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>
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>
In development period for Linux v4.18, a series of SNDRV_CTL_TLVO_XXX
macro was introduced to kernel stuffs for position offset of TLV data.
This commit adds these macros to backport header in this library.
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>
Otherwise, the boost value is 0, and the sound captured from that
LineIn jack is too weak for users.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The board configuration file and verb conf file are allowed to be
in the folder named of card_long_name, so when finding the verb conf
file, we need to check if it is in the folder of card_long_name or
card_name.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The gain algorithm used in softvol can handle gain factors of up to
32767 which is slightly more than 90 dB, so allow a max_dB of 90 dB.
This doesn't affect existing asound.conf files, but does allow a
max_dB of up to 90 dB when needed.
Tested using Audacity that there is no undue distorsion or other
artefacts when 90 dB of gain is applied to a suitable signal (i.e.
a signal quiet enough not be clipped whan applying 90 dB of gain).
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, some parameter names in conf differ from field names in struct.
These look like typos.
This commit suggests to add aliases for such parameters, so that the names
in conf are similar to names in struct. This solution is backwards
compatible.
If the difference between conf names and struct names is done on purpose -
this commit can be dropped.
Signed-off-by: Kirill Marinushkin <k.marinushkin@gmail.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Mark Brown <broonie@kernel.org>
Cc: Pan Xiuli <xiuli.pan@linux.intel.com>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current comment makes not clear the direction of mclk. Previously, similar
description caused a misunderstanding for bclk_master and fsync_master.
This commit solves the potential confusion the same way it is solved for
bclk_master and fsync_master.
Signed-off-by: Kirill Marinushkin <k.marinushkin@gmail.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Mark Brown <broonie@kernel.org>
Cc: Pan Xiuli <xiuli.pan@linux.intel.com>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: alsa-devel@alsa-project.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clock gating parameter is a part of `dai_fmt`. It is supported by
`alsa-lib` when creating a topology binary file, but ignored by kernel
when loading this topology file.
After applying this commit, the clock gating parameter is not ignored any
more. This solution is backwards compatible. The existing behaviour is
not broken, because by default the parameter value is 0 and is ignored.
snd_soc_tplg_hw_config.clock_gated = 0 => no effect
snd_soc_tplg_hw_config.clock_gated = 1 => SND_SOC_DAIFMT_GATED
snd_soc_tplg_hw_config.clock_gated = 2 => SND_SOC_DAIFMT_CONT
For example, the following config, based on
alsa-lib/src/conf/topology/broadwell/broadwell.conf, is now supported:
~~~~
SectionHWConfig."CodecHWConfig" {
id "1"
format "I2S" # physical audio format.
pm_gate_clocks "true" # clock can be gated
}
SectionLink."Codec" {
# used for binding to the physical link
id "0"
hw_configs [
"CodecHWConfig"
]
default_hw_conf_id "1"
}
~~~~
Signed-off-by: Kirill Marinushkin <k.marinushkin@gmail.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Mark Brown <broonie@kernel.org>
Cc: Pan Xiuli <xiuli.pan@linux.intel.com>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: linux-kernel@vger.kernel.org
Cc: alsa-devel@alsa-project.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The values of bclk and fsync are inverted WRT the codec. But the existing
solution already works for Broadwell, see the alsa-lib config:
`alsa-lib/src/conf/topology/broadwell/broadwell.conf`
This commit provides the backwards-compatible solution to fix this misuse.
This commit goes in pair with the corresponding patch for linux.
Signed-off-by: Kirill Marinushkin <k.marinushkin@gmail.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Tested-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Mark Brown <broonie@kernel.org>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: linux-kernel@vger.kernel.org
Cc: alsa-devel@alsa-project.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Phiree U2SX is the successor of Phiree U2 and has the same unusual
configuration.
See ea865bba46 for reference.
Signed-off-by: Bruno Pagani <bruno.n.pagani@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the draining behavior of ioplug in the following
ways:
- When no draining ioplug callback is defined, implement the draining
loop using snd_pcm_wait*() and sync with the drain finishes.
This is equivalent with the implementation in the kernel write().
Similarly as in kernel code, for non-blocking mode, it returns
immediately after setting DRAINING state.
- The hw_ptr update function checks the PCM state and stops the stream
if the draining finishes.
- When draining ioplug callback is defined, leave the whole draining
operation to it. The callback is supposed to return -EAGAIN for
non-blocking case, too.
- When an error happens during draining, it drops the stream, for a
safety reason.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_wait() & co checks the current avail value and returns
immediately if it satisfies <= avail_min condition. It's good in
general except for one situation: draining. When the draining is
being performed in the non-blocking mode, apps are supposed to wait
via poll(), typically via snd_pcm_wait(). So this ends up with the
busy loop because of the immediate return from snd_pcm_wait().
A simple workaround is to put the PCM state check and ignore the
avail_min condition if it's DRAINING state. The equivalent check is
found in the kernel xfer code, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
