Commit graph

627 commits

Author SHA1 Message Date
Linus Torvalds
4ef58d4e2a Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (42 commits)
  tree-wide: fix misspelling of "definition" in comments
  reiserfs: fix misspelling of "journaled"
  doc: Fix a typo in slub.txt.
  inotify: remove superfluous return code check
  hdlc: spelling fix in find_pvc() comment
  doc: fix regulator docs cut-and-pasteism
  mtd: Fix comment in Kconfig
  doc: Fix IRQ chip docs
  tree-wide: fix assorted typos all over the place
  drivers/ata/libata-sff.c: comment spelling fixes
  fix typos/grammos in Documentation/edac.txt
  sysctl: add missing comments
  fs/debugfs/inode.c: fix comment typos
  sgivwfb: Make use of ARRAY_SIZE.
  sky2: fix sky2_link_down copy/paste comment error
  tree-wide: fix typos "couter" -> "counter"
  tree-wide: fix typos "offest" -> "offset"
  fix kerneldoc for set_irq_msi()
  spidev: fix double "of of" in comment
  comment typo fix: sybsystem -> subsystem
  ...
2009-12-09 19:43:33 -08:00
Takashi Iwai
86e1d57e4f Merge branch 'topic/hda' into for-linus 2009-12-04 16:22:45 +01:00
Takashi Iwai
baf9226667 Merge branch 'topic/asoc' into for-linus 2009-12-04 16:22:41 +01:00
Takashi Iwai
57648cd52b Merge branch 'topic/misc' into for-linus 2009-12-04 16:22:37 +01:00
André Goddard Rosa
af901ca181 tree-wide: fix assorted typos all over the place
That is "success", "unknown", "through", "performance", "[re|un]mapping"
, "access", "default", "reasonable", "[con]currently", "temperature"
, "channel", "[un]used", "application", "example","hierarchy", "therefore"
, "[over|under]flow", "contiguous", "threshold", "enough" and others.

Signed-off-by: André Goddard Rosa <andre.goddard@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2009-12-04 15:39:55 +01:00
Jean Delvare
83cf0a9b86 comment typo fix: sybsystem -> subsystem
Signed-off-by: Jean Delvare <jdelvare@suse.de>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2009-12-04 15:39:49 +01:00
Takashi Iwai
75639e7ee1 Merge branch 'topic/beep-rename' into topic/core-change 2009-12-01 15:58:10 +01:00
Takashi Iwai
980f31c46b Merge branch 'topic/ice1724-quartet' into topic/hda 2009-12-01 15:57:01 +01:00
Mark Brown
c0fa59df72 ASoC: Add BCLK calculation utility for TDM mode too
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-11-25 19:55:46 +00:00
Krzysztof Helt
9dc9120c77 ALSA: opti-miro: expose ACI mixer to outside drivers
The ACI mixer is used to control the radio FM module
installed on the Miro PCM20 sound card. Expose ACI mixer
outside the sound card driver.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-23 09:41:55 +01:00
Krzysztof Helt
9aeba62971 ALSA: opti-miro: make miro.h header available outside the alsa directory
Move the miro.h header to the include/sound directory. It can
be used in the Miro PCM20 radio driver (v4l).

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-23 09:41:46 +01:00
Krzysztof Helt
b753e03e5e ALSA: cs4236: update control names
Update control names to be more closer to their meaning.
Change the "Mono" name to the "Beep" as this line is usually
used to forward the PC beeper signal to sound card's output.
Update names for both cs423x and wss.

Clean up cs4235 controls according to the cs4235 doc. Rename
some of the cs4235 controls to be consistent with the cs4236's
ones.

Also, delete one misnamed cs4231 register define.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-18 07:23:16 +01:00
Joonyoung Shim
c871a05315 ASoC: Add jack_status_check callback function for GPIO jacks
The jack_status_check callback function is the interface to check the
status of the jack. Some target provides the method to distinguish what
is the jack inserted - headphone jack, microphone jack, tvout jack, etc,
so we can implement it using the jack_status_check function.

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-11-12 16:45:53 +00:00
Mark Brown
7aae816dae ASoC: Add bit clock rate calculator utility functions
Many devices need to calculate the bit clock rate desired to
work out the clock configuration required for the device.
Provide utility functions to do this using both hw_params
structures and raw numbers.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2009-11-12 16:45:48 +00:00
Clemens Ladisch
7584af10cf sound: rawmidi: record a substream's owner process
Record the pid of the task that opened a RawMIDI substream.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-10 16:32:38 +01:00
Clemens Ladisch
e7373b702f sound: pcm: record a substream's owner process
Record the pid of the task that opened a PCM substream. For sound
cards with hardware mixing, this allows determining which process
is associated with a specific substream's volume control.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-10 16:32:20 +01:00
Clemens Ladisch
25d27eded1 control: use reference-counted pid
Instead of storing the PID number, take a reference to the task's pid
structure.  This protects against duplicates due to PID overflows, and
using pid_vnr() ensures that the PID returned by snd_ctl_elem_info() is
correct as seen from the current namespace.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-11-06 14:32:06 +01:00
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