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* Merge branch 'for-linus' of ↵Linus Torvalds2010-08-071-5/+110
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 ...
| * ASoC: Handle read failures in codec_regMark Brown2010-07-141-5/+17
| | | | | | | | | | | | | | | | | | | | | | When a device is powered down volatile registers can't be read so attempts to display codec_reg will show error values, and obviously it is also possible for there to be hardware errors too. Check for errors from reads and display them more clearly when formatting codec_reg. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * ASoC: Fix overflow bug in SOC_DOUBLE_R_SX_TLVStuart Longland2010-06-191-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When SX_TLV widgets are read, if the gain is set to a value below 0dB, the mixer control is erroniously read as being at maximum volume. The value read out of the CODEC register is never sign-extended, and when the minimum value is subtracted (read; added, since the minimum is negative) the result is a number greater than the maximum allowed value for the control, and hence it saturates. Solution: Mask the result so that it "wraps around", emulating sign-extension. Signed-off-by: Stuart Longland <redhatter@gentoo.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * ASoC: Pay attention to write errors in volsw_2r_sxMark Brown2010-06-161-4/+2
| | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * Merge commit 'v2.6.35-rc1' into for-2.6.36Mark Brown2010-05-311-0/+1
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| * | ASoC: Add SOC_DOUBLE_R_SX_TLV controlapatard@mandriva.com2010-05-161-0/+95
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | fix typos concerning "initiali[zs]e"Uwe Kleine-König2010-06-161-1/+1
| |/ |/| | | | | | | Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de> Signed-off-by: Jiri Kosina <jkosina@suse.cz>
* | Merge branch 'topic/asoc' into for-linusTakashi Iwai2010-05-201-46/+209
|\ \ | |/ | | | | | | Conflicts: sound/soc/codecs/ad1938.c
| * ASoC: core: Fix for the volume limiting when invert is in usePeter Ujfalusi2010-05-111-9/+21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2010-05-101-1/+37
| | | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2010-05-101-4/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Refactor DAPM suspend handlingMark Brown2010-05-101-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Instead of using stream events to handle power down during suspend integrate the handling with the normal widget path checking by replacing all cases where we report a connected endpoint in a path with a function snd_soc_dapm_suspend_check() which looks at the ALSA power state for the card and reports false if we are in a D3 state. Since the core moves us into D3 prior to initating the suspend all power checks during suspend will cause the widgets to be powered down. In order to ensure that widgets are powered up on resume set the card to D2 at the start of resume handling (ALSA API calls require D0 so we are still protected against userspace access). 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: core: Support for limiting the volumePeter Ujfalusi2010-05-071-0/+39
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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-2.6.34' into for-2.6.35Mark Brown2010-03-291-1/+2
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| * | ASoC: remove a card from the list, if instantiation failedGuennadi Liakhovetski2010-03-191-14/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If instantiation of a card failed, we still have to remove it from the card list on unregistration. This fixes an Oops on Migo-R, triggering, when after a failed firmware load attempt the driver modules are removed and re-inserted again. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ASoC: PCM_RATE: Check for KNOT and CONTINUOUS flagsJassi Brar2010-03-121-0/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | For ASoC, if either CPU or CODEC driver has set the flag, the MACHINE driver should be given a chance to figure out if the dai, that set the flag, can accomodate a rate that it does not explicitly specify but is specified by the dai at the other end of the link. Signed-off-by: Jassi Brar <jassi.brar@samsung.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 Brown2010-03-101-5/+15
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| * | | ASoC: core: Add delay operation to snd_soc_dai_opsPeter Ujfalusi2010-03-031-0/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The delay callback can be used by the core to query the delay on the dai caused by FIFO or delay in the platform side. In case if both CPU and CODEC dai has FIFO the delay reported by each will be added to form the full delay on the chain. If none of the dai has FIFO, than the delay will be kept as zero. 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: soc level wrapper for pcm_pointer callbackPeter Ujfalusi2010-03-031-1/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Create a soc level wrapper for pcm_pointer callback. This will facilitate the soc level handling of different HW buffers in the audio path. 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: fix tailing whitespace in soc_pcm_apply_symmetryPeter Ujfalusi2010-03-031-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | My editor removes the tailing spaces, which causes problems when changing the soc-core.c Removing the space. 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 mulitple usage count of codec and cpu daiJassi Brar2010-02-261-13/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If we are to have a snd_soc_dai i.e, cpu_dai and codec_dai, shared among two or more dai_links we need to log the number of active users of the dai. For that, we change semantics of the snd_soc_dai.active flag from indicator to reference counter. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Remove runtime field from DAIjassi brar2010-02-221-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In order for having snd_soc_dais shared among two or more dai_links, remove the relatively global runtime field from the struct snd_soc_dai Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Pass dai_link as argument to platform suspend and resumejassi brar2010-02-221-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Passing pointer to relevant dai_link provides easier reach to the ASoC tree in suspend/resume of snd_soc_platform. It also provides direct access to the dai at the other end of the dai_link. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | Merge branch 'for-linus' of ↵Linus Torvalds2010-04-071-1/+2
|\ \ \ \ | | |_|/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: ALSA: mixart: range checking proc file ALSA: hda - Fix a wrong array range check in patch_realtek.c ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream ALSA: hda - Enable amplifiers on Acer Inspire 6530G ASoC: Only do WM8994 bias off transition from standby ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction ASoC: Support second DC servo readback method for wm_hubs ASoC: Avoid wraparound in wm_hubs DC servo correction ALSA: echoaudio - Eliminate use after free ALSA: i2c: cleanup: change parameter to pointer ALSA: hda - Add MSI blacklist for Aopen MZ915-M ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code ALSA: hda - Update document about MSI and interrupts ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981 ALSA: hda - Add missing printk argument in previous patch ASoC: Fix passing platform_data to ac97 bus users and fix a leak ALSA: hda - Fix ADC/MUX assignment of ALC269 codec ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo() ASoC: wm8994: playback => capture
| * | | ASoC: Fix passing platform_data to ac97 bus users and fix a leakGraham Gower2010-03-291-1/+2
| | |/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | [The issue is an attempt to write the pdata without the AC97 device allocated when using ac97.c - also added a comment in soc-core.c for the special case for ac97. -- broonie] Signed-off-by: Graham Gower <graham.gower@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | include cleanup: Update gfp.h and slab.h includes to prepare for breaking ↵Tejun Heo2010-03-301-0/+1
|/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
* | ASoC: soc_pcm_open: Add missing bailout tagJassi Brar2010-03-031-5/+9
| | | | | | | | | | | | | | | | | | The codec_dai needs to be shutdown should the machine startup fails. This patch adds another bailout tag for that case and rename the tag for configuration failures. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: core: On resume also check the soc device statePeter Ujfalusi2010-02-221-0/+6
|/ | | | | | | | | | | Check the card->codec on soc_resume to detect if the soc device is properly initialized. If the card->codec is NULL, than do not continue the resume operation, since the device is not initialized properly. 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: Make pmdown_time a longMark Brown2010-02-171-1/+1
| | | | | | | Fixes a warning. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* ASoC: Make pmdown_time runtime configurableMark Brown2010-02-161-0/+27
| | | | | | | | Provide a sysfs file allowing userspace to inspect and change the pmdown_time setting at runtime. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* ASoC: Make pmdown_time a per-card settingMark Brown2010-02-161-1/+3
| | | | | | | | | | Make the pmdown_time a per-card setting rather than a global one, initialised before the card initialisation runs. This allows cards to override the default setting if it makes sense to do so (for example, due to an unavoidable pop). Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* ASoC: Set codec->dev for AC97 devicesMark Brown2010-01-281-0/+1
| | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* Merge branch 'for-2.6.33' into for-2.6.34Mark Brown2010-01-121-1/+1
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| * const: constify remaining dev_pm_opsAlexey Dobriyan2009-12-151-1/+1
| | | | | | | | | | | | Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
* | ASoC: Use snprintf() when generating stream namesMark Brown2009-12-311-2/+2
| | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | ASoC: Export snd_soc_update_bits_unlocked()Mark Brown2009-12-041-3/+4
|/ | | | | | | Allows custom controls to use it. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* ASoC: move setting ac97 platformdata earlier than ac97 read/writeBarry Song2009-11-121-4/+5
| | | | | | | | | | | | | | | | | | | | | While probing, AC97 codec drivers and soc-core generically execute the following sequence: snd_soc_new_ac97_codec -> snd_soc_new_pcms -> reset ac-link/read AC97 ID to detect ->... -> set platform_data to ac97 by soc-core commit 474828a40f6ddab6e2a3475a19c5c84aa3ec7d60 adds platform_data to snd_ac97 instance. But ac97 platform data hasn't given to snd_ac97 before actual ac97 operations. Then while ac97_read access platform_data of snd_ac97 for detecting, NULL pointer oops will fire. That means old platform_data patch doesn't work in real-life cases. This patch moves the operation of setting ac97 platform_data earlier than ac97 reading/writing operations. Then it makes platform_data of AC97 become practically useful. Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: Factor out snd_soc_init_card()Mark Brown2009-11-031-77/+64
| | | | | | | | | snd_soc_init_card() is always called as the last part of the CODEC probe function so we can factor it out into the core card setup rather than have each CODEC replicate the code to do the initialiastation. This will be required to support multiple CODECs per card. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: Move sysfs and debugfs functions to head of soc-core.cMark Brown2009-11-031-167/+167
| | | | | | | A fairly hefty change in diff terms but no actual code changes, will be used by the next commit. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: refactor snd_soc_update_bits()Eero Nurkkala2009-10-301-6/+30
| | | | | | | | | | | | Introduce a wrapper call snd_soc_update_bits_locked() that will take the codec mutex. This call is used when the codec mutex is not already taken. Drivers calling snd_soc_update_bits() may wish to make sure the codec mutex is taken from the driver. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: remove io_mutexEero Nurkkala2009-10-301-5/+0
| | | | | | | | | | | | Remove the io_mutex. It has a drawback of serializing all accesses to snd_soc_update_bits() even when multiple codecs are in use. In addition, it fails to actually do its task - during snd_soc_update_bits(), dapm_update_bits() may also be accessing the same register which may result in an outdated register value. Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Merge branch 'for-2.6.32' into for-2.6.33Mark Brown2009-10-191-3/+8
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| * ASoC: Fix possible codec_dai->ops NULL pointer problemsBarry Song2009-10-191-3/+8
| | | | | | | | | | | | | | | | | | | | Some codec DAIs like stac9766, wm9712, wm9713, ad1980 don't register themselves then it loses to the chance to be given a null_dai_ops in snd_soc_register_dai if they have no ops. When functions like soc_pcm_open, soc_pcm_hw_params etc. access the ops field in these DAIs, panic will happen. Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: Remove snd_soc_suspend_device()Mark Brown2009-10-151-39/+0
| | | | | | | | | | | | | | | | The PM core will grow pm_link infrastructure in 2.6.33 which can be used to implement the intended functionality of the ASoC-specific device suspend and resume callbacks so drop them. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: Improve the debugfs hierarchyPeter Ujfalusi2009-10-021-2/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Change the way the debugfs entries are created: If the codec->dev is valid, than use: debugfs/asoc/{codec->name}.{dev_name(codec->dev)}/ if the codec->dev is NULL: debugfs/asoc/{codec->name}/ as root for the debugfs entries. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: add support for multiple cards/codecs in debugfsPeter Ujfalusi2009-10-011-7/+19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In order to support multiple codecs on the same system in the debugfs the directory hierarchy need to be changed by adding directory per codec under the asoc direcorty: debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg /dapm_pop_time /dapm/{widgets} With the original implementation only the debugfs files are only created for the first codec, other codecs loaded later would fail to create the debugfs files (since they are already exist). Furthermore in this situation any of the codecs has been removed, would cause the debugfs entries to disappear, regardless if the codec, which created them are still loaded (the one which loaded first). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: Provide API for reordering channelsBarry Song2009-09-131-0/+24
| | | | | | | | | | | | | | | | | | | | | | The patch adds an interface to set the relationship between audio channel number and slot number. The interface should be really useful because audio channel n doesn't always use slot n in all platforms. And for some devices, the relationship even can change with sound mode switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc. Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: Add source argument to PLL configurationMark Brown2009-09-051-3/+5
|/ | | | | | | | | | More and more devices feature PLLs and FLLs with the ability to select between multiple input clocks. In order to better support these devices a new argument, source, has been added to the set_pll() configuration API. Using set_clkdiv() is often difficult due to the need to stop the PLL/FLL before any reconfiguration can be done. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: Add DAPM widget power decision debugfs filesMark Brown2009-08-211-0/+8
| | | | | | | | | | | | | | | | | | | | Currently when built with DEBUG DAPM will dump information about the power state decisions it is taking for each widget to dmesg. This isn't an ideal way of getting the information - it requires a kernel build to turn it on and off and for large hub CODECs the volume of information is so large as to be illegible. When the output goes to the console it can also cause a noticable impact on performance simply to print it out. Improve the situation by adding a dapm directory to our debugfs tree containing a file per widget with the same information in it. This still requires a decision to build with debugfs support but is easier to navigate and much less intrusive. In addition to the previously displayed information active streams are also shown in these files. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: change set_tdm_slot api to allow slot_width override.Daniel Ribeiro2009-08-061-3/+6
| | | | | | | | | | | | | | | | Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c still doesn't handle the slot_width override. While being there, correct an incorrect use of SlotsPerFrm(7) use in bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ). (this series is meant for Mark's for-2.6.32 branch) Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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