The module_platform_driver() conversion ended up tripping over the driver
name, leading to confusion in the macro with regards to 'driver' being
redefined. rename it to something slightly more suitable to avoid
namespace collisions.
sound/sh/sh_dac_audio.c:444:122: error: conflicting types for 'driver_init'
include/linux/device.h:773:6: note: previous declaration of 'driver_init' was here
make[3]: *** [sound/sh/sh_dac_audio.o] Error 1
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
get_min_max_with_quirks() must be called after the control id name
string is determined, but the current code changes the id name string
after calling the function.
Reported-by: Christian Melki <christian.melki@ericsson.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch just sets the codec probe_mask=0x101 value for the WinFast VP200 H
PCoIP card based on Teradici hardware matching the PCI subsystem vendor/device
IDs 3a21:040d. The user reported no codec detection issues without this
explicit codec configuration.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Maintain both streams (playback, capture) synchronized. Previous code
didn't take in account the small byte count drifts caused by the irq
position rounding.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Trying to flush completed packets is pointless when the pointer
callback was called from the packet completion callback; avoid it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By flushing all completed but not yet reported packets before reading
the PCM hardware position, the granularity of the pointer is improved
from the interrupt interval to the packet size.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The following patch might introduce this call chain:
PCM .pointer callback
+ fw_iso_context_flush_completions
+ packet callback
+ snd_pcm_period_elapsed
+ PCM .pointer callback
Recursive calls to the pointer callback are not possible due to the PCM
group locking, so avoid this by moving the period notification into
a separate tasklet.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Explained by Takashi in <s5hfwbtmz0q.wl%tiwai@suse.de>
> The reason is because get_min_max*() isn't called in the place you
> created these controls, and get_min_max() would be called only for
> integer volumes later even if uninitialized. A short cut for booleans.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the PCM read/write loop, the driver calls snd_pcm_update_hw_ptr()
at each time at the beginning of the loop. Russell King reported that
this hogs CPU significantly.
The current code assumes that the pointer callback is very fast and
cheap, also not too much fine grained. It's not true in all cases.
When the pointer advances short samples while the read/write copy has
been performed, the driver updates the hw_ptr and gets avail > 0
again. Then it tries to read/write these small chunks. This repeats
until the avail really gets to zero.
For avoiding this situation, a simple workaround is to call
snd_pcm_update_hw_ptr() only once at starting the loop, assuming that
the read/write copy is performed fast enough. If the available count
becomes short, it goes to snd_pcm_wait_avail() anyway, and this
processes right.
Tested-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When pdata->reset_pin is passed with a negative value (means gpio
is invalid), then chip->reset_pin will not be assigned to a vaule,
it will use default value 0. This will cause unexpected behavior.
So, add this patch to correct.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The upper limit of the available minors isn't necessarily 128 + unit,
but it's rather up to 256. Fixing this allows more than 8 devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are left-over codes from the ancient days with the static device
number limitation of 8. Actaully OSS can support up to 16 cards.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since this is a generic API which should support any userspace interface
for reporting jacks update the documentation a little to make that a bit
clearer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the PCI ID of the Asus Xonar DGX card; it's otherwise
identical with the DG.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/usb/endpoint.c: In function 'queue_pending_output_urbs':
sound/usb/endpoint.c:298: warning: 'packet' may be used uninitialized in this function
Cc: Daniel Mack <zonque@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds controls for the effects section on the FTU devices.
Some of these controls need volume quirks. They are added to
mixer.c.
[fixed missing break by tiwai]
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is in preparation for more FTU controls to come.
Should help keeping names a bit shorter.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds db gain information to M-Audio Fast Track Ultra (8R) devices.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename mixer_vol_tlv to snd_usb_mixer_vol_tlv and export it to make
it reuseable in mixer_quirks.c.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge snd_maudio_ftu_create_ctl() and snd_ebox44_create_ctl() into
snd_create_std_mono_ctl().
As opposed to the ftu and ebox-44 specific functions, a TLV callback
can be specified for controls created by snd_create_std_mono_ctl().
[fixed minor checkpatch.pl warnings by tiwai]
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop some struct members and definitions that became obsolete during
the refactorization of the driver.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mixer units from the firmware are corrupt, and even where they
are valid they presents mono controls as L and R channels of
stereo.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some interfaces reference endpoints which do not exists. To
accomodate these, do not fail completely, but skip over them.
This allows the Electrix Ebox-44 with earlier firmware to be
detected and used for audio.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many fields have been moved to struct snd_usb_endpoint.
Also fix the proc output to correspond to the new structure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ep->fill_max is a 1 bit flag, thus it has to be boolean.
sound/usb/endpoint.c: In function 'snd_usb_endpoint_set_params':
sound/usb/endpoint.c:785: warning: overflow in implicit constant conversion
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Document the new streaming code and some of the functions so that
contributers can catch up easier.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implicit feedback is a streaming mode that does not rely on dedicated
sync endpoints but uses the information provided by record streams to
clock output streams. Now that the streaming logic is decoupled from the
PCM streams, this is easy to implement.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With the previous commit that added the new streaming model, all
endpoint and streaming related code is now in endpoint.c, and pcm.c
only acts as a wrapper for handling the packet's payload.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new generic streaming logic for audio over USB.
It defines a model (snd_usb_endpoint) that handles everything that
is related to an USB endpoint and its streaming. There are functions to
activate and deactivate an endpoint (which call usb_set_interface()),
and to start and stop its URBs. It also has function pointers to be
called when data was received or is about to be sent, and pointer to
a sync slave (another snd_usb_endpoint) that is informed when data has
been received.
A snd_usb_endpoint knows about its state and implements a refcounting,
so only the first user will actually start the URBs and only the last
one to stop it will tear them down again.
With this sort of abstraction, the actual streaming is decoupled from
the pcm handling, which makes the "implicit feedback" mechanisms easy to
implement.
In order to split changes properly, this patch only adds the new
implementation but leaves the old one around, so the the driver doesn't
change its behaviour. The switch to actually use the new code is
submitted separately.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
release_firmware() deals gracefully with NULL pointers, no need to check first.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At the point of this error-handling code, both regions and the dma have
been allocated, so free it as done in previous and subsequent
error-handling code.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At the point of this error-handling code, HAVE_DSPCODEH may be undefined,
so free INITCODE and PERMCODE as done elsewhere. A jump and label are
introduced to avoid code duplication.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A bunch of driver-specific fixes and one generic fix for the new support
for platform DAPM contexts - we were picking the wrong default for the
idle_bias_off setting which was meaning we weren't actually achieving
any useful runtime PM on platform devices.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.11 (GNU/Linux)
iQIcBAABAgAGBQJPftQoAAoJEBus8iNuMP3du4IQAIxcJs4Q6wckpnu8bcFr4+Gr
UstCaoS7R3BROe1rtZJr6WPzkVGkriYJdRvH66q/PDJ9f97CFMR4b0IwKnHXUbbL
CVR8dod1I4+0PGF7D08UfHln5uUYO17HaKB5aYPzzcBegA4DZwUtSvnvaBVpg7kK
JQM5S6G6nb68mroyfSF6XsmCM3ehwgykiqlo1TH9WQQ0Z6KbNatV86mi2W8jxvwB
xh7EafL1v9MmpDkqEVEseaI6Mr7N/DiaNHzozqW7Na2nVgddRTXzh2dXA8gsr4VW
vsAoIrACc0zCgmwyYgRN0Dg8k2hqcbERbd/prmgHCYDUQc6n6mUf/AwE+ZZMo0FK
iCOMMZbESxxWqPNnSz9+H5pgoDLITB8kTOjp1TwGu9xYc0wEkHo+DiG5ZG3pZsCv
OKfF/FiBL30OlsefT+kKFzn+3Ve9ryJ4NZ7SYg6zWGqLmBidWle3c2yZqAldSavx
kHjw4I1YAO552fGK7v5Dij/ClUps5dYdiczxqzdIgH0GE0sTfsYQrtfl8nXy5+mf
cKAljqwM9ONAZ7CbAO6To6MPy9qhCmVLGPFGT7BdXJDyK2YGLGGWQiRITdnULSZd
AhG8EitOgtgOprMFLEgkcKb6cnHE4/Ry1V8jhYg0+1D+YIuTWhsYeO+mci+zto/r
0ZnX6E5R/aA8Z56+7xgU
=EiwZ
-----END PGP SIGNATURE-----
Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: fixes for 3.4
A bunch of driver-specific fixes and one generic fix for the new support
for platform DAPM contexts - we were picking the wrong default for the
idle_bias_off setting which was meaning we weren't actually achieving
any useful runtime PM on platform devices.
name pins consistently (MIC1/LINE1/HP-OUT/CD) on all controls
affecting those pins.
remove duplicate SET_AMP_GAIN_MUTE to 0x17/index 0 and 0x17/index 1
really select MIC1, not Mixer out for recording
"Mixer out" for recording is not a "pin", adjust comment
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CX20549 (ctx5045) doesn't accept data on index 1 for output pins,
as shown in the following hda-var transaction:
$ hda-verb /dev/snd/hwC0D0 0x10 set_amp_gain 0xb126
nid = 0x10, verb = 0x300, param = 0xb126
value = 0x0
$ hda-verb /dev/snd/hwC0D0 0x10 get_amp_gain 0x8001
nid = 0x10, verb = 0xb00, param = 0x8001
value = 0x0
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ID used for detection of the BenQ R55E actually identifies the
Quanta TW3 ODM design, which is also used for the Gigabyte W551 laptop
series. Schematics on the internet clearly indicate that the "Port C"
(analog input connected to record source #4 and mixer input #4) is
unconnected.
Playing an audio CD through analog playback (using cdplay from cdtools)
produces no sound, even with the mixer input labelled "CD" enabled, and
the volume control in the CD drive set to maximum. This indicates the
connection is really not present.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "input converter" widget of the CX20459 has only one input amplifier,
expose that one as "Capture Volume/Capture Switch". The actual record
source selection is already exposed through the separately installed
input mux.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This includes renaming "Line In" to line, also in the mixer settings.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CX20549 has only one single input amp on it's input converter
widget. Fix printing of values in the codec file in /proc/asound.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit d4a2eca "ASoC: Tegra I2S: Remove dependency on pdev->id" changed
the prototype of tegra_i2s_debug_add, but didn't update the dummy inline
used when !CONFIG_DEBUG_FS. Fix that.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: <stable@vger.kernel.org> # 3.3
The ASoC core currently defaults to using STANDBY rather than OFF for
idle ASoC platform devices, which causes a permanent pm_runtime_get() on
them. This keeps the device active unnecessarily. This can be especially
problematic when the ASoC platform device and DAI device are the same
device.
The distinction between OFF and STANDBY is likely not relevant for ASoC
platform drivers, since they aren't analog devices. So, solve this issue
by hard-coding idle_bias_off = 1 for all ASoC platform devices. If this
turns out to be a problem, this value could be sourced from the
snd_soc_platform_driver, similarly to soc_probe_codec().
Note: Prior to this change, this caused a large (10) runtime_active count
for the Tegra I2S controller even when not in use, and a leak in that
value as streams were started and stopped. This change probably hides a
bug.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Check for NULL pointer before accessing it.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
