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* ALSA: emu10k1 - delay the PCM interrupts (add pcm_irq_delay parameter)Jaroslav Kysela2010-08-181-0/+1
| | | | | | | | | | | | | | | | | | With some hardware combinations, the PCM interrupts are acknowledged before the period boundary from the emu10k1 chip. The midlevel PCM code gets confused and the playback stream is interrupted. It seems that the interrupt processing shift by 2 samples is enough to fix this issue. This default value does not harm other, non-affected hardware. More information: Kernel bugzilla bug#16300 [A copmile warning fixed by tiwai] Signed-off-by: Jaroslav Kysela <perex@perex.cz> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* include: replace unifdef-y with header-ySam Ravnborg2010-08-141-5/+4Star
| | | | | | | | | unifdef-y and header-y has same semantic. So there is no need to have both. Drop the unifdef-y variant and sort all lines again Signed-off-by: Sam Ravnborg <sam@ravnborg.org>
* Merge branch 'for-linus' of ↵Linus Torvalds2010-08-087-8/+90
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (214 commits) ALSA: hda - Add pin-fix for HP dc5750 ALSA: als4000: Fix potentially invalid DMA mode setup ALSA: als4000: enable burst mode ALSA: hda - Fix initial capsrc selection in patch_alc269() ASoC: TWL4030: Capture route runtime DAPM ordering fix ALSA: hda - Add PC-beep whitelist for an Intel board ALSA: hda - More relax for pending period handling ALSA: hda - Define AC_FMT_* constants ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs ALSA: hda - Add support for HDMI HBR passthrough ALSA: hda - Set Stream Type in Stream Format according to AES0 ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF ASoC: wm9081: fix resource reclaim in wm9081_register error path ASoC: wm8978: fix a memory leak if a wm8978_register fail ASoC: wm8974: fix a memory leak if another WM8974 is registered ASoC: wm8961: fix resource reclaim in wm8961_register error path ASoC: wm8955: fix resource reclaim in wm8955_register error path ASoC: wm8940: fix a memory leak if wm8940_register return error ASoC: wm8904: fix resource reclaim in wm8904_register error path ...
| * Merge branch 'topic/misc' into for-linusTakashi Iwai2010-08-052-2/+10
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| | * Merge branch 'devel' of git://git.alsa-project.org/alsa-kernel into topic/miscTakashi Iwai2010-07-051-1/+1
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| | | * ALSA: pcm_lib: avoid timing jitter in snd_pcm_read/write()David Dillow2010-06-281-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When using poll() to wait for the next period -- or avail_min samples -- one gets a consistent delay for each system call that is usually just a little short of the selected period time. However, When using snd_pcm_read/write(), one gets a jittery delay that alternates between less than a millisecond and approximately two period times. This is caused by snd_pcm_lib_{read,write}1() transferring any available samples to the user's buffer and adjusting the application pointer prior to sleeping to the end of the current period. When the next period interrupt occurs, there is then less than avail_min samples remaining to be transferred in the period, so we end up sleeping until a second period occurs. This is solved by using runtime->twake as the number of samples needed for a wakeup in addition to selecting the proper wait queue to wake in snd_pcm_update_state(). This requires twake to be non-zero when used by snd_pcm_lib_{read,write}1() even if avail_min is zero. Signed-off-by: Dave Dillow <dave@thedillows.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| | * | ALSA: pcm: Define G723 3-bit and 5-bit formatsBen Collins2010-05-312-1/+9
| | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This defines the 24bps and 40bps (8khz sample rate) G.723 codec formats. They are going to be used once I submit the driver for an mpeg4/g723 compression card. I've updated the signed value to -1 as per Takashi's comments since these are non-linear formats. Signed-off-by: Ben Collins <bcollins@bluecherry.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge branch 'for-2.6.36' of ↵Takashi Iwai2010-08-021-0/+2
| |\ \ | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
| | * | ASoC: tlv320dac33: Add support for automatic FIFO configurationPeter Ujfalusi2010-07-291-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Platform parameter to enable automatic FIFO configuration when the codec is in Mode1 or Mode7 FIFO mode. When this mode is selected, the controls for changing nSample (in Mode1), and UTHR (in Mode7) are not added. The driver configures the FIFO configuration based on the stream's period size in a way, that every burst will read period size of data from the host. In Mode7 we need to use a formula, which gives close enough aproximation for the burst length from the host point of view. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | ASoC: tlv320dac33: Revisit the FIFO Mode1 handlingPeter Ujfalusi2010-07-291-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Replace the hardwired latency definition with platform data parameter, and simplify the nSample parameter calculation. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * | | ASoC: fsi: Add new funtion for SPDIFKuninori Morimoto2010-07-291-0/+2
| |/ / | | | | | | | | | | | | | | | Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: fsi: Add specified ID for soc-audioKuninori Morimoto2010-07-171-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | Specified ID is necessary, when some codecs are used with FSI. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: fsi: Fixup for master modeKuninori Morimoto2010-07-131-0/+32
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch add hw_params to snd_soc_dai_ops, because board specific set_rate is needed when FSI was used as master mode. This patch remove fsi_clk_ctrl from fsi_dai_startup, because clock should be disabled before set_rate. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: fsi: modify format area definition on flagsKuninori Morimoto2010-07-131-6/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | There is no necessity that each bit in this area has the meaning. This patch modify it to sequence number Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: uda134x: replace a macro with a value in platform struct.Vladimir Zapolskiy2010-06-251-0/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | This change wipes out a hardcoded macro, which enables codec bias level control. Now is_powered_on_standby value shall be used instead. Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: Add SND_SOC_DAPM_PRE_POST_PMD eventapatard@mandriva.com2010-05-311-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | Some systems codecs need to configure some registers before and after powering down some of their part. As a convenience add a macro for that. Signed-off-by: Arnaud Patard <apatard@mandriva.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | Merge commit 'v2.6.35-rc1' into for-2.6.36Mark Brown2010-05-316-30/+32
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| * | ASoC: Add SOC_DOUBLE_R_SX_TLV controlapatard@mandriva.com2010-05-161-0/+21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch is adding a new control which has the following capabilities: - tlv - variable data size (for instance, 7 ou 8 bit) - double mixer - data range centered around 0 Signed-off-by: Arnaud Patard <apatard@mandriva.com> Acked-by: Liam Girdwood <lrg@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | pm_qos: Get rid of the allocation in pm_qos_add_request()James Bottomley2010-07-191-1/+1
| |/ |/| | | | | | | | | | | | | | | | | | | | | All current users of pm_qos_add_request() have the ability to supply the memory required by the pm_qos routines, so make them do this and eliminate the kmalloc() with pm_qos_add_request(). This has the double benefit of making the call never fail and allowing it to be called from atomic context. Signed-off-by: James Bottomley <James.Bottomley@suse.de> Signed-off-by: mark gross <markgross@thegnar.org> Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
* | Merge branch 'for-linus' of ↵Linus Torvalds2010-05-2015-49/+522
|\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (250 commits) ALSA: hda: Storage class should be before const qualifier ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT ASoC: sdp4430 - add sdp4430 pcm ops to DAI. ASoC: TWL6040: Enable earphone path in codec ASoC: SDP4430: Add support for Earphone speaker ASoC: SDP4430: Add sdp4430 machine driver ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function ALSA: sound/pci/asihpi: Use kzalloc ALSA: hdmi - dont fail on extra nodes ALSA: intelhdmi - add id for the CougarPoint chipset ALSA: intelhdmi - user friendly codec name ALSA: intelhdmi - add dependency on SND_DYNAMIC_MINORS ALSA: asihpi: incorrect range check ALSA: asihpi: testing the wrong variable ALSA: es1688: add pedantic range checks ARM: McBSP: Add support for omap4 in McBSP driver ARM: McBSP: Fix request for irq in OMAP4 OMAP: McBSP: Add 32-bit mode support ...
| * | Merge branch 'topic/asoc' into for-linusTakashi Iwai2010-05-2010-21/+485
| |\| | | | | | | | | | | | | Conflicts: sound/soc/codecs/ad1938.c
| | * ASoC: core: Fix for the volume limiting when invert is in usePeter Ujfalusi2010-05-111-11/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If the register for the volume needs invert, than the inversion need to be done from the chip maximum, and not from the platform dependent limit. Introduce soc_mixer_control.platform_max value, which initially equals to chip maximum. The snd_soc_limit_volume function only modify the platform_max, all volsw_info call returns this as well. The .max value holds the chip default (maximum), and it is used for the inversion, if it is needed. Additional check in the volsw_info call has been added to check the validity of the platform_max in case, when custom macros used by codec drivers are not initializing it correctly. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Allow DAI links to be kept active over suspendMark Brown2010-05-101-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As well as allowing DAPM pins to be marked as ignoring suspend allow DAI links to be similarly marked. This is primarily intended for digital links between CODECs and non-CPU devices such as basebands in mobile phones and will suppress all suspend calls for the DAI link. It is likely that this will need to be revisited if used with devices which are part of the SoC CPU. Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Support leaving paths enabled over system suspendMark Brown2010-05-101-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some devices can usefully run audio while the Linux system is suspended. One of the most common examples is smartphone systems, which are normally designed to allow audio to be run between the baseband and the CODEC without passing through the CPU and so can suspend the CPU when on a voice call for additional power savings. Support such systems by providing an API snd_soc_dapm_ignore_suspend(). This can be used to mark DAPM endpoints as not being sensitive to system suspend. When the system is being suspended paths between endpoints which are marked as ignoring suspend will be kept active. Both source and sink must be marked, and there must already be an active path between the two endpoints prior to suspend. When paths are active over suspend the bias management will hold the device bias in the ON state. This is used to avoid suspending the CODEC while it is still in use. Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Remove unused DAPM suspend flagMark Brown2010-05-101-1/+0Star
| | | | | | | | | | | | | | | | | | | | | | | | | | | We now manage suspend within the main power analysis rather than by flipping the state of widgets. Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * Revert "ASoC: tpa6130a2: Support for limiting gain"Peter Ujfalusi2010-05-071-1/+0Star
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This reverts commit 6f3991152f20933b77eff30413e893bf1a15e578. Since core has now support for limiting the volume on controls this patch is not needed. Furthermore, this patch actually prevents the core to set new volume on the TPA. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: core: Support for limiting the volumePeter Ujfalusi2010-05-071-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add support for the core to limit the maximum volume on an existing control. The function will modify the soc_mixer_control.max value of the given control. The new value must be lower than the original one (chip maximum) If there is a need for limiting a gain on a given control, than machine drivers can do the following in their snd_soc_dai_link.init function: snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21); This will modify the original 31 (chip maximum) to 21, so user space will not be able to set the gain higher than this. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * Merge branch 'for-2.6.35' of ↵Takashi Iwai2010-05-062-0/+18
| | |\ | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
| | | * ASoC: tpa6130a2: Support for limiting gainPeter Ujfalusi2010-05-061-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add support for platform dependent gain limiting on the tpa6130a2 (and tpa6140a2) Headset amplifier. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | * ASoC: tlv320aic3x: Add platform data and reset gpio handlingJarkko Nikula2010-05-061-0/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Handle the reset GPIO within the codec driver in order to follow the startup protocol for the tlv320aic3x codecs. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | ASoC: Add WM9090 amplifier driverMark Brown2010-04-301-0/+28
| | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM9090 is a high performance low power audio subsystem, including headphone and class D speaker drivers. Note that this driver is a standalone CODEC driver and so is only immediately suitable for use with the WM9090 as a standalone sound card taking line inputs, or with a DAC with no software control. The pending ASoC multi-CODEC support will expand the range of systems that can use the driver, or system-specific adaptations can be made. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: UDA134X: Add UDA1345 CODEC supportVladimir Zapolskiy2010-04-261-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds support for Philips UDA1345 CODEC. The CODEC has only volume control, de-emphasis, mute, DC filtering and power control features. Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Add indirection for CODEC private dataMark Brown2010-04-171-1/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | One of the features of the multi CODEC work is that it embeds a struct device in the CODEC to provide diagnostics via a sysfs class rather than via the device tree, at which point it's much better to use the struct device private data rather than having two places to store it. Provide an accessor function to allow this change to be made more easily, and update all the CODEC drivers are updated. To ensure use of the accessor the private data structure member is renamed, meaning that if code developed with older an older core that still uses private_data is merged it will fail to build. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * Merge branch 'for-2.6.34' into for-2.6.35Mark Brown2010-04-051-1/+1
| | |\ | | | | | | | | | | | | | | | | Conflicts due to context changes next to the backported DMA data change: include/sound/soc.h
| | * | ASoC: Add a notifier for jack status changesMark Brown2010-03-221-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some systems provide both mechanical and electrical detection of jack status changes. On such systems power savings can be achieved by only enabling the electrical detection methods when physical insertion has been detected. Begin supporting such systems by providing a notifier for jack status changes which can be used to trigger any reconfiguration. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_streamDaniel Mack2010-03-192-1/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This fixes a memory corruption when ASoC devices are used in full-duplex mode. Specifically for pxa-ssp code, where this pointer is dynamically allocated for each direction and destroyed upon each stream start. All other platforms are fixed blindly, I couldn't even compile-test them. Sorry for any breakage I may have caused. Reported-by: Sven Neumann <s.neumann@raumfeld.com> Reported-by: Michael Hirsch <m.hirsch@raumfeld.com> Signed-off-by: Daniel Mack <daniel@caiaq.de> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | Merge branch 'topic/jack' into for-2.6.35Mark Brown2010-03-191-0/+8
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| | * | | ASoC: Support GPIO based microphone detection for WM8904Mark Brown2010-03-161-0/+36
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM8904 allows microphone detection signals to be brought out as alternate functions of the GPIO signals which can be detected using interrupt inputs on the CPU. Allow this to be configured using platform data. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | ASoC: Allow configuration of WM8904 GPIO pin functionsMark Brown2010-03-161-2/+72
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Provide platform data allowing the configuration of the GPIO pins on the WM8904 to be selected, allowing alternate functions to be enabled. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | ASoC: Implement interrupt driven microphone detection for WM8903Mark Brown2010-03-161-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Support use of the WM8903 IRQ for reporting of microphone presence and short detection. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | ASoC: Add WM8903 interrupt supportMark Brown2010-03-161-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently used to detect completion of the write sequencer. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | ASoC: Initial WM8903 microphone bias and short detectionMark Brown2010-03-161-0/+29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Provide support for WM8903 microphone presence and short detection using the GPIOs to route out a logic signal suitable for handling using snd_soc_jack_add_gpios() on the processor GPIOs. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | ASoC: Add GPIO configuration support for WM8903Mark Brown2010-03-161-0/+216
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Allow users to pass in a default configuration for the GPIOs of the WM8903 as platform data. This allows configuration of the pin muxing of the device. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | ASoC: Allow pins to be force enabledMark Brown2010-03-161-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Allow pins to be forced on regardless of their power state. This is intended for use with microphone bias supplies which need to be enabled in order to support microphone detection - in systems without appropriate hardware leaving the microphone unbiased when not in use saves power. The force done at power check time in order to avoid disrupting other power detection logic. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | ASoC: Remove unused 'muted' flag from DAPM widgetsMark Brown2010-03-161-1/+0Star
| | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | ASoC: tlv320dac33: Add option for keeping the BCLK runningPeter Ujfalusi2010-03-121-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Platform data option for the codec to keep the BCLK clock continuously running in FIFO modes (codec master). OMAP3 McBSP when in slave mode needs continuous BCLK running on the serial bus in order to operate correctly. Since in FIFO mode the DAC33 can also shut down the BCLK clock and enable it only when it is needed, let the platforms decide if the CPU side needs the BCLK running or not. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | Merge commit 'v2.6.34-rc1' into for-2.6.35Mark Brown2010-03-107-6/+64
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| | * | | | ASoC: Remove unused pmdown_time flagMark Brown2010-03-051-1/+0Star
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The flag is no longer used in the code so it just wastes a bit. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | | ASoC: Add support for WM8960 capless modeMark Brown2010-03-031-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM8960 headphone outputs can be run in capless mode with OUT3 used to drive a pseudo ground for the headphone drivers. In this mode the mono mixer is not used, the mixer should be turned on in concert with the headphone output drivers and the device bias levels are managed differently. Also tweak the existing bias management to remove the use of active discharge while we're at it since that's often audible. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | | ASoC: Move WM8960 platform data into include/soundMark Brown2010-03-031-0/+22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Avoids machine files having to peer into sound/soc which is a bit rude and icky. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
generated by cgit v1.2.3-55-g7522 (git 2.39.0) at 2024-07-23 01:55:41 +0200