snd_soc_register_dai() uses fmt_single_name(), and
snd_soc_register_dais() uses fmt_multiple_name()
for dai->name which is used for name based matching.
This patch uses properly snd_soc_register_dai() it it was single driver,
and uses snd_register_dais() if it were multiple drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Compress core added metadata apis in 9727b4, so add same in ASoC
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch series covers both ASoC and extcon subsystems and fixes an
interaction between the HPDET function and the headphone outputs - we
really shouldn't run HPDET while the headphone is active. The first
patch is a refactoring to make the extcon side easier.
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Merge tag 'arizona-extcon-asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/misc into asoc-arizona
ASoC/extcon: arizona: Fix interaction between HPDET and headphone outputs
This patch series covers both ASoC and extcon subsystems and fixes an
interaction between the HPDET function and the headphone outputs - we
really shouldn't run HPDET while the headphone is active. The first
patch is a refactoring to make the extcon side easier.
Running HPDET while the headphone outputs are enabled can disrupt the
operation of HPDET. In order to avoid this HPDET needs to disable the
headphone outputs and ASoC needs to not enable them while HPDET is
running.
Do the ASoC side of this by storing the enable state in the core driver
structure and only writing to the device if a flag indicating that the
accessory detection side is in a state where it can have the headphone
output stage enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Used PTR_RET function instead of IS_ERR and PTR_ERR.
Patch found using coccinelle.
Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.com>
Signed-off-by: Mark 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: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark 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 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
Use the newly introduced mute_stream DAI operation, and don't mute the
codec if it's called for the _CAPTURE stream.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Alexander Sverdlin <subaparts@yandex.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since ASoC does not yet really have the framework features needed to
support propagating sample rates through the device well yet implement
basic support for the ISRCs equivalent to that we currently have for the
ASRCs. The user can opt for 8kHz or 16kHz as the rate for the DSP blocks
in addition to the main audio rate, these being the primary use cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that the device state does not diverge from the state we have set
in the register map in order to make the behaviour clearer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some system designs have been identified which repurpose portions of the
speaker driver circuits for other functions which will require that they
not be managed using DAPM. Prepare for this by factoring out the creation
of the speaker widgets into the core driver, the widgets will be replaced
by dummy ones when the additional functions are enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
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>
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>
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>
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
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>
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>
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>
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>
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>
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>
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>
"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: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi 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: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi 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: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi 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: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi 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: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
By using the regmap API directly, we can make use of the
.write_flag_mask for SPI, which allows us to drop the strange register
hacks that were necessary so far.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Alexander Sverdlin <alexander.sverdlin@gmx.de>
Signed-off-by: Mark 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/1155016
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Older .bin files report the global coefficients as absolute address writes
to zero; maintain compatibility with them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current ASoC has register function for platform/codec/dai/card,
but doesn't have for cpu.
It often produces confusion and fault on ASoC.
As result of ASoC community discussion,
we consider new struct snd_soc_component for CPU/CODEC,
and will switch over to use it.
This patch adds very basic struct snd_soc_component,
and register function for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As pointed of by Vaibhav, commit 2952b27e2 ("ASoC: davinci-mcasp:
Add support for multichannel playback") duplicated the logic of
counting the active serializers. That can be avoided by shifting
the code around a bit.
Also, drop two unused defines introduced by the same commit.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove probe function from the init section.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
