The missing break here means that we always return early and the
function is a no-op.
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org

This is my example conversion of a few existing mmap users.  The pcm
mmap case is one of the more straightforward ones.
Acked-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>

Reported-by: NRyo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org

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: NAlban Bedel <alban.bedel@avionic-design.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org

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: NEldad Zack <eldad@fogrefinery.com>
Acked-by: NDaniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NJiri Slaby <jslaby@suse.cz>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This reverts commit 6ab31741.

The commit [6ab31741: 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=56171Reported-and-tested-by: NNikita Tsukanov <keks9n@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

* 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: NRainer Koenig <Rainer.Koenig@ts.fujitsu.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NLars-Peter Clausen <lars@metafoo.de>
Reviewed-by: NStephen Warren <swarren@nvidia.com>
Tested-by: NStephen Warren <swarren@nvidia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NPrathyush K <prathyush.k@samsung.com>
Signed-off-by: NPadmavathi Venna <padma.v@samsung.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch fixes a possible crash in case drvdata for the secondary
device is not set.
Signed-off-by: NPrathyush K <prathyush.k@samsung.com>
Signed-off-by: NPadmavathi Venna <padma.v@samsung.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NTorstein Hegge <hegge@resisty.net>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NMengdong Lin <mengdong.lin@intel.com>
Acked-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NMengdong Lin <mengdong.lin@intel.com>
Cc: stable@vger.kernel.org
Acked-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NJoonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org

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 f0fba2ad("ASoC: multi-component - ASoC
Multi-Component Support").
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org

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: NLydia Wang <LydiaWang@viatech.com.cn>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NMartin Weishart <martin.weishart@telosalliance.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NSilviu-Mihai Popescu <silviupopescu1990@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NLars-Peter Clausen <lars@metafoo.de>
Acked-by: NRajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org

Source files shouldn't have the executable bit set.
Signed-off-by: NJoe Perches <joe@perches.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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: NDaniel Mack <zonque@gmail.com>
Reported-and-tested-by: NRodolfo Thomazelli <pe.soberbo@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NTorstein Hegge <hegge@resisty.net>
Acked-by: NDaniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NTakashi Iwai <tiwai@suse.de>

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: NTakashi Iwai <tiwai@suse.de>

This leaks the beep input device after module unload, which leads to
Oops.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=55321Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NTakashi Iwai <tiwai@suse.de>

"chn" here is a number between 0 and 255, but ->chn_info[] only has
16 elements so there is a potential write beyond the end of the
array.

If the seq_mode isn't SEQ_2 then we let the individual drivers
(either opl3.c or midi_synth.c) handle it.  Those functions all
do a bounds check on "chn" so I haven't changed anything here.
The opl3.c driver has up to 18 channels and not 16.
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

spec->dsp_state is initialized to DSP_DOWNLOAD_INIT, no need to reset
and check it in ca0132_download_dsp().
Signed-off-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Instead of using the dspload_is_loaded() function, check the dsp_state
that is kept in the spec.  The dspload_is_loaded() function returns
true if the DSP transfer was never started.  This false-positive leads
to multiple second delays when ca0132_setup_efaults() times out on
each write.
Signed-off-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

If dspload_image() fails, it was ignored and dspload_wait_loaded() was
still called.  dsp_loaded should never be set to true in this case,
skip it.  The check in dspload_wait_loaded() return true if the DSP is
loaded or if it never started.
Signed-off-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

*path is not yet initialized when we check if the widget is connected.

The compiler also warns about this:
sound/soc/soc-dapm.c: In function 'is_connected_output_ep':
sound/soc/soc-dapm.c:824:18: warning: 'path' may be used uninitialized in this function
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

If there are no internal speakers, we should not turn the eapd switch
off, because it might be necessary to keep high for Headphone.

BugLink: https://bugs.launchpad.net/bugs/1155016Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Remove probe function from the init section.
Signed-off-by: NMarkus Pargmann <mpa@pengutronix.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NSascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: NMarkus Pargmann <mpa@pengutronix.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org

'data' is malloced in snd_soc_bytes_put() and should be freed
before leaving from the error handling cases, otherwise it will cause
memory leak.
Signed-off-by: NWei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>