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According documentation bit ACLKRPOL is set to 0 (receiver samples data
on falling edge) and when set to 1 (receiver samples data on rising edge).
I2S data are always sampled on falling edge and valid during rising edge
of bit clock. So in case of capture data transmitter sample data on falling
edge and macsp must read then on rising edge.
Signed-off-by: Marek Belisko <marek.belisko@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The missing break here means that we always return early and the
function is a no-op.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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CBS_CFS format
When McASP is bit clock and frame clock master enable pin output for rx clocks.
Signed-off-by: Marek Belisko <marek.belisko@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Update dapm_clock_event to use clk_prepare_enable and
clk_disable_unprepare.
Signed-off-by: Fabio Baltieri <fabio.baltieri@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This is my example conversion of a few existing mmap users. The pcm
mmap case is one of the more straightforward ones.
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.9
A few updates, more than I'd like, fixing some relatively small issues
but mostly driver specific ones. Nothing wildly exciting so if it
doesn't make v3.9 it won't be the end of the world but it'd be nice.
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Reported-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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The Charge Pump needs the DSP clock to work properly, without it the
bypass to HP/LINEOUT is not working properly. This requirement is not
mentioned in the datasheet but has been confirmed by Mark Brown from
Wolfson.
Signed-off-by: Alban Bedel <alban.bedel@avionic-design.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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The usb_control_msg() function expects __u16 types and performs
the endianness conversions by itself.
However, in three places, a conversion is performed before it is
handed over to usb_control_msg(), which leads to a double conversion
(= no conversion):
* snd_usb_nativeinstruments_boot_quirk()
* snd_nativeinstruments_control_get()
* snd_nativeinstruments_control_put()
Caught by sparse:
sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:512:38: expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:512:38: got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types)
sound/usb/mixer_quirks.c:543:35: expected unsigned short [unsigned] [usertype] value
sound/usb/mixer_quirks.c:543:35: got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:543:56: expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:543:56: got restricted __le16 [usertype] <noident>
sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types)
sound/usb/quirks.c:502:35: expected unsigned short [unsigned] [usertype] value
sound/usb/quirks.c:502:35: got restricted __le16 [usertype] <noident>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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changed is not initialized in path_power_down_sync, but it is expected
to be false in case no change happened in the loop. So set it to
false.
Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This reverts commit 6ab317419c62850a71e2adfd1573e5ee87d8774f.
The commit [6ab317419c: ALSA: hda - Allow power_save_controller option
override DCAPS] changed the behavior of power_save_controller so that
it can override the driver capability. This assumed that this option
is rarely changed dynamically unlike power_save option. Too naive.
It turned out that the user-space power-management tool tries to set
power_save_controller option to 1 together with power_save option
without knowing what's actually doing. This enabled forcibly the
runtime PM of the controller, which is known to be broken om many
chips thus disabled as default.
So, the only sane fix is to revert this commit again. It was intended
to ease debugging/testing for runtime PM enablement, but obviously we
need another way for it.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=56171
Reported-and-tested-by: Nikita Tsukanov <keks9n@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Rename "Digitial In" to "Digital In". This function is only used for
proc output, so should not cause any problems to change.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* Added the device ID to the modalias list and assinged ALC662 patches
for it
* Added 4 port support for the device ID 0671 in alc662_parse_auto_config
Signed-off-by: Rainer Koenig <Rainer.Koenig@ts.fujitsu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The tegra dmaengine driver does not support pausing and resuming a DMA stream.
The tegra PCM driver still claims to support pause and resume though and
implements them by stopping and restarting the stream. This is not what an
application using pause/resume would expect. Usually applications have support
for working around PCMs which do not support suspend and resume, so don't set
the SNDRV_PCM_INFO_PAUSE and SNDRV_PCM_INFO_RESUME flags for the tegra PCM and
use the default snd_dmaengine_pcm_trigger callback.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Currently, a new platform device is created for secondary device
by calling platform_device_register_resndata and then the drvdata
is set for this device.
The following patch has been added to driver core:
"driver core: fix possible missing of device probe".
This results in the added device getting probed immediately but
the drvdata for the secondary device is not yet set.
This patch removes the platform_device_register_resndata call and
instead calls platform_device_alloc, platform_set_drvdata and
platform_device_add which fixes the above issue.
Signed-off-by: Prathyush K <prathyush.k@samsung.com>
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch fixes a possible crash in case drvdata for the secondary
device is not set.
Signed-off-by: Prathyush K <prathyush.k@samsung.com>
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The C-Media CM6631 USB receiver doesn't respond to changes in sample rate
while the interface is active. The same behavior is observed in other UAC2
hardware like the VIA VT1731.
Reset the interface after setting the sampling frequency on sample rate
changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is
used. Otherwise, the device will try to use the sample rate of the previous
stream, causing distorted sound on sample rate changes.
The reset is performed for all UAC2 devices, as it should not affect a
standards compliant device, but it is only necessary for C-Media CM6631,
VIA VT1731 and possibly others.
Failure to read sample rate from the device is not handled as an error in
set_sample_rate_v2(), as (permanent or intermittent) failure to read sample
rate isn't essential for a successful sample rate set.
Signed-off-by: Torstein Hegge <hegge@resisty.net>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch let ELD debug message show 'pin_eld->monitor_present' which reflects
the real pin response to verb GET_PIN_SENSE.
'eld->monitor_present' should not be used here because 'eld' is a temp
structure now and so its "monitor_present" is not set.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In function snd_hdmi_get_eld(), the variable 'ret' should be initialized to 0.
Otherwise it will be returned uninitialized as non-zero after ELD info is got
successfully. Thus hdmi_present_sense() will always assume ELD info is invalid
by mistake, and /proc file system cannot show the proper ELD info.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Cc: stable@vger.kernel.org
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When a new stream is being opened it is necessary to cancel any delayed
power down of the audio.
[Fixed unused variable -- broonie]
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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It can be 0 or 1 return value of snd_soc_update_bits_locked() when it is
success. So just check return value is negative.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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The dma-sh7760 currently fails with the following compile error:
sound/soc/sh/dma-sh7760.c:346:2: error: unknown field 'pcm_ops' specified in initializer
sound/soc/sh/dma-sh7760.c:346:2: warning: initialization from incompatible pointer type
sound/soc/sh/dma-sh7760.c:347:2: error: unknown field 'pcm_new' specified in initializer
sound/soc/sh/dma-sh7760.c:347:2: warning: initialization makes integer from pointer without a cast
sound/soc/sh/dma-sh7760.c:348:2: error: unknown field 'pcm_free' specified in initializer
sound/soc/sh/dma-sh7760.c:348:2: warning: initialization from incompatible pointer type
sound/soc/sh/dma-sh7760.c: In function 'sh7760_soc_platform_probe':
sound/soc/sh/dma-sh7760.c:353:2: warning: passing argument 2 of 'snd_soc_register_platform' from incompatible pointer type
include/sound/soc.h:368:5: note: expected 'struct snd_soc_platform_driver *' but argument is of type 'struct snd_soc_platform *'
This is due the misnaming of the snd_soc_platform_driver type name and 'ops'
field. The issue was introduced in commit f0fba2a("ASoC: multi-component - ASoC
Multi-Component Support").
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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The generic parser should evaluate the availability of the independent
HP when specified. Otherwise a DAC without the direct connection to
the corresponding pin may be assigned for the HP, but the driver
doesn't check it at all. The problem was actually seen on some
machines with VT1708s or equivalent codec, where DAC0 is assigned to
HP although it can be connected only via aamix.
This patch adds the badness evaluation for the independent HP to make
it working properly.
Reported-by: Lydia Wang <LydiaWang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The current DSP loader code abuses snd_hda_lock_devices() for ensuring
the DSP loader not conflicting with the other normal operations. But
this trick obviously doesn't work for the PM resume since the streams
are kept opened there where snd_hda_lock_devices() returns -EBUSY.
That means we need another lock mechanism instead of abuse.
This patch provides the new lock state to azx_dev. Theoretically it's
possible that the DSP loader conflicts with the stream that has been
already assigned for another PCM. If it's running, the DSP loader
should simply fail. If not -- it's the case for PM resume --, we
should assign this stream temporarily to the DSP loader, and take it
back to the PCM after finishing DSP loading. If the PCM is operated
during the DSP loading, it should get an error, too.
Reported-and-tested-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There is a typo in convert_to_spdif_status() about checking the
emphasis IEC958 status bit. It should check the given value instead
of the resultant value.
Reported-by: Martin Weishart <martin.weishart@telosalliance.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The objects allocated by devm_* APIs are managed by devres and are freed when
the device is detached. Hence there is no need to use kfree() explicitly.
Signed-off-by: Silviu-Mihai Popescu <silviupopescu1990@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Commit 552d1ef6 ("ASoC: core - Optimise and refactor pcm_new() to pass only
rtd") updated the pcm_new() callback to take the rtd as the only parameter. The
spear PCM driver (which was merged much later) still uses the old API. This
patch updates the driver to the new API.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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Source files shouldn't have the executable bit set.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Creation of individual mixer controls may fail, but that shouldn't cause
the entire mixer creation to fail. Even worse, if the mixer creation
fails, that will error out the entire device probing.
All the functions called by parse_audio_unit() should return -EINVAL if
they find descriptors that are unsupported or believed to be malformed,
so we can safely handle this error code as a non-fatal condition in
snd_usb_mixer_controls().
That fixes a long standing bug which is commonly worked around by
adding quirks which make the driver ignore entire interfaces. Some of
them might now be unnecessary.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Rodolfo Thomazelli <pe.soberbo@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In check_input_term() and parse_audio_feature_unit(), propagate the
error value that has been returned by a failing function instead of
-EINVAL. That helps cleaning up the error pathes in the mixer.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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UAC2_EXTENSION_UNIT_V2 differs from UAC1_EXTENSION_UNIT, but can be handled in
the same way when parsing the unit. Otherwise parse_audio_unit() fails when it
sees an extension unit on a UAC2 device.
UAC2_EXTENSION_UNIT_V2 is outside the range allocated by UAC1.
Signed-off-by: Torstein Hegge <hegge@resisty.net>
Acked-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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I forgot to update spec->gpio_data in the automute hook, so it will be
overridden at the init sequence, thus the machine is still silent when
no headphone jack is plugged at boot time.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The argument passed to snd_hda_attach_beep_device() is a widget NID
while spec->beep_amp holds the composed value for amp controls.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This leaks the beep input device after module unload, which leads to
Oops.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=55321
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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During the transition to the generic parser, the hook to the codec
specific automute function was forgotten. This resulted in the silent
output on some MacBooks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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