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* Merge branch 'for_linus' of ↵Linus Torvalds2008-10-13
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-2.6 * 'for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-2.6: (313 commits) V4L/DVB (9186): Added support for Prof 7300 DVB-S/S2 cards V4L/DVB (9185): S2API: Ensure we have a reasonable ROLLOFF default V4L/DVB (9184): cx24116: Change the default SNR units back to percentage by default. V4L/DVB (9183): S2API: Return error of the caller provides 0 commands. V4L/DVB (9182): S2API: Added support for DTV_HIERARCHY V4L/DVB (9181): S2API: Add support fot DTV_GUARD_INTERVAL and DTV_TRANSMISSION_MODE V4L/DVB (9180): S2API: Added support for DTV_CODE_RATE_HP/LP V4L/DVB (9179): S2API: frontend.h cleanup V4L/DVB (9178): cx24116: Add module parameter to return SNR as ESNO. V4L/DVB (9177): S2API: Change _8PSK / _16APSK to PSK_8 and APSK_16 V4L/DVB (9176): Add support for DvbWorld USB cards with STV0288 demodulator. V4L/DVB (9175): Remove NULL pointer in stb6000 driver. V4L/DVB (9174): Allow custom inittab for ST STV0288 demodulator. V4L/DVB (9173): S2API: Remove the hardcoded command limit during validation V4L/DVB (9172): S2API: Bugfix related to DVB-S / DVB-S2 tuning for the legacy API. V4L/DVB (9171): S2API: Stop an OOPS if illegal commands are dumped in S2API. V4L/DVB (9170): cx24116: Sanity checking to data input via S2API to the cx24116 demod. V4L/DVB (9169): uvcvideo: Support two new Bison Electronics webcams. V4L/DVB (9168): Add support for MSI TV@nywhere Plus remote V4L/DVB: v4l2-dev: remove duplicated #include ...
| * V4L/DVB (8777): tea575x-tuner: replace video_exclusive_open/releaseHans Verkuil2008-10-12
| | | | | | | | | | | | | | | | Move the video_exclusive_open/release functionality into the driver itself. Signed-off-by: Hans Verkuil <hverkuil@xs4all.nl> Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
* | Merge branch 'for-linus' of ↵Linus Torvalds2008-10-13
|\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: (33 commits) ALSA: ASoC codec: remove unused #include <version.h> ALSA: ASoC: update email address for Liam Girdwood ALSA: hda: corrected invalid mixer values ALSA: hda: add mixers for analog mixer on 92hd75xx codecs ALSA: ASoC: Add destination and source port for DMA on OMAP1 ALSA: ASoC: Drop device registration from GTA01 lm4857 driver ALSA: ASoC: Fix build of GTA01 audio driver ALSA: ASoC: Add widgets before setting endpoints on GTA01 ALSA: ASoC: Fix inverted input PGA mute bits in WM8903 ALSA: ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver ALSA: ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver ALSA: ASoC: OMAP: Add multilink support to McBSP DAI driver ALSA: ASoC: Make TLV320AIC26 user-visible ALSA: ASoC - clean up Kconfig for TLV320AIC2 ALSA: ASoC: Make WM8510 microphone input a DAPM mixer ALSA: ASoC: Implement WM8510 bias level control ALSA: ASoC: Remove unused AUDIO_NAME define from codec drivers ALSA: ASoC: tlv320aic3x: Use uniform tlv320aic naming ALSA: ASoC: Add WM8510 SPI support ALSA: ASoC: Add WM8753 SPI support ...
| * | ALSA: ASoC: Allow machine drivers to mark pins as not connectedMark Brown2008-10-12
| |/ | | | | | | | | | | | | | | | | | | | | | | Add a new API call snd_soc_dapm_nc_pin() which allows machine drivers to mark pins as being permanently disabled. At present this is identical to snd_soc_dapm_disable_pin() except in terms of improving the internal documentation of machine drivers that use it. The intention is that in future it will be extended to provide additional features such as hiding controls that are only relevant to paths using the disconnected pin. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'master' of ↵David S. Miller2008-10-11
|\| | | | | | | | | | | | | | | master.kernel.org:/pub/scm/linux/kernel/git/torvalds/linux-2.6 Conflicts: sound/core/memalloc.c
| * ALSA: Increase components array sizeTakashi Iwai2008-10-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Increase the card components[] (and thus snd_card_info.components[], too) array size from 80 to 128 chars so that more strings can be stored. The 80 chars aren't enough for more than 2 HD-audio codecs, and this hits an ugly snd_BUG() as reported by Wu Fegguang for HP 2230s. The control protocol number is increased to 2.0.6 as well, in case it matters. Reported-by: Wu Fengguang <wfg@linux.intel.com> Acked-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Remove bitwise from snd_pcm_hw_param_tTakashi Iwai2008-10-10
| | | | | | | | | | | | | | | | | | We have some arithmetic operations against snd_pcm_hw_param_t, thus bitwise isn't correct for it. Better to remove the flag to shut up sparse warnings. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Separate common pxa2xx-pcm codeDmitry Baryshkov2008-09-23
| | | | | | | | | | | | | | | | | | | | | | ASoC and non-ASoC drivers for PCM DMA on PXA share lots of common code. Move it to pxa2xx-lib. [Fixed some checkpatch warnings -- broonie] Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Separate common pxa2xx-ac97 codeDmitry Baryshkov2008-09-23
| | | | | | | | | | | | | | | | | | | | | | | | ASoC and non-ASoC drivers for ACLINK on PXA share lot's of common code. Move all common code into separate module snd-pxa2xx-lib. [Fixed handing of SND_AC97_CODEC in Kconfig and some checkpatch warnings -- broonie] Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Release v1.0.18rc3Jaroslav Kysela2008-09-09
| | | | | | | | Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: remove stale filesTakashi Iwai2008-09-09
| | | | | | | | | | | | | | | | Empty files remained likely due to wrong patching. Remove them now. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: ALS4000 driver work, step 2Andreas Mohr2008-08-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | - more register naming work - finally figured out that weird CR register stuff (and did I mention that I hate _really_ undecipherable open-coded values?) - fix handling of IRQ sharing in interrupt handler (hopefully properly, otherwise I'd be grateful to hear your pedantic comments ;) - add handy SPECS_PAGE references wherever useful - comments, cleanup - add me as module author Signed-off-by: Andreas Mohr <andi@lisas.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Allocate larger pages in sgbufTakashi Iwai2008-08-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Most hardwares have limited buffer-descriptor table length. This also restricts the max buffer size of the sound driver. For example, snd-hda-intel has 1MB buffer size limit, and this is because it can have at most 256 BDL entries. For supporting larger buffers, we need to allocate larger pages even for sg-buffers. This patch changes the sgbuf allocation code to try to allocate larger pages first. At each head of the allocated pages, the number of allocated pages is stored in the lowest bits of the corresponding entry of the table addr field. This change isn't visible as long as the driver uses snd_sgbuf_get_addr() helper. Also, the patch adds a new function, snd_pcm_sgbuf_get_chunk_size(). This returns the size of the chunk on continuous pages starting at the given position offset. If the chunk reaches to a non-continuous page, it returns the size to the boundary. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Clean up SG-buffer helper functions and macrosTakashi Iwai2008-08-25
| | | | | | | | | | | | | | | | Clean up SG-buffer helper functions and macros. Helpers take substream as arguments now. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: release v1.0.18rc1Jaroslav Kysela2008-08-15
| | | | | | | | Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Clean up snd_BUG()Takashi Iwai2008-08-13
| | | | | | | | | | | | | | Use the standard WARN() macro for snd_BUG(). Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Kill snd_assert() definitionTakashi Iwai2008-08-13
| | | | | | | | | | | | | | Remove snd_assert() completely now. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Kill snd_assert() in other placesTakashi Iwai2008-08-13
| | | | | | | | | | | | | | | | Kill snd_assert() in other places, either removed or replaced with if () with snd_BUG_ON(). Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Kill snd_assert() in sound/core/*Takashi Iwai2008-08-13
| | | | | | | | | | | | | | | | Kill snd_assert() in sound/core/*, either removed or replaced with if () with snd_BUG_ON(). Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Introduce snd_BUG_ON() macroTakashi Iwai2008-08-13
| | | | | | | | | | | | | | | | | | | | Introduced snd_BUG_ON() macro as a replacement of snd_assert() macro. snd_assert() is pretty ugly as it has the control flow in its argument. OTOH, snd_BUG_ON() behaves like a normal conditional, thus it's much easier to read the flow. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: wss_lib: use wss detection code instead of ad1848 oneKrzysztof Helt2008-08-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Use the wss detection code and kill the ad1848 library. The library is fully assimilated into the new wss library. This required reworking of the AD1848 family code so the code is changed to correctly detect chips from the AD1848 and CS4231 families. I have tested it on following cards: Gallant SC-6600 (codec: AD1848, driver: snd-sc6600) SoundScape VIVO/90 (codec: AD1845, driver: snd-sscape) SG Waverider (codec: CS4231A, driver: Rene Herman's snd-galaxy) Opti930 (codec: built-in - CS4231 compatible, driver: snd-opti93x) Opti931 (codec: built-in - CS4231 compatible, driver: snd-opti93x) Gallant SC-70P (chip/codec: CS4237B, driver: snd-cs4236) Audio Plus 3D (chip/codec: CMI8330A, driver: snd-cmi8330) Dell Latitude CP (chip/codec: cs4236, driver snd-cs4232) Sound playback and recording works on all these cards. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Reviewed-by: Rene Herman <rene.herman@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: wss_lib: use wss pcm code instead of ad1848 oneKrzysztof Helt2008-08-06
| | | | | | | | | | | | | | | | | | | | | | | | | | Use the wss pcm code and kill the ad1848 pcm code. The AD1848 chip is much slower than CS4231 chips so the waiting loop was increased 100x (10x is not enough). Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Reviewed-by: Rene Herman <rene.herman@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: wss_lib: use wss mixer code instead of ad1848 oneKrzysztof Helt2008-08-06
| | | | | | | | | | | | | | | | | | Use the wss mixer code and kill the ad1848 mixer code. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Reviewed-by: Rene Herman <rene.herman@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: wss_lib: use CS4231P instead of AD1848P (kill the AD1848P)Krzysztof Helt2008-08-06
| | | | | | | | | | | | | | | | | | Use CS4231P instead of AD1848P (kill the AD1848P). Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Reviewed-by: Rene Herman <rene.herman@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: wss_lib: replace ad1848 mixer element macros with wss onesKrzysztof Helt2008-08-06
| | | | | | | | | | | | | | | | | | Use the wss macros instead of ad1848 ones. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Reviewed-by: Rene Herman <rene.herman@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: wss_lib: use wss constants instead of ad1848 onesKrzysztof Helt2008-08-06
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Use wss constants for mode. Move ad1848 hardware constants to the wss.h. Move mixer tlv macros into the ad1848_lib.c from the ad1848.h. Drop the MODE_RUNNING spurious IRQ guard on AD1848 as it doesn not seem to be needed. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Reviewed-by: Rene Herman <rene.herman@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: wss_lib: use struct snd_wss instead of snd_ad1848Krzysztof Helt2008-08-06
| | | | | | | | | | | | | | | | | | | | The snd_wss is superset of the snd_ad1848 so kill the latter and replace it with the snd_wss. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Reviewed-by: Rene Herman <rene.herman@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: wss_lib: rename cs4321_foo to wss_fooKrzysztof Helt2008-08-06
| | | | | | | | | | | | | | | | | | | | | | Rename functions and structures from the former cs4321_lib to names more corresponding with the new name: wss_lib. Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Reviewed-by: Rene Herman <rene.herman@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: wss_lib: rename cs4231.h into wss.hKrzysztof Helt2008-08-06
| | | | | | | | | | | | | | | | | | | | Rename file include/sound/cs4231.h into include/sound/wss.h Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Reviewed-by: Rene Herman <rene.herman@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Fix limit of 8 PCM devices in SNDRV_CTL_IOCTL_PCM_NEXT_DEVICEPawel MOLL2008-08-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When compiled with CONFIG_SND_DYNAMIC_MINORS the ALSA core is fine to have more than 8 PCM devices per card, except one place - the SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE ioctl, which will not enumerate devices > 7. This patch fixes the issue, changing the devices list organisation. Instead of adding new device to the tail, the list is now kept always ordered (by card number, then device number). Thus, during enumeration, it is easy to discover the fact that there is no more given card's devices. Additionally the device field of struct snd_pcm had to be changed to int, as its "unsignednity" caused a lot of problems when comparing it to potentially negative signed values. (-1 is 0xffffffff or even more then ;-) Signed-off-by: Pawel Moll <pawel.moll@st.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Add USB US122L driverKarsten Wiese2008-08-01
| | | | | | | | | | | | | | | | Added a new US122L usb-audio driver. This driver works together with a dedicated alsa-lib plugin. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * sound: Revert "ALSA: Fix limit of 8 PCM devices in ↵Jaroslav Kysela2008-08-01
| | | | | | | | | | | | | | | | | | | | SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE" This reverts commit fb3d6f2b77bdec75d45aa9d4464287ed87927866. New, updated patch with same subject replaces this commit. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: IEC958 definition for consumer status channel updatePawel MOLL2008-08-01
| | | | | | | | | | | | | | | | Updated IEC958 consumer status channel definitions according to the third edition of IEC60958-3 spec. Signed-off-by: Pawel Moll <pawel.moll@st.com> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Fix limit of 8 PCM devices in SNDRV_CTL_IOCTL_PCM_NEXT_DEVICEPawel MOLL2008-07-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When compiled with CONFIG_SND_DYNAMIC_MINORS the ALSA core is fine to have more than 8 PCM devices per card, except one place - the SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE ioctl, which will not enumerate devices > 7. This patch fixes the issue, changing the devices list organisation. Instead of adding new device to the tail, the list is now kept always ordered (by card number, then device number). Thus, during enumeration, it is easy to discover the fact that there is no more given card's devices. The same limit was present in OSS emulation code. It has been fixed as well. Additionally the device field of struct snd_pcm is now int, instead of unsigned int, as there is no obvious reason for keeping it unsigned. This caused a lot of problems with comparing this value with other (almost always signed) variables. There is just one more place where device number is unsigned - in struct snd_pcm_info, which should be also sorted out in future. Signed-off-by: Pawel MOLL <pawel.moll@st.com> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: ASoC: convert use of uint to unsigned intJon Smirl2008-07-29
| | | | | | | | | | | | | | | | | | ASOC: convert use of uint to unsigned int Signed-off-by: Jon Smirl <jonsmirl@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: ASoC: Make OpenFirmware helper include file conditionalMark Brown2008-07-29
| | | | | | | | | | | | | | | | | | | | The OpenFirmware API headers don't build on all platforms so ensure that they are not included unless they are being used. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Grant Likely <grant.likely@secretlab.ca> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: ASoC: Add OpenFirmware helper for matching bus and codec driversGrant Likely2008-07-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Simple utility layer for creating ASoC machine instances based on data in the OpenFirmware device tree. OF aware platform drivers and codec drivers register themselves with this framework and the framework automatically instantiates a machine driver. At the moment, the driver is not very capable and it is expected to be extended as more features are needed for specifying the configuration in the device tree. This is most likely temporary glue code to work around limitations in the ASoC v1 framework. When v2 is merged, most of this driver will need to be reworked. Signed-off-by: Grant Likely <grant.likely@secretlab.ca> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: ASoC: Rename mask to max to reflect usageJon Smirl2008-07-29
| | | | | | | | | | | | | | | | | | | | | | | | Most of the ASoC controls refer to the maximum value that can be set for a control as mask but there is no actual requirement for all bits to be set at the highest possible value making the name mask misleading. Change the code to use max instead. Signed-off-by: Jon Smirl <jonsmirl@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: ASoC: Convert bitfields in ASoC into full int widthJon Smirl2008-07-29
| | | | | | | | | | | | | | | | | | | | | | Convert bitfields in ASoC into full int width. This is a simple mechanical conversion. Two places in the DAPM code were fixed to properly use mask. Signed-off-by: Jon Smirl <jonsmirl@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: ASoC: Allow codecs to override register displayMark Brown2008-07-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some codecs have unusual features in their register maps such as very large registers representing arrays of coefficients. Support these codecs in the register cache sysfs file by allowing them to provide a function register_display() overriding the default output for register contents. Also ensure that we don't overflow PAGE_SIZE while writing out the register dump. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
| * ALSA: Add jack reporting APIMark Brown2008-07-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently very few systems provide information about jack status to user space, even though many have hardware facilities to do detection. Those systems that do use an input device with the existing SW_HEADPHONE_INSERT switch type to do so, often independently of ALSA. This patch introduces a standard method for representing jacks to user space into ALSA. It allows drivers to register jacks for a sound card with the input subsystem, binding the input device to the card to help user space associate the input devices with their sound cards. The created input devices are named in the form "card longname jack" where jack is provided by the driver when allocating a jack. By default the parent for the input device is the sound card but this can be overridden by the card driver. The existing user space API with SW_HEADPHONE_INSERT is preserved. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* | alsa: Remove special SBUS dma support code.David S. Miller2008-08-29
|/ | | | | | No longer used. Signed-off-by: David S. Miller <davem@davemloft.net>
* ALSA: ASoC: Export dapm_reg_event() fullyMark Brown2008-07-29
| | | | | | | | | dapm_reg_event() is used by devices using SND_SOC_DAPM_REG() so needs to be exported to support building them as modules and prototyped to avoid sparse warnings and potential build issues. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: Release v1.0.17Jaroslav Kysela2008-07-14
| | | | Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: ALSA driver for SGI O2 audio boardThomas Bogendoerfer2008-07-14
| | | | | | | | | | This patch adds a new ALSA driver for the audio device found inside most of the SGI O2 workstation. The hardware uses a SGI custom chip, which feeds a AD codec chip. Signed-off-by: Thomas Bogendoerfer <tsbogend@alpha.franken.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: remove CONFIG_KMOD from soundJohannes Berg2008-07-10
| | | | | | | | | | A bunch of things in alsa depend on CONFIG_KMOD, use CONFIG_MODULES instead where the dependency is needed at all. Signed-off-by: Johannes Berg <johannes@sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: asoc: core - add Digital Audio Interface (DAI) control functions.Liam Girdwood2008-07-10
| | | | | | | | | | This patch adds several functions for DAI control and config and replaces the current method of calling function pointers within the DAI struct. Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: asoc: core - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.Liam Girdwood2008-07-10
| | | | | | | | | | | | | This patch series merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai in preparation for further ASoC v2 patches. This merger removes duplication in both DAI structures and simplifies the API for other users. Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: asoc: core - refactored DAPM pin control API.Liam Girdwood2008-07-10
| | | | | | | | | | | | | Refactored snd_soc_dapm_set_endpoint() to snd_soc_dapm_enable_pin() and snd_soc_dapm_disable_pin(). Renamed snd_soc_dapm_sync_endpoints() to snd_soc_dapm_sync(). Renamed snd_soc_dapm_get_endpoint_status() to snd_soc_dapm_get_pin_status(). Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* ALSA: Release v1.0.17rc3Jaroslav Kysela2008-06-27
| | | | Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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