fixes this error:
sound/firewire/fcp.c: In function 'fcp_avc_transaction':
sound/firewire/fcp.c:103: error: implicit declaration of function 'msleep'
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add an AMDTP stream error state that occurs when we fail to queue
another packet. In this case, the stream is stopped, and the error can
be reported when the application tries to restart the PCM stream.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For correct cache coherency on some architectures, DMA buffers must be
allocated in a different cache line than data that is concurrently used
by the CPU.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In non-blocking mode, the SYT_INTERVAL is larger than the number of
audio frames in each packet, so there are packets that do not contain
any frame to which the SYT could be applied. For these packets, the
SYT must not be the timestamp of the next valid SYT frame, but the
special no-info SYT value.
This fixes broken playback on the FireWave at 44.1 kHz.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a driver for two playback-only FireWire devices based on the OXFW970
chip.
v2: better AMDTP API abstraction; fix fw_unit leak; small fixes
v3: cache the iPCR value
v4: FireWave constraints; fix fw_device reference counting;
fix PCR caching; small changes and fixes
v5: volume/mute support; fix crashing due to pcm stop races
v6: fix build; one-channel volume for LaCie
v7: use signed values to make volume (range checks) work; fix function
block IDs for volume/mute; always use channel 0 for LaCie volume
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Acked-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Tested-by: Jay Fenlason <fenlason@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MONO was renamed to MONO1.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
This patch adds ASoC support for the MAX9850 codec with headphone
amplifier.
Supported features:
- Playback
- 16, 20 and 24 bit audio
- 8k - 48k sample rates
- DAPM
Signed-off-by: Christian Glindkamp <christian.glindkamp@taskit.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added a new API function snd_ctl_activate_id() for activate / inactivate
the control element dynamically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without this fix the driver won't instantiate properly on relevant
devices.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
Without this fix the driver won't instantiate properly on relevant
devices.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
Enable 192kHz sample rate for EP93xx.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Improve EP93xx I2S clocks management.
Some freqs values are set not exact as they requested for MCLK and
original code was not able to find divisors for SCLK and LRCLK.
This code just picks up nearest value from 3 possible variants.
This patch makes 44100 and 192000 rates working and fixes
capture function (by selecting SCLK/LRCLK=64 where possible).
All other rates should work as before.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Manage I2S rates according to datasheet for CS4271 CODEC in EDB93xx
machine driver.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Manage mode and rate bits correctly, according to datasheet in CS4271 CODEC.
This is done to make capture work properly.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We're not only prefixing all controls, we're also prefixing the widget
names in the runtime data. This causes us to add the prefix twice - once
when using the widget name to generate the control name and once when
adding the control.
Really we shouldn't be prefixing the widget names at all, the matching
code should be handing this as we always know which DAPM context a
widget came from and always display the widget name in terms of a DAPM
context. However, we're quite close to the merge window and that's
relatively invasive.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Now we've got multi-component we need to make sure that the DAPM context
(and hence register I/O context) we use to apply the pending updates at
the end of a DAPM sequence is the one we were processing rather than the
one that was used to initate the state change.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Now we have a register write minimisation code in DAPM we don't need to
worry about the ordering of the enable and disable of the PGA and the
output stage.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
McBSP sidetone is needed in telephony applications. McBSP sidetone is a
configurable FIR filter that forms a loopback from McBSP input to output.
This patch enables the McBSP2 sidetone ALSA controls so that it can be used
on Nokia RX-51/N900.
Sidetone feature can be tested with following commands:
(set up codec input and output paths)
# Enable and configure sidetone
amixer -D hw:0 set 'McBSP2 Sidetone' on
amixer set -D hw:0 'McBSP2 Sidetone Channel 0' 32767
echo 32767 >/sys/devices/platform/omap-mcbsp.2/st_taps
# Do not loop audio via CPU
arecord -f dat >/dev/null |aplay /dev/zero
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The "ldo" variable was dereferenced after free on the error path.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently will ignore prefixes when creating DAPM controls. Since currently
all control creation goes through snd_soc_cnew() we can fix this by factoring
the prefixing into that function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Symmetric rate configuration can fail if the second stream starting tries
to apply the symmetric constraint before the first stream has got far
enough to pick a rate. Rather than try to enforce a nonsensical rate of
0Hz log a warning and allow the application to carry on. Things might go
wrong later on but the user will know about it and there's unlikely to be
lasting damage.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
When multi component systems use DAIless amplifiers which require clocking
configuration it is at best hard to use the current clocking API as this
requires a DAI even though the device may not even have one. Address this
by adding set_sysclk() and set_pll() operations and APIs for CODECs.
In order to avoid issues with devices which could be used either with or
without DAIs make the DAI variants call through to their CODEC counterparts
if there is no DAI specific operation. Converting over entirely would create
problems for multi-DAI devices which offer per-DAI clocking setup.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Annoying as the __devinitdata is actually correct.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow a slight simplification of CODEC drivers by allowing DAPM routes and
widgets to be provided in a table. They will be instantiated at the end of
CODEC probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Remove warnings in ep93xx-i2s.c
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Ryan Mallon <ryan@bluewatersys.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Extend range of supported sample rates for CS4271 CODEC.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a codec driver for the dfbmcs320 bluetooth module, which is used
on the neo1973 boards.
The patch also modifies the neo1937_wm8753 sound board driver to use the new
driver instead of registering the bluetooth DAI manually.
Previously there was a name mismatch between the bluetooth DAI and the bluetooth
DAI link and the sound card was not instantiated, with this patch the issue is
no longer present and sound support works again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The neo1973(GTA01) and neo1973_gta02(GTA02) have a very similar audio hardware
setup. They both use the same codec with the same routing to the gsm modem and
bluetooth chip. But they do use different AMPs though and there are some minor
differences in the speaker setup.
As a result most of the code of those two drivers is identical.
So from a maintenance point of view it makes sense to merge them into a single
driver. It also reduces the size of kernel images supporting both the GTA01 and
GTA02.
As a side-effect of this merge the GTA01 for example gains support for routing
audio to and from the bluetooth DAI.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Using gpio_request_array instead of requesting and setting up each gpio by hand
makes the code more readable and more compact.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch drops the lm4853_{set,get}_state functions and the "Amp State Switch"
control.
Those were noops which existed to maintain alsa state file compatibility. Since
the control names have changed due to internal changes in the ASoC core and
state file compatibility was broken anyway it makes sense to drop them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch moves the code for the lm4857 AMP from the neo1973_wm8753 sound
board driver to its own module.
The lm4857 is a generic AMP IC and not specific to the neo1973.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It has been proven to be inflexible to do scenario management in kernel space.
Since actual neo1973 board support has not been merged in mainline and this
patch has been in the neo1973 tree for some time now it should be safe to remove
this functionality without breaking existing userspace.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Visstrim_M10 boards have an external tlcv320aic3205 codec
attached to SSI1. This driver glues together both interfaces.
External amplifier is not supported in this first version.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The device table is required to load modules based on modaliases.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Don't set to D3 in Cirrus errata init verbs
ALSA: hda - add new Fermi 5xx codec IDs to snd-hda
ASoC: WM8994: Ensure late enable events are processed for the ADCs
ASoC: WM8994: Don't disable the AIF[1|2]CLK_ENA unconditionaly
ASoC: Fix WM9081 platform data initialisation
ALSA: hda - Fix unable to record issue on ASUS N82JV
ALSA: HDA: Realtek: Fixup jack detection to input subsystem
