Commit Graph

660 Commits

Author SHA1 Message Date
Clemens Ladisch
31cef7076e control: remove snd_konctrol_volatile::owner_pid field
We do not need to save the ID of the process that locked a control
because that information is already available in the owner's file data.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-06 14:32:03 +01:00
Krzysztof Helt
d114cd84a1 ALSA: cs4236: detect chip in one pass
The cs4236 was two step detection with call to the snd_wss_free()
between two steps. The snd_wss_free() did not free a sound device
created in the snd_wss_create(). This caused an OOPS during module
removal as the same sound device was released twice. The same OOPS
happened if the cs4236 module loading failed.

Fix this by adapting the snd_cs4236_create() to correctly work with
chips less capable then cs4236. The snd_cs4236_create() behaves the
same as the snd_wss_create() if the chip is less capable than the cs4236.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-05 18:10:25 +01:00
Rafael Ignacio Zurita
9dcaa7b25f ALSA: sh: add SuperH DAC audio driver for ALSA V4
This is a port of the sound/oss/sh_dac_audio.c driver.
The driver uses an on-chip 8-bit D/A converter, which has a speaker connected
to one of its channels, found in several ancient HP machines.
For interrupts it uses a high-resolution timer (hrtimer).
Tested on SH7709 based hp6xx (HP Jornada 680/690 and HP Palmtop 620lx/660lx).

Also, since OSS Emulation works, the old OSS sound/oss/sh_dac_audio.c driver
would be obsolete soon, and it could be removed.

Signed-off-by: Rafael Ignacio Zurita <rizurita@yahoo.com>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-04 09:17:40 +01:00
Mark Brown
fe3e78e073 ASoC: Factor out snd_soc_init_card()
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>
2009-11-03 22:14:43 +00:00
Peter Ujfalusi
c8bf93f0fe ASoC: Codec driver for Texas Instruments tlv320dac33 codec
Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo
audio DAC.

TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low
power audio playback.

The digital interface can use I2S, DSP (A or B), Right and Left
justified formats.
DAC33 has stereo analog input, which can be bypassed to the analog
outputs.

Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass'
mode (default) and nSample mode (FIFO is in use).
a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is
working synchronously as a normal codec (it needs constant stream of
data on the digital interface).

b) The nSample mode implementation uses one interrupt line from DAC33 to
the host:
Alarm threshold is set to 10ms of audio data (limit by the driver
implementation).
DAC33 will signal an interrupt, when the FIFO level goes under the
Alarm threshold.
The host will write to nSample register a value (number of stereo
samples), to tell DAC33 how many samples it should read in a burst from
the host. When the DAC33 received the number of samples, it disables the
clocks on the I2S bus. When the FIFO use again goes under the Alarm
threshold, DAC33 signals the host with an interrupt, and the process is
repeated.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-15 15:02:04 +01:00
Mark Brown
d2058b0cd0 ASoC: Remove snd_soc_suspend_device()
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>
2009-10-15 15:01:43 +01:00
Peter Ujfalusi
493b67efff ASoC: TPA6130A2 amplifier driver
Driver for Texas Instruments TPA6130A2 stereo headphone
amplifier.

The driver provides playback gain control and also pre-defined
DAPM_HP widgets and DAPM routings for power management.

The DAPM_HP widget names are:
"TPA6130A2 Headphone Left"
"TPA6130A2 Headphone Right"

From soc machine drivers to use with the tpa6130a2 amplifier,
the tpa6130a2_add_controls has to be called, which adds the alsa
controls and the DAPM routing needed for the tpa6130a2.
After that the machine driver can connect the codec's output
with 'TPA6130A2 Left' and 'TPA6130A2 Right':

        {"TPA6130A2 Left", NULL, "CODEC LEFT OUT"},
        {"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"},

Internally the left and right channels are powered separately.
When none of the channels are needed the amplifier is powered
down:
hard power: valid GPIO number is passed within platform data
soft power: Using the software shutdown of the amplifier

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-09 18:50:37 +01:00
Mark Brown
907bc6c7fc Merge branch 'for-2.6.32' into for-2.6.33 2009-10-06 16:01:27 +01:00
Mark Brown
d2b247a8be ASoC: Add virtual enumeration support for DAPM muxes
Sometimes it is desirable to have a mux which does not reflect any
direct register configuration but which will instead only have an
effect implicitly (for example, as a result of changing which parts
of the device are powered up). Provide a virtual mux for this purpose.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-06 15:57:02 +01:00
Takashi Iwai
7c824f4b69 ALSA: sscape - Remove sscap_ioctl.h from include/sound/Kbuild
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-02 07:22:58 +02:00
Peter Ujfalusi
88439ac793 ASoC: add support for multiple cards/codecs in debugfs
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>
2009-10-01 12:13:04 +01:00
Krzysztof Helt
acd4710091 ALSA: sscape: convert to firmware loader framework
The conversion solves the problem that firmware size was set to 64KB
while non PnP cards have 128KB firmware files.

An additional firmware initialization code has been moved from the OSS
driver.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-01 07:51:56 +02:00
Lopez Cruz, Misael
be2500b835 ASoC: Add PDM DAI format definition
Add DAI format definition for PDM interfaces.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-28 14:43:27 +01:00
Pavel Hofman
42cfa276ae ALSA: ak4113 support
* complete support for ak4113
* based on code for ak4114 and ak4117

Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-21 15:45:07 +02:00
Pavel Hofman
8f34692f63 ALSA: ak4620 support, codec regs listed in proc
* complete support for ak4620
* codec regs listed in proc for all codecs/chips
* adding total regs for each codec
* fixing nb. of steps in input attenuation controls

Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-21 15:44:51 +02:00
Pavel Hofman
c0a9eedf9a ALSA: ak4114 - fix errors in output selector bits
* the previous version had a typo - values of AK4114_OPS10-12 were
  identical with AK4114_OPS00-02
* Since no cards actually use this feature, the bug was not identified earlier

Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-21 15:44:39 +02:00
Mark Brown
9f072b7b22 Merge branch 'for-2.6.32' into for-2.6.33 2009-09-18 15:09:44 +01:00
Barry Song
472df3cbae ASoC: Provide API for reordering channels
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>
2009-09-13 12:37:53 +01:00
Takashi Iwai
1110afbe72 Merge branch 'topic/ymfpci' into for-linus
* topic/ymfpci:
  sound: ymfpci: increase timer resolution to 96 kHz
2009-09-10 15:33:09 +02:00
Takashi Iwai
b34c866394 Merge branch 'topic/tlv-minmax' into for-linus
* topic/tlv-minmax:
  ALSA: usb-audio - Correct bogus volume dB information
  ALSA: usb-audio - Use the new TLV_DB_MINMAX type
  ALSA: Add new TLV types for dBwith min/max
2009-09-10 15:33:06 +02:00
Takashi Iwai
9d416811f8 Merge branch 'topic/snd-printk' into for-linus
* topic/snd-printk:
  ALSA: Fixed a typo of printk()
  ALSA: Add debug module option
  ALSA: core - strip too long file names in snd_print*()
2009-09-10 15:33:03 +02:00
Takashi Iwai
2c0d19a78d Merge branch 'topic/pcm-drain-nonblock' into for-linus
* topic/pcm-drain-nonblock:
  ALSA: pcm - Increase protocol version
  ALSA: pcm - Fix drain behavior in non-blocking mode
2009-09-10 15:33:00 +02:00
Takashi Iwai
9cd9f42767 Merge branch 'topic/misc' into for-linus
* topic/misc:
  ALSA: Remove unneeded ifdef from sound/core.h
  ALSA: Remove struct snd_monitor_file from public sound/core.h
  ALSA: Release v1.0.21
2009-09-10 15:32:57 +02:00
Takashi Iwai
6a0f402146 Merge branch 'topic/dummy' into for-linus
* topic/dummy:
  ALSA: dummy - Increase MAX_PCM_SUBSTREAMS to 128
  ALSA: dummy - Add debug proc file
  ALSA: Add const prefix to proc helper functions
  ALSA: Re-export snd_pcm_format_name() function
  ALSA: dummy - Fake buffer allocations
  ALSA: dummy - Fix the timer calculation in systimer mode
  ALSA: dummy - Add more description
  ALSA: dummy - Better jiffies handling
  ALSA: dummy - Support high-res timer mode
2009-09-10 15:32:51 +02:00
Takashi Iwai
f9892a52e2 Merge branch 'topic/dma-sgbuf' into for-linus
* topic/dma-sgbuf:
  ALSA: Fix SG-buffer DMA with non-coherent architectures
2009-09-10 15:32:50 +02:00
Mark Brown
215edda3ad ASoC: Allow per-route connectedness checks for supplies
Some chips with complex internal supply (particularly clocking)
arragements may have multiple options for some of the supply
connections. Since these don't affect user-visible audio routing
the expectation would be that they would be managed automatically
by one of the drivers.

Support these users by allowing routes to have a connected function
which is queried before the connectedness of the path is checked as
normal. Currently this is only done for supplies, other widgets
could be supported but are not currently since the expectation for
them is that audio routing will be under the control of userspace.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-08 19:24:56 +01:00
Takashi Iwai
4f7454a997 ALSA: Add const prefix to proc helper functions
Add appropriate const prefix to char * arguments in proc helper functions.
Also fixed the caller side to be proper const pointers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-08 14:45:06 +02:00
Takashi Iwai
6e5265ec34 ALSA: Re-export snd_pcm_format_name() function
Re-export snd_pcm_format_name() function to be used outside the PCM core.
As a first example, usbaudio is changed to use it now again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-08 14:26:51 +02:00
Takashi Iwai
b8c60ede6a ALSA: Remove unneeded ifdef from sound/core.h
Remove the old hack that was needed for building alsa-driver modules
externally for old kernels.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-07 15:58:30 +02:00
Takashi Iwai
82a783f4bc ALSA: Remove struct snd_monitor_file from public sound/core.h
The struct snd_monitor_file is used locally only in sound/core/init.c,
thus it should be moved there from the public sound/core.h.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-07 15:50:18 +02:00
Mark Brown
236cc52856 ASoC: Remove unuused hw_read_t
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-07 12:46:42 +01:00
Mark Brown
85488037bb ASoC: Add source argument to PLL configuration
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>
2009-09-05 18:52:16 +01:00
Jaroslav Kysela
9d32e03d01 ALSA: Release v1.0.21
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-03 12:03:48 +02:00
Takashi Iwai
cf0baf16c3 ALSA: Fixed a typo of printk()
Fixed a silly typo of printk() included in the previous patch...

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-28 07:22:05 +02:00
Takashi Iwai
5a53a7640a ALSA: pcm - Increase protocol version
Increase the PCM protocol version to indicate the drain ioctl behavior
change.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-27 21:04:12 +02:00
Takashi Iwai
36ce99c1dc ALSA: Add debug module option
Add debug module option to snd core.
This controls the debug print level.  When CONFIG_SND_DEBUG_VERBOSE
is set, you can suppress the debug messages by giving or changing this
parameter to a lower value.  debug=0 means no debug messsages.
As default, it's set to the verbose level 2.

Since this option can be changed dynamically via sysfs file, you can
suppress the verbose debug messages on the fly, which wasn't possible
before.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-27 17:42:08 +02:00
Mark Brown
e4aa8dd5ca Merge branch 'topic/digital-mixing' into for-2.6.32 2009-08-24 20:44:41 +01:00
Mark Brown
79fb9387f8 ASoC: Add DAPM widget power decision debugfs files
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>
2009-08-21 17:17:59 +01:00
Kuninori Morimoto
a4d7d550a9 ASoC: Add SuperH FSI driver support for ALSA
This driver is very simple.
It support playback only now.
This patch is tested by ms7724se board.

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-20 20:01:42 +01:00
Mark Brown
010ff26226 ASoC: Add input and output AIF widgets
Currently DAPM interfaces with the audio streams to and from the
processor at the DAC and ADC widgets. As the digital capabilities
of parts increases this is becoming a less and less able to meet
the needs of parts.

To meet the needs of these devices create new widgets interfacing
with the TDM bus but not integrated into any other functionality.
Audio can then be routed to and from these widgets using existing
routing widgets.

A slot number is provided in the definition but this is currently
not used yet. This is intended to support devices which can use
more than one TDM slot on a single interface.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 16:06:08 +01:00
Marek Vasut
4ac0478f2a ALSA: Allow passing platform_data for pxa2xx-ac97
This patch adds support for passing platform data to ac97 bus devices
from PXA2xx-AC97 driver..

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:37 +01:00
Mark Brown
1921bab217 Merge commit 'a5479e389e989acfeca9c32eeb0083d086202280' into for-2.6.32 2009-08-11 13:09:27 +01:00
Clemens Ladisch
6e2efaacb3 sound: ymfpci: increase timer resolution to 96 kHz
Allow the interval timer to be programmed with its full 96 kHz
precision.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:14:46 +02:00
Mark Brown
8f738d5842 ASoC: Define more formats for the AC97 CODECs
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-09 20:08:31 +01:00
Mark Brown
06cddefc1f Merge branch 'reg-cache' into for-2.6.32 2009-08-07 11:43:58 +01:00
Daniel Ribeiro
a5479e389e ASoC: change set_tdm_slot api to allow slot_width override.
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>
2009-08-06 15:52:24 +01:00
Mark Brown
afa2f1066e ASoC: Factor out I2C 8 bit address 16 bit data I/O
As part of this refactoring the type of the CODEC hw_read operation
is changed to match the regular read operation.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-03 16:59:15 +01:00
Mark Brown
7084a42b96 ASoC: Add I/O control bus information to factored out cache setup
While writes tend to be able to use a fairly bus independant format to
do the writes reads are all bus specific. To allow us to factor out
this code include the bus type as a parameter when setting up the
cache.

Initially just use this to factor out hw_write_t for I2C.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-03 16:59:09 +01:00
Mark Brown
77ee09c67e ASoC: Allow CODECs to flag invalid registers
This helps CODECs with sparse register maps work better with the
register cache display interface.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-31 18:54:48 +01:00
Marek Vasut
474828a40f ALSA: Allow passing platform_data to devices attached to AC97 bus
This patch allows passing platform_data to devices attached to AC97 bus
(like touchscreens, battery measurement chips ...).

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-23 11:30:56 +01:00
Joonyoung Shim
3ce91d5a5a ASoC: add SOC_DOUBLE_R_EXT_TLV control type
This is a macro for double controls with special callback function and
TLV. The SOC_DOUBLE_R_EXT_TLV needs two registers and one shift for
double controls.

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-15 16:59:06 +01:00
Joonyoung Shim
d0af93db12 ASoC: add SOC_DOUBLE_EXT_TLV control type
This is a macro for double controls with special callback function and
TLV. The SOC_DOUBLE_EXT_TLV needs one register and two shifts for double
controls.

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-15 16:59:06 +01:00
Peter Meerwald
47db8e89ac ASoC: fixes multiple typos in comments, no functional change
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-13 23:05:11 +01:00
Mark Brown
942c435ba7 ASoC: Add WM8993 CODEC driver
The WM8993 is a highly integrated ultra-low power hi-fi CODEC designed
for portable devices such as multimedia phones.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-08 17:20:20 +01:00
Takashi Iwai
cc6a8acdee ALSA: Fix SG-buffer DMA with non-coherent architectures
Using SG-buffers with dma_alloc_coherent() is often very inefficient
on non-coherent architectures because a tracking record could be
allocated in addition for each dma_alloc_coherent() call.
Instead, simply disable SG-buffers but just allocate normal continuous
buffers on non-supported (currently all but x86) architectures.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-08 14:20:20 +02:00
Mark Brown
17a52fd60a ASoC: Begin to factor out register cache I/O functions
A lot of CODECs share the same register data formats and therefore
replicate the code to manage access to and caching of the register
map. In order to reduce code duplication centralised versions of
this code will be introduced with drivers able to configure the use
of the common code by calling the new snd_soc_codec_set_cache_io()
API call during startup.

As an initial user the 7 bit address/9 bit data format used by many
Wolfson devices is supported for write only CODECs and the drivers
with straightforward register cache implementations are converted to
use it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 17:24:50 +01:00
Mark Brown
096e49d5e6 ASoC: Add CODEC volatile register operation
Add a volatile_register() operation to the CODEC structure providing a
standard operation to query if a register is volatile. This will be used
to factor out the register cache I/O operations for the CODECs.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 15:12:22 +01:00
Takashi Iwai
62b1653e29 Merge branch 'for-2.6.32' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2009-06-25 15:28:39 +02:00
Mark Brown
517374704d ASoC: Add a shutdown callback
Ensure that the audio subsystem is powered down cleanly when the system
shuts down by providing a shutdown operation. This ensures that all the
components have been returned to an off state cleanly which should avoid
audio issues from partially charged capacitors or noise on digital inputs
if the system is restarted quickly.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Ben Dooks <ben-linux@fluff.org>
2009-06-23 23:48:53 +01:00
Takashi Iwai
085f306541 ALSA: Add new TLV types for dBwith min/max
Add new types for TLV dB scale specified with min/max values instead
of min/step since the resolution can't match always with the one
a device provides.  For example, usb audio devices give 1/256 dB
resolution while ALSA TLV is based on 1/100 dB resolution.
The new min/max types have less problems because the possible
rounding error happens only at min/max.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-17 10:56:53 +02:00
Philipp Zabel
1abd918499 ASoC: UDA1380: refactor device registration
This patch mostly follows commit 5998102b90
"ASoC: Refactor WM8731 device registration" to make UDA1380 use standard
device instantiation. Similarly, the I2C device registration temporarily
moves into the magician machine driver before it will find its final
resting place in the board file.

At the same time, platform specific configuration is moved to platform data
and common power/reset GPIO handling moves into the codec driver.

Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-15 21:54:48 +01:00
Mark Brown
831dc0f10f ASoC: Add stub suspend and resume calls for ASoC subdevices
Now that ASoC subdevices can be regular devices they can have normal
suspend and resume calls from their buses.  However, suspending them
individually is not desirable since this can lead to problems such as
pops and clicks from devices being suspended with their signals being
amplified or clocks being stopped suddenly.

This will be resolved by having the normal device model suspend and
resume calls call into ASoC which will suspend the entire card while any
of its components are suspended.  At present this is not yet implemented
but in order to aid the transition of drivers to the standard device
model this patch adds API calls for the notifications.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-13 20:06:28 +01:00
Mark Brown
0e09b67e58 Merge branch 'dapm' into for-2.6.32 2009-06-11 21:04:04 +01:00
Takashi Iwai
3b88bc5229 Merge branch 'topic/pcm-jiffies-check' into for-linus
* topic/pcm-jiffies-check:
  ALSA: pcm - A helper function to compose PCM stream name for debug prints
  ALSA: pcm - Fix update of runtime->hw_ptr_interrupt
  ALSA: pcm - Fix a typo in hw_ptr update check
  ALSA: PCM midlevel: lower jiffies check margin using runtime->delay value
  ALSA: PCM midlevel: Do not update hw_ptr_jiffies when hw_ptr is not changed
  ALSA: PCM midlevel: introduce mask for xrun_debug() macro
  ALSA: PCM midlevel: improve fifo_size handling
2009-06-10 07:26:41 +02:00
Takashi Iwai
eabaf0634a Merge branch 'topic/pcm-delay' into for-linus
* topic/pcm-delay:
  ALSA: usbaudio - Add delay account
  ALSA: Add extra delay count in PCM
2009-06-10 07:26:40 +02:00
Takashi Iwai
19b1a15a3d Merge branch 'topic/div64-cleanup' into for-linus
* topic/div64-cleanup:
  ALSA: Clean up 64bit division functions
2009-06-10 07:26:28 +02:00
Takashi Iwai
d108728ea2 Merge branch 'topic/cleanup' into for-linus
* topic/cleanup:
  ALSA: Remove deprecated include/sound/driver.h
  ALSA: Remove deprecated snd_card_new()
2009-06-10 07:26:24 +02:00
Takashi Iwai
ab2f06cb6b Merge branch 'topic/caiaq' into for-linus
* topic/caiaq:
  ALSA: snd_usb_caiaq: bump version number
  ALSA: snd_usb_caiaq: give better shortname
  ALSA: Core - add snd_card_set_id() function
  ALSA: snd_usb_caiaq: give better longname
  ALSA: snd_usb_caiaq: use strlcpy
  ALSA: snd_usb_caiaq: clean whitespaces
2009-06-10 07:26:23 +02:00
Takashi Iwai
ba252af8d6 Merge branch 'topic/asoc' into for-linus
* topic/asoc: (135 commits)
  ASoC: Apostrophe patrol
  ASoC: codec tlv320aic23 fix bogus divide by 0 message
  ASoC: fix NULL pointer dereference in soc_suspend()
  ASoC: Fix build error in twl4030.c
  ASoC: SSM2602: assign last substream to the master when shutting down
  ASoC: Blackfin: document how anomaly 05000250 is handled
  ASoC: Blackfin: set the transfer size according the ac97_frame size
  ASoC: SSM2602: remove unsupported sample rates
  ASoC: TWL4030: Check the interface format for 4 channel mode
  ASoC: TWL4030: Use reg_cache in twl4030_init_chip
  ASoC: Initialise dev for the dummy S/PDIF DAI
  ASoC: Add dummy S/PDIF codec support
  ASoC: correct print specifiers for unsigneds
  ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()
  ASoC: Switch FSL SSI DAI over to symmetric_rates
  ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved
  ASoC: Fabric bindings for STAC9766 on the Efika
  ASoC: Support for AC97 on Phytec pmc030 base board.
  ASoC: AC97 driver for mpc5200
  ASoC: Main rewite of the mpc5200 audio DMA code
  ...
2009-06-10 07:26:18 +02:00
Mark Brown
291f3bbcac ASoC: Make DAPM power sequence lists local variables
They are now only accessed within dapm_power_widgets() so can be local
to that function.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-08 13:52:06 +01:00
Daniel Ribeiro
46f5822f78 ASoC: Allow 32 bit registers for DAPM
Replace the remaining unsigned shorts with unsigned ints.
Tested with pcap2 codec (25 bits registers).

Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-08 10:53:12 +01:00
Takashi Iwai
3f7440a6b7 ALSA: Clean up 64bit division functions
Replace the house-made div64_32() with the standard div_u64*() functions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05 17:45:17 +02:00
Jaroslav Kysela
10a8ebbb08 ALSA: Core - add snd_card_set_id() function
Introduce snd_card_set_id() function to allow lowlevel drivers to set
default identification name for card slot. The function checks also
for identification name collisions and tries to create unique name.

Also, the snd_card_create() function is simplified, because this new
function is used. As bonus, proper name collision checks are evaluated
at the card create time.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02 12:47:46 +02:00
Jaroslav Kysela
8bea869c5e ALSA: PCM midlevel: improve fifo_size handling
Move the fifo_size assignment to hw->ioctl callback to allow lowlevel
drivers overwrite the default behaviour.

fifo_size is in frames not bytes as specified in asound.h and alsa-lib's
documentation, but most hardware have fixed byte based FIFOs. Introduce
internal SNDRV_PCM_INFO_FIFO_IN_FRAMES.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-29 11:47:33 +02:00
Takashi Iwai
e93721a702 Merge branch 'fix/pcm-jiffies-check' into topic/pcm-jiffies-check 2009-05-29 11:46:10 +02:00
Mark Brown
86ed3669f0 ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier driver
The WM9081 is designed to provide high power output at low distortion
levels in space-constrained portable applications.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-22 15:11:22 +01:00
Mark Brown
5c82f56736 AsoC: Make snd_soc_read() and snd_soc_write() functions
Should be no impact on the generated code but it helps the compiler
print clearer messages.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-22 10:22:38 +01:00
Mark Brown
452c5eaa0d ASoC: Integrate bias management with DAPM power management
Rather than managing the bias level of the system based on if there is
an active audio stream manage it based on there being an active DAPM
widget. This simplifies the code a little, moving the power handling
into one place, and improves audio performance for bypass paths when no
playbacks or captures are active.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 15:53:16 +01:00
Mark Brown
6d3ddc81f5 ASoC: Split DAPM power checks from sequencing of power changes
DAPM has always applied any changes to the power state of widgets as soon
as it has determined that they are required. Instead of doing this store
all the changes that are required on lists of widgets to power up and
down, then iterate over those lists and apply the changes. This changes
the sequence in which changes are implemented, doing all power downs
before power ups and always using the up/down sequences (previously they
were only used when changes were due to DAC/ADC power events). The error
handling is also changed so that we continue attempting to power widgets
if some changes fail.

The main benefit of this is to allow future changes to do optimisations
over the whole power sequence and to reduce the number of walks of the
widget graph required to check the power status of widgets.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-18 15:53:14 +01:00
Jon Smirl
d34c430782 ASoC: Add SNDRV_PCM_FMTBIT_S32_BE as a valid AC97 format
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-14 12:47:33 +01:00
Jaroslav Kysela
35edb4003c ALSA: Release v1.0.20
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-06 12:32:26 +02:00
Takashi Iwai
4bbe1ddf89 ALSA: Add extra delay count in PCM
Added runtime->delay field to adjust the delayed samples for snd_pcm_delay().
Typically a hardware FIFO length is stored in this field, so that the
extra delay between hwptr and applptr can be computed.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-05-05 14:47:21 +02:00
Mark Brown
bbd993077d ASoC: Remove redundant codec pointer from DAIs
The DAI structure has two pointers to the codec, one in the body of the
DAI and one in a union for a parent pointer.  Drop the parent pointer
version.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-05 10:27:38 +01:00
Mark Brown
f3831a592f Merge commit 'takashi/topic/asoc' into for-2.6.31 2009-05-05 10:12:55 +01:00
Takashi Iwai
8560b9321f Merge branch 'fix/asoc' into topic/asoc 2009-05-04 16:05:23 +02:00
Mark Brown
4072604b9d ASoC: Remove unused DAI format defines
The defines for TDM and synchronous clocks are not used - they are
mostly a legacy of the automatic clocking configuration.  TDM will
require configuration of the number of timeslots and which ones to use
so can't be fit into the DAI format and synchronous mode is handled by
symmetric_rates (and needs to be done by constraints rather than when
the DAI format is being configured).

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-02 12:32:10 +01:00
Mark Brown
33f503c96c ASoC: Use a shared define for AC97 CODEC data formats
The AC97 wire format is completely fixed so CODECs don't have any choice
about the formats they accept but controllers accept a variety of data
formats and render them down onto the bus.  Have a shared define so all
the CODEC drivers will interoperate with any of our controller drivers.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-05-02 12:32:09 +01:00
Daniel Mack
7629ad24f2 ASoC: add SOC_DOUBLE_EXT macro
Add a macro for double controls with special callback functions.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-24 17:39:31 +01:00
Mark Brown
246d0a17f5 ASoC: Add power supply widget to DAPM
Many modern CODECs have shared resources on chip which must be enabled
for portions of the chip to work but which can be disabled at other times
in order to achieve power savings. Examples of such resources include
power supplies and some internal clocks.

Since these widgets are dependencies for the audio path but do not carry
audio signals they require slightly different handling to most widgets -
they do not contribute to the audio path and so should not be counted as
either inputs or outputs during path walks.

Cases where one supply provides a supply for another will require
additional work. There is also room for more optimisation of the graph
walking to avoid repeated checks for the same thing.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-22 19:10:13 +01:00
Takashi Iwai
ef9dfa4b10 ALSA: Remove deprecated include/sound/driver.h
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-04-21 08:53:41 +02:00
Takashi Iwai
cd474f2d54 ALSA: Remove deprecated snd_card_new()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-04-21 08:53:08 +02:00
Mark Brown
b75576d76d ASoC: Make the DAPM power check an operation on the widget
Rather than having switch statements at point of use make the DAPM
power check a member of the widget structure and set it when we
instantiate the widget.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-20 18:09:48 +01:00
Russell King
64bd43a086 Merge branch 'fix' of git://git.kernel.org/pub/scm/linux/kernel/git/ycmiao/pxa-linux-2.6 2009-04-20 14:03:04 +01:00
Takashi Iwai
2e8e59f437 Merge branch 'topic/hda' into for-linus
* topic/hda:
  ALSA: hda - Add quirk mask for Fujitsu Amilo laptops with ALC883
  ALSA: hda - Avoid call of snd_jack_report at release
  ALSA: add private_data to struct snd_jack
2009-04-15 11:24:09 +02:00
Mark Brown
eae17754ab [ARM] pxa: merge AC97 platform data structures
Currently there are two possible platform datas for the PXA AC97 driver:
one supported by the generic AC97 driver only which provides callbacks
to allow board-specific configuration at stream startup and teardown,
and another for pxa2xx-ac97-lib which allows configuration of the reset
GPIO for PXA2xx CPUs.

Obviously this won't actually work when using the generic AC97 driver
since the drivers will attempt to parse the platform data in both
formats. Fix this by merging the two structures.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Robert Jarzmik <robert.jarzmik@free.fr>
Cc: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
2009-04-15 10:54:06 +08:00
Takashi Iwai
9d59065cd6 ALSA: add private_data to struct snd_jack
Added private_data and private_free fields to struct snd_jack so that
the caller can assign the data.  It'll be helpful for avoiding the
double-free of the jack instance.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-04-14 16:15:09 +02:00
Mark Brown
6967963d6d Merge branch 'for-2.6.30' into for-2.6.31 2009-04-14 13:22:37 +01:00
Mark Brown
f6d655a6e6 ASoC: Support DAPM events for DACs and ADCs
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-13 11:59:01 +01:00
Jaroslav Kysela
bbf6ad1399 [ALSA] pcm-midlevel: Add more strict buffer position checks based on jiffies
Some drivers like Intel8x0 or Intel HDA are broken for some hardware variants.
This patch adds more strict buffer position checks based on jiffies when
internal hw_ptr is updated. Enable xrun_debug to see mangling of wrong
positions.

As a side effect, the hw_ptr interrupt update routine might do slightly better
job when many interrupts are lost.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2009-04-10 12:28:58 +02:00
Mark Brown
06f409d76f ASoC: Provide core support for symmetric sample rates
Many devices require symmetric configurations of capture and playback
data formats, often due to shared clocking but sometimes also due to
other shared playback and record configuration in the device. Start
providing core support for this by allowing the DAIs or the machine
to specify that the sample rates used should be kept symmetric.

A flag symmetric_rates is provided in the snd_soc_dai and
snd_soc_dai_link structures. If this is set in either of the DAIs or in
the machine then a constraint will be applied when a stream is already
open preventing any changes in sample rate.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-07 18:51:22 +01:00