Add a UCM configuration for the rt5650 codec. Tested on
a Samsung Chromebook 3. Adapted with minor modifications
from GitHub user evan-a-a's gist:
https://gist.github.com/evan-a-a/86b2a698708074530e2d0ee7c6498767
Signed-off-by: Russell Parker <russell.parker7@gmail.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The recent change to support the drain via polling caused a regression
for pulse plugin; with speaker-test -c2 -twav with pulse, it leads to
either no sounds or stall.
The only sensible behavior change in the commit wrt pulse plugin is
that now it starts the stream before calling drain callback. This
supposed to be correct, but it seems hitting a pulse plugin bug.
The start before drain callback is only a matter of consistency, and
since this doesn't work for the single existing plugin using drain
callback, we don't need to stick with this behavior.
For addressing the regression, we check the presence of the drain
callback and start the stream only when it doesn't exist, i.e. only in
drain-via-poll mode.
Fixes: ce2095c41f ("pcm: ioplug: Implement proper drain behavior")
Reported-by: Diego Viola <diego.viola@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this patch it is not possible to link the channel and format
parameter if snd_pcm_extplug_set_param_*() or
snd_pcm_extplug_set_slave_param_*() is called. Therefore the client and
slave parameter can differ. So the extplug has to implement conversion.
To avoid this the new snd_pcm_extplug_set_param_link() function can be
called.
As a reproduction sceanrio the following extplug source code can be used:
===
static snd_pcm_sframes_t my_transfer(snd_pcm_extplug_t *e,
const snd_pcm_channel_area_t *da, snd_pcm_uframes_t dof,
const snd_pcm_channel_area_t *sa, snd_pcm_uframes_t sof,
snd_pcm_uframes_t s) {
return s;
}
static const snd_pcm_extplug_callback_t my_own_callback = {
.transfer = my_transfer
};
SND_PCM_PLUGIN_DEFINE_FUNC(my_plug) {
snd_config_iterator_t i, next;
snd_config_t *slave = NULL;
snd_pcm_extplug_t *myplug;
snd_config_for_each(i, next, conf) {
snd_config_t *n = snd_config_iterator_entry(i);
const char *id;
if (snd_config_get_id(n, &id) < 0)
continue;
if (strcmp(id, "comment") == 0 || strcmp(id, "type") == 0)
continue;
if (strcmp(id, "slave") == 0) {
slave = n;
continue;
}
return -EINVAL;
}
myplug = calloc(1, sizeof(*myplug));
myplug->version = SND_PCM_EXTPLUG_VERSION;
myplug->callback = &my_own_callback;
snd_pcm_extplug_create(myplug, name, root, slave, stream, mode);
snd_pcm_extplug_set_param_minmax(myplug,
SND_PCM_EXTPLUG_HW_CHANNELS, 1, 16);
// snd_pcm_extplug_set_param_link(myplug, SND_PCM_EXTPLUG_HW_CHANNELS, 1);
*pcmp = myplug->pcm;
return 0;
}
SND_PCM_PLUGIN_SYMBOL(my_plug);
===
To use this plugin the following ALSA configuration is required:
pcm.myplug {
type my_plug
slave.pcm hw:Dummy
}
With this configuration without this patch
snd_pcm_hw_params_get_channels_max() will always return 16 channels
independent of the supported channels of the dummy device. Due to that for
example the start up of JACK would fail:
$ modprobe snd_dummy
$ jackd -d alsa -P myplug
ALSA: cannot set channel count to 16 for playback
ALSA: cannot configure playback channel
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many UCM profiles include the UCM profile components under ucm/*
subdirectories and thusly put <searchdir:ucm> at each place. This is
rather cumbersome.
This patch makes the UCM parser to set the default include path, so
that each profile no longer needs to set searchdir. All the
<searchdir:ucm> lines currently found in the profiles are removed
gracefully, too.
For the needed implementation, a new helper,
_snd_config_load_with_include() is introduced. It's not exported,
only for the use inside alsa-lib.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have placed UCM profile snippets to be included by the main config
files also in the same directory, src/conf/ucm, it confuses alsaucm
program that scans over all subdirectories. It thinks such a file is
also the main config file, and spews errors like:
% alsaucm
ALSA lib utils.c:67:(uc_mgr_config_load) could not open configuration file /usr/share/alsa/ucm/bytcr/bytcr.conf
ALSA lib parser.c:1427:(load_master_config) error: could not parse configuration for card bytcr
alsaucm: unable to obtain card list: No such file or directory
Actually we already defined the subdirectory for such components, and
they are skipped at parsing the main configs. So we just need to move
the files there -- this is what's done here.
One more thing done here is to add a new component subdirectory,
platforms, for definitions bytcr/* that don't match with neither the
existing ones (codecs nor dsps).
Suggested-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Tested-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This Dock doesn't have IEC958 physical output, so add it to the
blacklist to prevent it being opened.
[ Also adding WD15 Dock entry that has the same problem -- tiwai ]
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Configuration to reproduce:
~~~~
pcm.share_right {
type dshare
ipc_key 73
ipc_perm 0666
slave {
pcm "hw:0,0"
}
bindings {
# the seagfault happens when we don't bind channel 0
1 1
}
}
~~~~
Execute to reproduce:
~~~~
$ aplay -D plug:share_right test.wav
Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
Segmentation fault
~~~~
For channels whithout binding, values are set to UINT_MAX in function
`snd_pcm_direct_parse_bindings()`:
~~~~
for (chn = 0; chn < count; chn++)
bindings[chn] = UINT_MAX; /* don't route */
~~~~
But, these values are not checked when playing, which causes the segfault.
This commit fixes the issue.
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These changes are required due to the kernel
commit 07b7acb51d283d8469696c906b91f1882696a4d4
("ASoC: rsnd: update pointer more accurate")
Issue is that snd_pcm_wait() goes back to waiting because the hw_ptr
is not period aligned. Therefore snd_pcm_wait() will block for a longer
time as required.
With these rcar driver changes the exact position of the dma is returned.
During snd_pcm_start they read hw_ptr as reference, and this hw_ptr
is now not period aligned, and is a little ahead over the period while it
is read. Therefore when the avail is calculated during snd_pcm_wait(),
it is missing the avail_min by a few frames.
An additional option hw_ptr_alignment is provided to dmix configuration,
to allow the user to configure the slave application and hw pointer
alignment at startup
[ Slight indentation and parentheses removals by tiwai ]
Signed-off-by: Laxmi Devi <Laxmi.Devi@in.bosch.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this change an interval of (x x+1] will be interpreted as an
empty interval but the right value would be x+1.
This leads to a failing snd_pcm_hw_params() call which returns -EINVAL.
An example issue log is given in the following:
snd_pcm_hw_params failed with err -22 (Invalid argument)
ACCESS: MMAP_NONINTERLEAVED
FORMAT: S16_LE
SUBFORMAT: STD
SAMPLE_BITS: 16
FRAME_BITS: 16
CHANNELS: 1
RATE: 16000
PERIOD_TIME: (15999 16000]
PERIOD_SIZE: (255 256]
PERIOD_BYTES: (510 512]
PERIODS: [2 3)
BUFFER_TIME: 32000
BUFFER_SIZE: 512
BUFFER_BYTES: 1024
In case of (x x+1) we have to interpret it anyway as a single value of x to
compensate rounding issues.
For example the period size will result in an interval of (352 353) when
the period time is 16ms and the sample rate 22050 Hz
(16ms * 22,05 kHz = 352,8 frames). But 352 has to be chosen to allow a
buffer size of 705 (32ms * 22,05 kHz = 705,6 frames) which has to be >= 2x
period size to avoid Xruns. The buffer size will not end up with an
interval of (705 706) simular to the period size because
snd_pcm_rate_hw_refine_cchange() calls snd_interval_floor() for the buffer
size. Therefore this value will be interpreted as an integer interval
instead of a real interval further on.
This issue seems to exist since the change of 9bb985c38 ("pcm:
snd_interval_refine_first/last: exclude value only if also excluded
before")
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The device name string for Dell WD15 (and its variants) dock is set as
"WD15Dock", while the actual device name to be used is "Dock".
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1112292
Fixes: 8ebb40c969 ("conf/ucm: Add a UCM profile for Dell WD15 Dock USB-audio")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The qlen field of struct snd_midi_event was declared as size_t while
status_events[] assigns the qlen to -1 indicating to skip. This leads
to the misinterpretation since size_t is unsigned, hence it passes the
check "dev.qlen > 0" incorrectly in snd_midi_event_encode_byte(),
which eventually results in a memory corruption.
Also, snd_midi_event_decode() doesn't consider about a negative qlen
value and tries to copy the size as is.
This patch fixes these issues: the first one is addressed by simply
replacing size_t with ssize_t in snd_midi_event struct. For the
latter, a check "qlen <= 0" is added to bail out; this is also good as
a slight optimization.
Reported-by: Prashant Malani <pmalani@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When seccomp policy is applied to filter ioctl syscall with
SNDRV_CTL_IOCTL_TLV_COMMAND, SNDRV_CTL_IOCTL_TLV_READ and
SNDRV_CTL_IOCTL_TLV_WRITE in whiltelist, alsa-lib still breaks
in at snd_ctl_hw_elem_tlv().
The problem behind is because ioctl() takes unsigned long cmd
argument, and the signed bit of local int variable could cause
0xff bytes appended after casted to unsigned long.
In kernel, seccomp data struct takes 64 bits argument to check
against seccomp rules, these unexpected 0xff bytes could make
the rule check fail.
Fix the problem by passing unsigned int to ioctl.
Signed-off-by: Hsin-Yu Chao <hychao@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Comment field is displayed tot the end user in various UIs as such
names like MonoSpeaker and DigitalMics without any spaces are no good.
Also the names themselves as well as how they get displayed in the
typical UI (in separate input / output tabs) makes the adding of
playback and capture to the comment superfluous and this looks weird
in the UI, so drop it.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add a longname profile for devices with a mono speaker, the Internal Mic
hooked up to IN2 and the left and right channels of their headphones
output swapped.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
After recent kernel work, the kernel now sets a long-name for bytcr-rt5651
boards which indicates if a single (mono) speaker or stereo speakers are
used and if in1, in2, or in1 and 2 are used for the internal mic(s) (the
headset mic sofar is always on in3).
This commit adds UCM profiles for bytcr-rt5651 boards using these new
long-names, based on the generic bytcr-rt5651 profile.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Many rt5651 devices only have a single speaker and even though there is
some external mixing done on the PCB, the quality of that mixing is quite
poor and various sounds come out garbled when relying on the on PCB mixing.
Using the codecs builtin mixer to mix left + right to the left output works
much better. This commits adds a new MonoSpeaker.conf output profile which
allows this.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Split the bytcr-rt5651 config into 1 .conf file per input / output as
has already been done for the bytcr-rt5640 and the chtnau8824 profiles.
This allows easy creation of long-name profiles with the specific input /
output combinations found on a board without needing to copy and paste
things.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Note this commit replaces the pre-existing "Handset Microphone" and
"Main Microphone" options, these come from the first commit of the
bytcr-rt5651 UCM profile and were based on wrong assumptions about the
input mappings. None of the existing devices has the Hand/Headset mic
on IN1 as these options assumed.
The rt5651 is used in various configurations with the Internal Mic(s)
hooked up to IN1, IN2, or to IN1 and IN2 and the Headset Mic hooked up
to IN3.
Add support for all these to the generic bytcr-rt5651 profile and name
them accotding to their input + functions.
A follow up commit will add specialized longname configs which
will only expose the inputs actually used on the board with that
longname.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Fix ADC and Mic capture volumes, so that the microphone inputs actually
work.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the generic SSP enable sequence from bytcr/PlatformEnableSeq.conf,
for boards using SSP2 this is identical the code it replaces and this
adds support for boards using SSP0.
This fixes sound not working on Bay Trail CR tablets with a rt5651 codec.
This commit also calls the generic disable sequence on shutdown
(this is new).
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
pulseaudio will run the DisableSequence of the current playback device
before running the EnableSequence of the new playback device.
This causes the Platform Clock and BIAS to temporarily get turned off which
on the rt5651 breaks audio-streams which are playing when switching.
This commit moves the disabling to the EnableSequence of the other device
fixing this.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Start with all switches disabled, so that e.g. the
LOUT L/R Playback Switches are not left enabled when starting with
headphones plugged in.
This fixes the platform clock being kept on by these in some cases.
While at also move the IN? Boost and IF1 ASRC Switch lines around
a bit to match the order from https://github.com/plbossart/UCM so
the profiles can be more easily compared.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The volumes are taken from this commit:
753e2430cd
That commit also adds line-in support, so it has not been
taken in its entirety.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The headphones can either be driven directly from DAC1, or through
the HP volume mixer chain to allow volume control, both can be enabled
at the same time, but this should not be done.
Mix only DAC1 to the headphones and not the HP volume path, there
are 2 reasons to choice the DAC1 path;
1) It is the power-on-reset default
2) We don't expose the volume control to e.g. pulseaudio anyways so it
is not useful
While at it also move the "HPO MIX DAC1" and "HPO MIX HPVOL" entries up a
bit so that they are no longer inbetween the "HPO L Playback Switch" and
"HPO R Playback Switch" entries.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the generic Intel SSP bytcr/PlatformEnableSeq.conf file, it is
identical to all the cset statements this commit removes.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The snd_pcm_mmap_begin() call returns the amount of contiguous data,
which is less than the total available if it wraps around the buffer
boundary.
If we don't handle this split we leave stale data in the buffer that
should have been overwritten, as well as unread data in the io_plugin
that gets transferred on a subsequent call at the wrong offset.
Signed-off-by: Rob Duncan <rduncan@teslamotors.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without these changes a negative error code returned by
snd_pcm_avail_update() will be not handled correctly.
With this patch the returned error code of snd_pcm_avail_update() will be
returned by snd_pcm_rate_avail_update().
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This function can be called without calling snd_pcm_avail_update().
The call to snd_pcm_avail_update() can take some time.
Therefore some developers would not like to call it from a real-time
context (e.g. from JACK client context).
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this commit the following intervals [x y), (x y) were be
replaced to (y-1 y) by snd_interval_refine_last(). This was also done if
y-1 is part of the previous interval.
With this changes it will be replaced with [y-1 y) in case of y-1 is
part of the previous interval. A similar behavior will be used for
snd_interval_refine_first().
This solves the issue reported here:
https://bugzilla.opensuse.org/show_bug.cgi?id=1033179
and work arounded with commit
e736715 ("pcm: dmix: Disable var_periodsize as default").
I am able to reproduce the issue with a simplified aplay use case using
the following configuration:
pcm_slave.adr3_tdm_8ch {
pcm {
type hw
card "Loopback"
device 0
}
rate 48000
period_size 128
buffer_size 1024
channels 2
}
pcm.dshare_Playback_3 {
type dmix
ipc_key 600
ipc_perm 0660
ipc_gid audio
var_periodsize true
slave adr3_tdm_8ch
}
pcm.AdevAcousticoutSpeech {
type rate
slave.pcm dshare_Playback_3
slave.rate 48000
}
$ modprobe snd_aloop
$ aplay -v --period-size=352 -c2 -fS16_LE -r22500 -DAdevAcousticoutSpeech /dev/urandom
...
Rule 9 (0xffff91d1f230): PERIODS=(0 2) -> NONE BUFFER_SIZE=480 PERIOD_SIZE=[240 240]
refine_soft 'AdevAcousticoutSpeech' (end--22)
...
aplay: ../../alsa-utils-1.1.5/aplay/aplay.c:1390: set_params: Assertion `err >= 0' failed.
Aborted by signal Aborted...
The following stack trace shows where the -EINVAL will be thrown:
__snd_pcm_hw_params_set_period_size_near()
snd1_pcm_hw_param_set_near()
snd1_pcm_hw_param_set_last()
snd1_pcm_hw_refine_slave()
snd1_pcm_hw_refine_soft()
snd_pcm_hw_rule_div()
snd1_interval_refine()
This issue exists due to PERIODS does not include 2
Rule 9 (0xffff91d1f230): PERIODS=(0 9) -> (0 2) BUFFER_SIZE=[120 480]
PERIOD_SIZE=(240 241)
because of an invalid integer inverval of PERIOD_SIZE of (240 241).
This interval is set by snd_interval_refine_last().
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of the expanded bit numbers like 0x81ffffff, list up the all
supported PCM bits explicitly for refine_masks[] in pcm_params.c.
This makes easier to update any additional formats or other
parameters, and easier to spot out missing ones.
Actually the GSM and DSD formats were missing; with this commit, they
are supported properly now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Other fixes include output/input names (comments) for UIs (pavucontrol)
to display, and Playback/CapturePCM entries so pulseaudio initializes
correctly on this hardware.
Signed-off-by: Urja Rannikko <urjaman@gmail.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
In the commit 38a2d2eda8 ("pcm: dmix: Do not discard slave reported
delay in status result"), the PCM dmix hwptr update code was rewritten
to follow the slave PCM hwptr update. This is based on the similar
change in PCM dshare, the commit faf53c197c.
There was a bug in the commit 38a2d2eda8 regarding the PCM state
change, and it was addressed in commit 3752e6b873 ("pcm: dmix: Fix
the inconsistent PCM state"). However, we've hit yet another bug in
this commit. Namely, the hwptr update was forgotten in the
snd_pcm_dmix_sync_ptr0() function. So the hwptr value passed from
snd_pcm_dmix_status() isn't properly stored, and it screws up at some
long run occasionally.
This patch covers the bug by replacing with the right value.
Fixes: 38a2d2eda8 ("pcm: dmix: Do not discard slave reported delay in status result")
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=200013
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Cube iWork8 Air and Pipo W2S tablets both only have a single speaker.
Add long-name profiles for them which are identical to the default
chtnau8824 profile, except that they include the nau8824/MonoSpeaker.conf
snippet instead of the nau8824/Speaker.conf one.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add UCM profile for chtnau8824 boards based on:
https://github.com/plbossart/UCM/blob/master/chtnau8824
Split into multiple files in the same way as this was done for the
bytcr-rt5640 support, re-using the existing ucm/PlatformEnableSeq.conf
and ucm/PlatformDisableSeq.conf files for the SST mixer settings.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add a disable sequence powering off the SST mixer elements, loosely
based on the default DisableSequence from:
https://github.com/plbossart/UCM/blob/master/chtnau8824/HiFi.conf
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
With a recently merged kernel commit, the kernel now sets a long-name for
bytcr-rt5640 boards which indicates if a single (mono) speaker or stereo
speakers are used and wether dmic1, in1 or in3 is used for the internal
mic (the headset mic sofar is always in2).
This commit adds UCM profiles for bytcr-rt5640 boards using these new
long-names, based on the generic bytcr-rt5640 profile.
The added profiles have the unnecessary input / output options from the
generic profile removed leaving only 2 input and 2 output options, which
are automatically switched between by e.g. pulse based on jack-detect.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This commit adds the generic UCM profile for bytcr-rt5640 boards from:
https://github.com/plbossart/UCM, plus the fixes from this pull-req:
https://github.com/plbossart/UCM/pull/31
The profile has been split up into separate per input / output files to
allow for creation of long-name profiles with the specific input / output
combinations found on a board without needing to copy and paste things.
Note this profile exports all inputs and both stereo/mono speaker setups
even though a typical device will not use all. Ideally a long-name based
device specific profile made up of the various parts should be used
instead.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset
of TLV data. This commit applies a code optimization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset
of TLV data. This commit applies a code optimization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset
of TLV data. This commit applies a code optimization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A series of SNDRV_CTL_TLVO_XXX macro was introduced for position offset
of TLV data. This commit applies a code optimization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB-audio device on Dell WD15 docking station provides two individual
PCM streams, one for headphone and another for line out. A UCM
profile gives the proper roles for these.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Lenovo Ideapad Miix 320 uses a digital mic connected to the DMIC2 input
(unlike the Asus T100HA which has it connected to the DMIC1 input), add a
long-name config specific for the Miix 320, which is a copy of the standard
chtrt5645 config with the internal analog mic section replaced with one
for a digital mic connected to the DMIC2 input.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Asus T100HA uses a digital mic rather then an analog one, add
long-name config specific for the T100HA, which is a copy of the standard
chtrt5645 config with the internal analog mic section replaced with one
for the digital mic found on the Asus T100HA.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The internal analog mic switch is called 'Int Analog Mic Switch'
(not 'Int Mic Switch') and is connected to BST2 not BST1.
Also change the analog mic volume levels so that we get better
audio / less noise.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
