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* ARM: mach-shmobile: ap4evb: FSI clock use proper process for HDMIKuninori Morimoto2010-11-241-2/+4
| | | | | | | | | | | Current AP4 FSI set_rate function used bogus clock process which didn't care enable/disable and clk->usecound. To solve this issue, this patch also modify FSI driver to call set_rate with enough options. This patch modify it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Paul Mundt <lethal@linux-sh.org>
* Merge branch 'for-linus' of ↵Linus Torvalds2010-10-2513-168/+360
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: (365 commits) ALSA: hda - Disable sticky PCM stream assignment for AD codecs ALSA: usb - Creative USB X-Fi volume knob support ALSA: ca0106: Use card specific dac id for mute controls. ALSA: ca0106: Allow different sound cards to use different SPI channel mappings. ALSA: ca0106: Create a nice spot for mapping channels to dacs. ALSA: ca0106: Move enabling of front dac out of hardcoded setup sequence. ALSA: ca0106: Pull out dac powering routine into separate function. ALSA: ca0106 - add Sound Blaster 5.1vx info. ASoC: tlv320dac33: Use usleep_range for delays ALSA: usb-audio: add Novation Launchpad support ALSA: hda - Add workarounds for CT-IBG controllers ALSA: hda - Fix wrong TLV mute bit for STAC/IDT codecs ASoC: tpa6130a2: Error handling for broken chip ASoC: max98088: Staticise m98088_eq_band ASoC: soc-core: Fix codec->name memory leak ALSA: hda - Apply ideapad quirk to Acer laptops with Cxt5066 ALSA: hda - Add some workarounds for Creative IBG ALSA: hda - Fix wrong SPDIF NID assignment for CA0110 ALSA: hda - Fix codec rename rules for ALC662-compatible codecs ALSA: hda - Add alc_init_jacks() call to other codecs ...
| * Merge branch 'topic/hda' into for-linusTakashi Iwai2010-10-251-1/+3
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| | * ALSA: tlv - Define numbers in sound/tlv.hTakashi Iwai2010-10-171-1/+3
| | | | | | | | | | | | Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge branch 'topic/asoc' into for-linusTakashi Iwai2010-10-258-165/+349
| |\ \ | | | | | | | | | | | | | | | | Conflicts: arch/powerpc/platforms/85xx/p1022_ds.c
| | * | ASoC: Restore MAX98088 CODEC driverMark Brown2010-10-181-0/+50
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This reverts commit f6765502f8daae3d237a394889276c8987f3e299 and adds the missing include file. Signed-off-by: Peter Hsiang <Peter.Hsiang@maxim-ic.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | ASoC: don't register AC97 devices twiceMika Westerberg2010-10-131-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | With generic AC97 ASoC glue driver (codec/ac97.c), we get following warning when the device is registered (slightly stripped the backtrace): kobject (c5a863e8): tried to init an initialized object, something is seriously wrong. [<c00254fc>] (unwind_backtrace+0x0/0xec) [<c014fad0>] (kobject_init+0x38/0x70) [<c0171e94>] (device_initialize+0x20/0x70) [<c017267c>] (device_register+0xc/0x18) [<bf20db70>] (snd_soc_instantiate_cards+0x924/0xacc [snd_soc_core]) [<bf20e0d0>] (snd_soc_register_platform+0x16c/0x198 [snd_soc_core]) [<c0175304>] (platform_drv_probe+0x18/0x1c) [<c0174454>] (driver_probe_device+0xb0/0x16c) [<c017456c>] (__driver_attach+0x5c/0x7c) [<c0173cec>] (bus_for_each_dev+0x48/0x78) [<c0173600>] (bus_add_driver+0x98/0x214) [<c0174834>] (driver_register+0xa4/0x130) [<c001f410>] (do_one_initcall+0xd0/0x1a4) [<c0062ddc>] (sys_init_module+0x12b0/0x1454) This happens because the generic AC97 glue driver creates its codec->ac97 via calling snd_ac97_mixer(). snd_ac97_mixer() provides own version of snd_device.register which handles the device registration when snd_card_register() is called. To avoid registering the AC97 device twice, we add a new flag to the snd_soc_codec: ac97_created which tells whether the AC97 device was created by SoC subsystem. Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | ASoC: Use delayed work for debounce of GPIO based jacksMark Brown2010-10-071-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Rather than block the workqueue by sleeping to do the debounce use delayed work to implement the debounce time. This should also means that we extend the debounce time on each new bounce, potentially allowing shorter debounce times for clean insertions. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | ASoC: Add support for WM8962 GPIO outputsMark Brown2010-10-021-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM8962 features five GPIOs, add support for controlling their output state via gpiolib. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | ASoC: Provide microphone bias configuration for WM8962Mark Brown2010-09-301-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add the widget for MICBIAS power control and allow configuration of the microphone bias setup via the platform data for the WM8962. When microphone status signals are brought out to GPIO this should be sufficient to enable microphone detection. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | ASoC: Initial WM8962 IRQ supportMark Brown2010-09-291-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Provide an initial hookup for interrupts on the WM8962. Currently we simply report error status via log messages if an IRQ is provided for the device. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | ASoC: Add event variants of the AIF widgetsMark Brown2010-09-061-0/+10
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | ASoC: fsi: modify compile errorKuninori Morimoto2010-08-311-3/+0
| | | | | | | | | | | | | | | | | | | | | | | | 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: Swap bias level enumerationJarkko Nikula2010-08-311-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Swapping the bias level enumeration is only meant to help debugging. It is easier if number 0 means bias off and bigger number means bigger bias level. Signed-off-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 'for-2.6.36' into for-2.6.37Mark Brown2010-08-163-8/+15
| | |\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fairly simple conflicts, the most serious ones are the i.MX ones which I suspect now need another rename. Conflicts: arch/arm/mach-mx2/clock_imx27.c arch/arm/mach-mx2/devices.c arch/arm/mach-omap2/board-rx51-peripherals.c arch/arm/mach-omap2/board-zoom2.c sound/soc/fsl/mpc5200_dma.c sound/soc/fsl/mpc5200_dma.h sound/soc/fsl/mpc8610_hpcd.c sound/soc/pxa/spitz.c
| | * \ \ Merge branch 'topic/multi-component' of ↵Mark Brown2010-08-126-160/+254
| | |\ \ \ | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into for-2.6.37
| | | * | | ASoC: multi-component - ASoC Multi-Component SupportLiam Girdwood2010-08-126-160/+254
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * | | | ASoC: Add initial WM8962 CODEC driverMark Brown2010-08-051-0/+23
| | |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The WM8962 is a low power, high performance stereo CODEC designed for portable digital audio applications. This initial driver release supports the key audio paths of the WM8962. Extended functionality, such as microphone detection, digital microphones and the advanced DSP signal enhancements provided by the device are not yet supported. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ALSA: emu10k1: Fix warning: "CCR" redefinedNobuhiro Iwamatsu2010-10-181-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | CCR is defined in emu10k1, but SuperH is defined too. If user use this driver with SuperH, it becomes a double definition. Signed-off-by: Nobuhiro Iwamatsu <nobuhiro.iwamatsu.yj@renesas.com> Cc: Paul Mundt <lethal@linux-sh.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: fix unused warnings with snd_power_get_stateMike Frysinger2010-10-171-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If we compile the ASoC code with PM disabled, we hit stuff like: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_suspend_check': sound/soc/soc-dapm.c:440: warning: unused variable 'codec' So tweak the stub macro to avoid these issues. Signed-off-by: Mike Frysinger <vapier@gentoo.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: Add more jack button slotsMark Brown2010-09-071-1/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some devices have more flexible microphone detection and can detect a wider range of buttons. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: pcm midlevel code - add time check for double interrupt acknowledgeJaroslav Kysela2010-08-181-0/+1
| | |_|/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The current code in pcm_lib.c do all checks using only the position in the ring buffer. Unfortunately, where the interrupts gets delayed or merged into one, we need another timing source to check when the buffer size boundary overlaps to avoid the wrong updating of the ring buffer pointers. This code uses jiffies to check the right time window without any performance impact. Signed-off-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* / | | driver core: remove CONFIG_SYSFS_DEPRECATED_V2 but keep it for block devicesKay Sievers2010-10-221-6/+0
|/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch removes the old CONFIG_SYSFS_DEPRECATED_V2 config option, but it keeps the logic around to handle block devices in the old manner as some people like to run new kernel versions on old (pre 2007/2008) distros. Signed-off-by: Kay Sievers <kay.sievers@vrfy.org> Cc: Jens Axboe <axboe@kernel.dk> Cc: Stephen Hemminger <shemminger@vyatta.com> Cc: "Eric W. Biederman" <ebiederm@xmission.com> Cc: Alan Stern <stern@rowland.harvard.edu> Cc: "James E.J. Bottomley" <James.Bottomley@suse.de> Cc: Andrew Morton <akpm@linux-foundation.org> Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru> Cc: Randy Dunlap <randy.dunlap@oracle.com> Cc: Tejun Heo <tj@kernel.org> Cc: "David S. Miller" <davem@davemloft.net> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.de> Cc: Ingo Molnar <mingo@elte.hu> Cc: Peter Zijlstra <a.p.zijlstra@chello.nl> Cc: David Howells <dhowells@redhat.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* | / 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/+4
| | | | | | | | | | | | | | | | | | 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 Torvalds2010-08-077-8/+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
| | |\
| | | * 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/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | 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/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
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