The DSP in the CA0132 codec adds a variable latency to audio depending
on what processing is being done. Add a new patch op to return that
latency for capture and playback streams. The latency is determined
by which blocks are enabled and knowing how much latency is added by
each block.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new codec PCM ops, get_delay(), to obtain the codec/stream-
specific PCM delay count. When it's NULL, nothing changes.
This new feature was requested for CA0132, which has significant
delays in the path depending on the running DSP code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some clocks might be read-only, e.g., external clocks (see also
UAC2 4.7.2.1).
In this case, setting the sample frequency will always fail
(even if the rate is equal to the current clock rate),
therefore do not write, but read the value and compare to the
requested rate.
If the clock is read only, avoid reading it twice.
If it doesn't match, return -ENXIO since the clock is invalid for
this configuration.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Show the error code returned from the USB subsystem in
the debug messages.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a module param to disable auto clock selection.
This is provided for users that expect the audio stream to
fail when the clock source is invalid (e.g., the word clock
was unintentionally disconnected).
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If a selector is available on a device, it may be pointing to a
clock source which is currently invalid.
If there is a valid clock source which can be selected, switch
to it.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the check that parse_audio_format_rates_v2() do after
receiving the clock source entity ID directly into the find
function and add a validation flag to the function.
This patch does not introduce any logic flow change.
It is provided to allow introducing automatic clock switching
easier later. By moving this uac_clock_source_is_valid callsite,
2 additional callsites can be avoided.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the endianness conversions with the kernel-wide swabbing macros
in get/set_sample_rate_v2.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct spelling of snd_usb_endpoint_implict_feedback_sink in all
occurances.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Minor style fix, following a general code style in the kernel.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just for cleaning up, introduce a new function get_sample_rate_v2()
for replacing two identical calls in set_sample_rate_v2().
No functional change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
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>
Turing on the headphone amp interferes with the impedance measurement
used to detect a TRRS style headset microphone. Delay the HP turn on
until 500ms after the jack is detected, allowing the mic detection
state machine to run to completion.
Signed-off-by: Chih-Chung Chang <chihchung@chromium.org>
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Used kmemdup instead of replicating it's behaviour with kmalloc followed
by memcpy.
Patch found using coccinelle.
Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Used kmemdup instead of replicating it's behaviour with kmalloc followed
by memcpy.
Patch found using coccinelle.
Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for the independent HP reduced the availability of the
side surround output, because there are only 4 DACs for 7.1 and a HP
outputs. Adjust the badness tables for VIA so that 7.1 outputs are
activated for the cost of missing independent HP.
Once when we implement the dynamic DAC switching to multiple outputs,
this conflicts will be eased in future...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The lack of independent HP mode shouldn't be too bad, but currently
its badness is set a bit too high. Let's lower it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The standard badness values don't seem to fit to all preferences.
Some configuration prefer the side output over the headphone, some
want the speaker over the surround, etc.
This patch moves the badness table pointers into hda_gen_spec, so that
the codec driver can override them.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge back for-linus branch for the badness table adjustment for VIA codecs
* for-linus:
ALSA: hda - Fix DAC assignment for independent HP
ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader
ALSA: hda - Fix typo in checking IEC958 emphasis bit
ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()
ALSA: snd-usb: mixer: propagate errors up the call chain
ALSA: usb: Parse UAC2 extension unit like for UAC1
ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
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
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>
Now that we have a "Headset Mic" name, let's use it for some devices
we know for sure has a headset mic jack.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Headset mic jacks, i e TRRS style jacks with Headphone Left,
Headphone Right, Mic and GND signals, are becoming increasingly
common and are now being shipped by several manufacturers.
Unfortunately, the HDA specification does not give us any hint
of whether a Mic pin belongs to such a jack or not, but it would
still be helpful for the user to know (especially if there is one
TRS Mic jack and one TRRS headset jack).
This new fixup causes the first (non-dock, non-internal) mic to
be a headset mic jack. The algorithm can be later refined if needed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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 new HP desktop machines have Realtek codecs and their LEDs are
controlled via GPIO as for many laptop models. Add similar hooks as
well as in patch_sigmatel.c for controlling LEDs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The verb to set up the digital beep via AC_VERB_SET_DIGI_CONVERT_2
should be executed at resume as well. Use the cached write for it
being performed automatically at resume.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While playing the digital beep tone, the codec shouldn't be turned
off. This patch adds proper snd_hda_power_up()/down() calls at each
time when the beep is played or off.
Also, this fixes automatically an unnecessary codec power-up at
detaching the beep device when the beep isn't being played.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of calling snd_hda_attach_beep_device() and
snd_hda_detach_beep_device() in each codec driver, move them to the
generic parser. The codec driver just needs to set spec->beep_nid for
activating the digital beep.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dylan Reid