None of the platforms are actually using the SoC device so remove it
(only atmel actually has a suspend method).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is in preparation for the removal of struct snd_soc_device.
The pop time configuration should really be a property of the card not
the codec but since DAPM currently uses the codec rather than the card
using the codec is fine for now.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 does not use the struct snd_soc_device at runtime, using struct
snd_soc_card as the root of the card. Begin removing data from
snd_soc_device by pushing the workqueue data into snd_soc_card, using a
backpointer to the snd_soc_device to keep things going for the time
being.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC card initialisation is completed by a function called
snd_soc_register_card(). As part of the work to allow independant
registration of cards, codecs and machines in ASoC v2 a new function of
the same name has been added so rename the existing function to
facilitate the merge of v2.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.
This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 factors most of the contents of soc.h out into separate headers,
including soc-dai.h for the DAI. Factor the existing DAI API out into
this file in order to prepare for backporting of the ASoC v2 DAI API.
Also backport some of Liam's improvements to the documentation.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.
Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/pci/pcxhr/pcxhr_core.c: In function 'pcxhr_set_pipe_cmd_params':
sound/pci/pcxhr/pcxhr_core.c:700: warning: statement with no effect
sound/pci/pcxhr/pcxhr_core.c:706: warning: statement with no effect
sound/pci/pcxhr/pcxhr_core.c:710: warning: statement with no effect
Due to
try to fix this, and be more conventional about the empty stubs.
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rather than try to remember to keep the core version number updated
(which hasn't been happening) just remove it. It was much more useful
when ASoC was out of tree.
Signed-off-by: Mark brown <broonie@opensource.wolfsonmicro.com>
i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg
will set register 0x06 to a value of 0x59.
Also, pop_time debugfs interface setup is moved so that it
is setup in the same function as codec_reg
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When ASoC was converted to support full int width masks SOC_SINGLE_VALUE()
omitted the assignment of rshift, causing the control operatins to report
some mono controls as stereo. This happened to work some of the time due
to a confusion between shift and min in snd_soc_info_volsw().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This was marked as deprecated in 2.6.27 and all users except for
playpaq_wm8510 fixed in that release.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* '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
...
* '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
...
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>
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>
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>
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>
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>
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>
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>
- 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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>