In the original code the condition was always true (hopefully) because
WM8776_HPLVOL is zero.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Master Control port (MC) is available as the last
PnP resource (OPT005). Use this value instead fo guessing.
Also, add some comments to the code.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch works around misbehaviour of Creative Creative VF0470 Live Cam
which reports 16 kHz sample rate for audio capture while actually producing
8 kHz stream.
Signed-off-by: Arseniy Lartsev <arseniy@fizlesh.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some code that is no longer needed now that the relevant parts of
the driver have been tested.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
sound/usb/caiaq/midi.h:6: ERROR: "foo* bar" should be "foo *bar"
Signed-off-by: Andrea Gelmini <andrea.gelmini@gelma.net>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/oss/v_midi.h:5: ERROR: code indent should use tabs where possible
sound/oss/v_midi.h:7: ERROR: trailing whitespace
Signed-off-by: Andrea Gelmini <andrea.gelmini@gelma.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the Edirol UA-1000 to the UA-101 driver.
Both devices behave the same, so we just have to shuffle around some
interface numbers and name strings.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the definitions from linux/usb/audio.h all over the ALSA USB audio
driver and add some missing definitions there as well.
Use the endpoint attribute macros from linux/usb/ch9 and remove the own
things from sound/usb/usbaudio.h.
Now things are also nicely prefixed which makes understanding the code
easier.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is just a quick hack that needs to be removed once the new units
defined by the audio class v2.0 standard are supported.
However, it allows using these devices for now, without mixer support.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a number of parsers for audio class v2.0. In particular, the
following internals are different and now handled by the code:
* the number of streaming interfaces is now reported by an interface
association descriptor. The old approach using a proprietary
descriptor is deprecated.
* The number of channels per interface is now stored in the AS_GENERAL
descriptor (used to be part of the FORMAT_TYPE descriptor).
* The list of supported sample rates is no longer stored in a variable
length appendix of the format_type descriptor but is retrieved from
the device using a class specific GET_RANGE command.
* Supported sample formats are now reported as 32bit bitmap rather than
a fixed value. For now, this is worked around by choosing just one of
them.
* A devices needs to have at least one CLOCK_SOURCE descriptor which
denotes a clockID that is needed im the class request command.
* Many descriptors (format_type, ...) have changed their layout. Handle
this by casting the descriptors to the appropriate structs.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds some definitions for audio class v2.
Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have
different numerical representations in both standards, so there is need
for a _V1 add-on now. usbmixer.c is changed accordingly.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation of support for v2.0 audio class, use the structs from
linux/usb/audio.h and add some new ones to describe the fields that are
actually parsed by the descriptor decoders.
Also, factor out code from usb_create_streams(). This makes it easier to
adopt the new iteration logic needed for v2.0.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_cs46xx_codec_reset() bypassing the register cache, so as to not
clobber the cached register value during resume.
Signed-off-by: Florian Zumbiehl <florz@florz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes a division by zero error in the irq handler.
There is a small window between the hw_params() callback and when
runtime->frame_bits is set by ALSA middle layer. When another substream is
already running, if an interrupt is delivered during that window the irq
handler calls pcm_pointer() which does a division by zero. The patch below
makes the irq handler skip substreams that are initialized but not started
yet. Cc to Clemens Ladisch because he proposed an alternate fix.
For more information, please read the original thread in the linux-kernel
mailing list: http://lkml.org/lkml/2010/2/2/187
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usbmixer proc file contains mapping between ALSA control API and
USB mixer control units. The purpose of this file is for debugging
and a problem diagnostics.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Here's a patch that adds MIDI support through USB for one of the Access
Music synths, the VirusTI.
The synth uses standard USBMIDI protocol on its USB interface 3, although
it does signal "vendor specific" class. A magic string has to be sent on
interface 3 to enable the sending of MIDI from the synth (this string was
found by sniffing usb communication of the Windows driver). This is all
my patch does, and it works on my computer.
Please note that the synth can also do standard usb audio I/O on its
interfaces 2&3, which already works with the current snd-usb-audio driver,
except for the audio input from the synth. I'm going to work on it when I
have some time.
Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> (cosmetics, list terminator)
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Extend the list of devices whose firmware does not expect more than one
USB MIDI packet in one USB packet.
bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3752
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch adds rearranges parts of the initialization code and adds
suspend and resume callbacks.
This patch adds suspend and resume callbacks.
It also rearranges parts of the initialization code so it can be
used in both the first initialization (when the module is loaded we
also have to load default settings) and the resume callback (where
we have to restore the previous settings).
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the controls init code outside the init_hw() function because is must
not be called during resume.
This patch moves the code that initializes the card's controls with
default valued from the init_hw() function into a separated
set_mixer_defaults() function (one for each of the 16 supported
cards). This change is necessary because during resume we must
resurrect the hardware without losing the previous
settings. set_mixer_defaults() must be called only once when the
module is loaded.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements a simple cache for the firmware files when CONFIG_PM is defined.
This patch changes get_firmware(), free_firmware() and adds
free_firmware_cache(). The first two functions implement a very
simple cache and the latter is used to actually release all the stored
firmwares when the module is unloaded.
When CONFIG_PM is not enabled those functions act as before, that is
free_firmware() releases the firmware immediately and
free_firmware_cache() does nothing.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changes the way the firmware is passed through functions.
When CONFIG_PM is enabled the firmware cannot be released because the
driver will need it again to resume the card.
With this patch the firmware is passed as an index of the struct
firmware card_fw[] in place of a pointer. That same index is then used
to locate the firmware in the firmware cache.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the zero-division warning message with WARN_ON_ONCE() per the
advice by Linus. This shouldn't happen, but if it happens, it's
possible that the bug happens often due to buggy IRQs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB devices tends to represent dB ranges in different way than ALSA expects.
Add possibility to override these values and add guessed values for
SoundBlaster MP3+.
Also rename 'Capture Input Source' control to 'Capture Source' for
SoundBlaster MP3+ and Extigy.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
On my AMD780V chipset, hda_intel.c can crash the kernel with a divide by
zero
for as-yet unknown reasons. A simple check for zero prevents it, though
the problem that causes it remains. Since the workaround is harmless and
won't affect anyone except victims of this bug, it should be safe;
moreover,
because this crash can be triggered by a user-mode application, there are
denial of service implications on the systems affected by the bug without
the patch.
Signed-off-by: Jody Bruchon <jody@nctritech.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use DEFINE_PCI_DEVICE_TABLE() to make PCI device ids go to
.devinit.rodata section, so they can be discarded in some cases,
and make them const.
Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pandora's external DAC is using 256*Fs output from the TWL4030
codec, and TWL4030 needs to have APLL enabled for it's 256*Fs
output to function.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The volume register is from 0..0x7f and 0..0x1a range is mute.
Also, fix mute combining in wm_vol_put(). The wrong behaviour was
noticed by Peter Christensen.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This renames the interrupt name in /proc/interrupt.
HDA Intel -> hda_intel
This also eliminates space from the name, probably helping some
parsers.
Don't think anybody depends on this name in userspace
Signed-off-by: Takashi Iwai <tiwai@suse.de>
My sound codec seems sometimes (very rarely) to omit interrupts (ALC268)
However, interrupt mode still works.
Thus if we get timeout, poll the codec once.
If we get 3 such polls in a row, then switch to polling mode.
This patch is maybe an bandaid, but this might be a workaround for hardware bug.
Signed-off-by: Maxim Levitsky <maximlevitsky@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I found that the sampling rate locking setting of the ice1712 sound driver
was only half-respected : when the driver was locked to, let's say, 44100Hz,
and a usermode app was requesting 48000Hz playback, the request was succesful
although the soundcard would continue to run at 44100Hz.
Here's a patch that will make those requests to fail.
Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After hours of debugging, I finally found the reason why some source
and runtime combination does not work. The PTP (page table pages)
address must be aligned. I am not sure how much, but alignment to
PAGE_SIZE is sufficient. Also, use ALSA's page allocation routines
to ensure proper virtual -> physical address translation.
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This is a cleanup for the dummy driver. The model kernel module parameter
is introduced to select the soundcard emulation.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
