (ALS120). Signed-off-by: Ondrej Zary <linux@rainbow-software.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | ALSA: hda - Fix leftover codec->power_transitionTakashi Iwai2012-08-201-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When the codec turn-on operation is canceled by the immediate power-on, the driver left the power_transition flag as is. This caused the persistent avoidance of power-save behavior. Cc: <stable@vger.kernel.org> [v3.5+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | Merge tag 'asoc-3.6' of ↵Takashi Iwai2012-08-2032-267/+160
|\ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Additional updates for 3.6 A batch more bugfixes, all driver-specific and fairly small and unremarkable in a global context. The biggest batch are for the newly added Arizona drivers.
| * | | | ASoC: wm9712: Fix inverted capture volumeMark Brown2012-08-171-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The capture volume increases with the register value so it shouldn't be flagged as inverted. Reported-by: Christoph Fritz <chf.fritz@googlemail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: wm9712: Fix microphone source selectionMark Brown2012-08-171-2/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently the microphone input source is not selectable as while there is a DAPM widget it's not connected to anything so it won't be properly instantiated. Add something more correct for the input structure to get things going, even though it's not hooked into the rest of the routing map and so won't actually achieve anything except allowing the relevant register bits to be written. Reported-by: Christop Fritz <chf.fritz@googlemail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | | | ASoC: wm5102: Remove DRC2Mark Brown2012-08-171-16/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | It will be removed from future device revisions. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: jack: Always notify full jack statusMark Brown2012-08-131-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Don't just notify for the bits we've updated, notify the full state of the jack otherwise users might get confused by misleading reports. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: wm5110: Add missing input PGA routesMark Brown2012-08-131-0/+12
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: wm5102: Add missing input PGA routesMark Brown2012-08-131-0/+9
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: Samsung: Fix build errorSachin Kamat2012-08-101-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fixes the following build error: In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0, from arch/arm/plat-samsung/include/plat/dma-ops.h:17, from arch/arm/plat-samsung/include/plat/dma.h:128, from sound/soc/samsung/pcm.c:23: arch/arm/plat-samsung/include/plat/dma-pl330.h:106:8: error: redefinition of ‘struct s3c2410_dma_client’ arch/arm/plat-samsung/include/plat/dma.h:40:8: note: originally defined here make[3]: *** [sound/soc/samsung/pcm.o] Error 1 Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com> Acked-by: Kukjin Kim <kgene.kim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: core: Upgrade the severity of probe deferral errors to dev_err()Mark Brown2012-08-091-4/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the past when ASoC had a custom probe deferral mechanism people complained about the logspam it generated and didn't want to know about the fact that we were doing probe deferral so all the error messages for it were at dev_dbg(), making diagnostics hard. Now that we have probe deferral as an accepted thing and it's generating log messages anyway there's no need to worry about this so upgrade the severity of all the probe deferral sources to dev_err() so that they are displayed by default. Also add one for missing aux_devs since there wasn't one. Reported-by: Russell King <linux@arm.linux.org.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: wm8994: Add missing dapm routes for WM8958 rev AChris Rattray2012-08-091-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: wm8962: Don't duplicate bias level management in resumeMark Brown2012-08-091-15/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The core will bring the bias level up for us since we use idle_bias_off, duplicating this may be harmful. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: bfin: fix memory leak in sport3 controller driverScott Jiang2012-08-091-0/+7
| | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: Davinci: McASP: Flush the FIFO before enablingVaibhav Bedia2012-08-091-2/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | FIFO should be flushed before it is enabled for the first time. This fixes the I/O errors reported by the ASoC core on a fresh boot Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com> Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: imx-ssi: Remove mono supportFabio Estevam2012-08-081-2/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Playing a mono track results in incorrect playback rate, ie, the audio is played at a faster rate. Remove mono support in the driver by setting 'channes_min' to dual-channel and this allows mono tracks to be played correctly. Reported-by: Gaëtan Carlier <gcembed@gmail.com> Tested-by: Gaëtan Carlier <gcembed@gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: mxs: Fix the name of the SoC familyFabio Estevam2012-08-081-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | SND_SOC_MXS_SGTL5000 is used on MXS boards, so fix the SoC family name. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: omap-mcbsp: Fix 6pin mux configurationPeter Ujfalusi2012-08-071-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The check for the mux_signal callback was wrong which prevents us to configure the 6pin port's FSR/CLKR signal mux. Reported-by: CF Adad <cfadad@rocketmail.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org (3.4+)
| * | | | Merge tag 'asoc-3.6' of ↵Mark Brown2012-08-0418-13/+64
| |\ \ \ \ | | |/ / / | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-3.6 ASoC: Additional updates for 3.6 A few updates for issues discovered during the merge window, the main one being the fix for the issues with defaulting to use of regmap without properly checking if there was I/O in place already.
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-17 00:45:53 +0100