Fixed the irq handler in soc/at91-at91rm9200-i2s.c to follow the
new style without pt_regs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added the support for mixer matrix of RME9632 rev 152.
Signed-off-by: Remy Bruno <remy.bruno@trinnov.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Let the emu10k1 driver select FW_LOADER because the new Emu1010 support
requires it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Let the AudioScience, Echoaudio and Riptide drivers select FW_LOADER
instead of depending on it so that they can be configured without having
to enable FW_LOADER manually.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Properly quote a string that had an embedded newline.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds a Makefile and Kconfig to build the ASoC AT91RM9200
support.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds support for the Endrelia ETI_B1 machine using the WM8731
codec and the AT91RM9200 platform.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds I2S support to the Atmel AT91RM9200 CPU.
Features:-
o Playback/Capture supported.
o 16 Bit data size.
o 8k - 48k sample rates.
o ssc0, ssc1 and ssc2 supported as I2S ports.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds ASoC audio DMA support to the Atmel AT91RM9200 CPU.
Features:-
o Playback/Capture supported.
o 16 Bit data size.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds an ASoC Makefile and Kconfig for the WM8731, WM8750 and
WM9712 codecs.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch allows the std Alsa AC97 codec driver to use any AsoC AC97
controller driver. Currently, only HiFi playback and Capture are
supported atm.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds ASoC support for the WM9712 codec.
Supported features:-
o Capture/Playback/Sidetone/Bypass.
o Aux DAC.
o 8k - 48k sample rates.
o DAPM.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds ASoC support for the WM8750 codec.
Supported features:-
o Capture/Playback/Sidetone/Bypass.
o 16 & 24 bit audio.
o 8k - 96k sample rates.
o DAPM.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds ASoC support for the WM8731 codec.
Supported features:-
o Capture/Playback/Sidetone/Bypass.
o 16 & 24 bit audio.
o 8k - 96k sample rates.
o DAPM.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds support for building the ASoC core and the dynamic audio
power management support.
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds Dynamic Audio Power Management (DAPM) to ASoC.
Dynamic Audio Power Management (DAPM) is designed to allow portable and
handheld Linux devices to use the minimum amount of power within the
audio subsystem at all times. It is independent of other kernel PM and
as such, can easily co-exist with the other PM systems.
DAPM is also completely transparent to all user space applications as
all power switching is done within the ASoC core. No code changes or
recompiling are required for user space applications. DAPM makes power
switching decisions based upon any audio stream (capture/playback)
activity and audio mixer settings within the device.
DAPM spans the whole machine. It covers power control within the entire
audio subsystem, this includes internal codec power blocks and machine
level power systems.
There are 4 power domains within DAPM:-
1. Codec domain - VREF, VMID (core codec and audio power)
Usually controlled at codec probe/remove and suspend/resume, although
can be set at stream time if power is not needed for sidetone, etc.
2. Platform/Machine domain - physically connected inputs and outputs
Is platform/machine and user action specific, is configured by the
machine driver and responds to asynchronous events e.g when HP are
inserted
3. Path domain - audio subsystem signal paths
Automatically set when mixer and mux settings are changed by the user.
e.g. alsamixer, amixer.
4. Stream domain - DAC's and ADC's.
Enabled and disabled when stream playback/capture is started and stopped
respectively. e.g. aplay, arecord.
All DAPM power switching decisions are made automatically by consulting
an audio routing map of the whole machine. This map is specific to each
machine and consists of the interconnections between every audio
component (including internal codec components).
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch is the core of ASoC functionality.
The ASoC core is designed to provide the following features :-
o Codec independence. Allows reuse of codec drivers on other platforms
and machines.
o Platform driver code reuse. Reuse of platform specific audio DMA and
DAI drivers on different machines.
o Easy I2S/PCM digital audio interface configuration between codec and
SoC. Each SoC interface and codec registers their audio interface
capabilities with the core at initialisation. The capabilities are
subsequently matched and configured at run time for best power and
performance when the application hw params are known.
o Machine specific controls/operations: Allow machines to add controls
and operations to the audio subsystem. e.g. volume control for speaker
amp.
To achieve all this, ASoC splits an embedded audio system into 3
components :-
1. Codec driver: The codec driver is platform independent and contains
audio controls, audio interface capabilities, codec dapm and codec IO
functions.
2. Platform driver: The platform driver contains the audio dma engine
and audio interface drivers (e.g. I2S, AC97, PCM) for that platform.
3. Machine driver: The machine driver handles any machine specific
controls and audio events. i.e. turning on an amp at start of playback.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Use pci_iomap and ioread*/iowrite*() functions for accessing
hardwares. pci_iomap is suitable for hardwares like ICH and
compatible that have both PIO and MMIO.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch converts most uses of list_for_each to list_for_each_entry all
across alsa. In some place apparently an item can be on a list with
different pointers so of course that isn't compatible with list_for_each, I
therefore didn't touch those places.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch changes i2sbus_attach_codec to implement a proper error handling
strategy using labels to jump to the right part. Since it has an elaborate
set-up sequence it also needs that tear-down, which I had hard-coded
inbetween all the checks. This increases readability and should reduce .text
size as well.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch makes a few whitespace cleanups and makes i2sbus assign the new
struct device pointer in struct snd_pcm so that the proper device symlink
shows up in sysfs.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds a struct device pointer to struct snd_pcm in order to be
able to give it a different device than the card. It defaults to the card's
device, however, so it should behave identically for drivers not touching
the field.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds snd_register_device_for_dev taking a struct device
pointer to link the new device to and makes snd_register_device a simple
static inline wrapper around it.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Enable the analog loopback of the Revolution 5.1 card.
This patch adds support for the PT2258 volume controller and modifies
the Revolution 5.1 driver to make use of this facility. This allows
to control the analog loopback of the card.
Signed-off-by: Jochen Voss <voss@seehuhn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Enable capture from line-in and CD on the Revolution 5.1 card.
This patch adds support for switching between the 5 input channels of
the AK5365 ADC and modifies the Revolution 5.1 driver to make use of
this facility. Previously the capture channel was fixed to channel 0
(microphone on the Revolution 5.1 card).
Signed-off-by: Jochen Voss <voss@seehuhn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add pause capabilities for both USB playback and capture streams.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The hardware information structures for playback and capture streams,
respectively, are the same, so we can use just one structure for both
streams.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The recent change for a new sysfs tree with card* object breaks the
/sys/class/sound tree if CONFIG_SYSFS_DEPRECATED is enabled.
The device in each entry doesn't point the correct device object:
/sys/class/sound
...
|-- pcmC0D0c
| |-- dev
| |-- device -> ../../../class/sound/card0
| |-- pcm_class
| |-- power
| | `-- wakeup
| |-- subsystem -> ../../../class/sound
| `-- uevent
Also, this change breaks some drivers (like sound/arm/*) referring
card->dev directly to obtain the device object for memory handling.
This patch reverts the semantics of card->dev to the former version,
which points to a real device object. The card* object is stored in a
new card->card_dev field, instead. The device parent is chosen either
card->dev or card->card_dev according to CONFIG_SYSFS_DEPRECATED to
keep the tree compatibility.
Also, card* isn't created if CONFIG_SYSFS_DEPRECATED is enabled. The
reason of card* object is a root of all beloing devices, and it makes
little sense if each sound device points to the real device object
directly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Monty Montgomery <xiphmont@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
The previous patch 'Repair snd-usb-usx2y for usb 2.6.18' assumed
urb->start_frame roll over beyond MAX_INT for both UHCI & OHCI.
This isn't true until now (kernel 2.6.20).
Fix this by only looking at the common between OHCI & UHCI Frame number
range.
This is for mainline and stable kernels >= 2.6.18.
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fixed the error from kobject_add() at reconnection the usb audio device.
This happens when an app keeps opening a device while the device is
replugged, due to the confliction of the internal bookkept index and
the really empty slot.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Without the patch below namelist[0] will not be freed in case
of kmalloc error.
Signed-off-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Playing with spdif output on cmipci i've noticed the SPDO5V option does
not change appropriate bits the register.
The _snd_cmipci_uswitch_put checks the change in flags in wrong way.
If 'active' state of an option corresponds to a _zero_ bits in a hw
register then function fails. The SPDO5V is the sample.
In the most cases 'active' state of option is set through an non-zerio
bits in a register. This case works fine.
The fix attached.
Unfortunately i was unable to change spdif output voltage anyway.
Although the register changes right at least.
From: Timofei V. Bondarenko <tim@ipi.ac.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds the Intel ICH9 HD Audio controller DID's for ALSA.
Signed-off-by: Jason Gaston <jason.d.gaston@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The C-Media CM6501 chip's descriptors say that altsetting 5 supports
48 kHz, but it actually plays at 96 kHz.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix races between the timer handler and the close function.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add the support for HD audio controllers of MCP51,MCP55,MCP61,MCP65 & MCP67.
Signed-off-by: Peer Chen <pchen@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
SBUS: Change IRQ-handler return value from 0 to IRQ_HANDLED and
fix some initialisation problems.
Change period_bytes_min from 4096 to 256 to allow driver to work with
low latency (VOIP) applications. Hope this does not break EBUS.
Signed-off-by: Georg Chini <georg.chini@triaton-webhosting.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch fixes a couple of bit update functions in
alsa-kernel/pci/ac97/ac97_codec.c, which could possibly corrupt bits not
in the given mask.
Specifically, it'll clobber unset bits in the target that are not in the
mask, when the corresponding bit in the given new value is set.
Signed-off-by: James C Georgas <jgeorgas@rogers.com>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
If snd_pcm_new_stream() fails to initalize a substream (if
snd_pcm_substream_proc_init() returns error), snd_pcm_new_stream()
immediately return without unlinking that kfree()d substram.
It causes oops when snd_pcm_free() iterates the list of substream to
free them by invalid reference.
Signed-off-by: Akinobu Mita <akinobu.mita@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai