The full identifier check must be executed for the new melem
creation, otherwise the duplicate control element code check
is reached.
Example (using the snd-aloop driver):
numid=56,iface=PCM,name='PCM Notify',device=1,subdevice=1
numid=62,iface=PCM,name='PCM Notify',device=1,subdevice=2
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/730>
The new helem must be tracked and old helem must be cleared
to make the code work properly. Introduce the pointer to helem
as the private value for melem and add the necessary code.
Also, add a check for the duplicate mixer elements. The duplicate
mixer element invokes the abort check in alsa-lib. Print a warning
instead and handle the exit gracefully.
Fixes: def8eb074 ("alsa-mixer: allow to re-attach the mixer control element")
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/730>
It may be possible that the ALSA control element appears
again. Allow this combination by checking, if the pulseaudio
mixer element already exists. Do not create the duplicate
mixer element in this case.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/729>
PulseAudio v5.99 or later hits assertion at alsa-lib mixer API due to
wrong handling of removal event for mixer element.
pulseaudio: mixer.c:149: hctl_elem_event_handler: Assertion `bag_empty(bag)' failed.
The removal event is defined as '~0U', thus it's not distinguished from
the other type of event just by bitwise operator.
At the removal event, class implementator for mixer API should detach
mixer element from hcontrol element in callback handler since alsa-lib
has assertion to check the list of mixer elements for a hcontrol element
is empty or not after calling all of handlers. In detail, please refer to
MR to alsa-lib:
* https://github.com/alsa-project/alsa-lib/pull/244
This commit fixes the above two issues. The issue can be regenerated by
`samples/ctl` Python 3 script of alsa-gobject.
* https://github.com/alsa-project/alsa-gobject/
It adds some user-defined elements into sound card 0. When terminated by
SIGINT signal, it removes the elements. Then PulseAudio dies due to the
assertion.
Fixes: 1fd8848e64 ("alsa-util: Add functions for accessing mixer elements through mixer class")
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/728>
The hw: device can be addressed using the card index (hw:0)
or the card identifier (ASCII string - hw:Loopback). Both
mixers are equal.
The previous code was fine for the mixers without the UCM
private prefixes (_ucmXXXX). Make code more robust, create
two aliased mixer structures in the mixers array.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/673>
Almost all reports from users, I have seen in last years, were not valid.
The report is also printed when the system scheduler does not wake
the pulseaudio thread in the right time. Users are not able to distinguish
between slow machine and the real problem.
Move the log level from 'error' to 'debug' for those messages.
The right fix should be to measure the time between the call invocation and
return to determine (and skip) the scheduling problems, but it is another
extra code just to debug things.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The ALSA mixer can be opened multiple times (especially for UCM
in the probe). This adds a simple mixer cache to prevent
multiple open calls.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This is just invalid. It results to an error in almost all cases.
The hw:<number> scheme is sufficient to get the right card mixer.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Hardwares without SNDRV_PCM_INFO_RESUME capability, like USB Audio,
don't support snd_pcm_resume() when it's in suspended state.
Let's use snd_pcm_hw_params_can_resume() to check hardware's capability
before snd_pcm_resume() attempt. If it doesn't support resume, just go
to snd_pcm_drop() to leave suspended state directly.
Consumers are expected to use <alsa/asoundlib.h> instead of
<asoundlib.h>.
This is in preparation of an change to pkgconfig(alsa) to
not pollute CFLAGS with -I/usr/include/alsa anymore.
Signed-off-by: Olaf Hering <olaf@aepfle.de>
Currently, when a system is waking up from suspend, the resume process of the
ALSA sink and source is unstable. Sometimes the device needs to be restarted
multiple times and when the system was suspended between snd_pcm_mmap_begin()
and snd_pcm_mmap_commit(), pulseaudio crashes on resume.
Additionally, variables are not reset after the resume, so that sink/source
report wrong latencies.
This patch fixes the issues by closing and re-opening the PCM if recovery
from an error condition is not possible. Additionally, the variables are
reset, so that latencies are reported correctly.
There has been a function to get supported sample rates from alsa and
an array for it in userdata of each module-alsa-sink/source. Similarly,
this patch adds a function to get supported sample formats(bit depth)
from alsa and an array for it to each userdata of the modules.
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
The commit "alsa-util: Set ALSA report_delay flag in pa_alsa_safe_delay()"
broke the build on ALSA versions below 1.1.0 because the time stamp
configuration function was introduced in 1.1.0.
This patch makes the usage of snd_pcm_status_set_audio_htstamp_config()
dependent on ALSA version.
The current code does not call snd_pcm_status_set_audio_htstamp_config()
to configure the way timestamps are updated in ALSA. In kernel 4.14 and
above a bug in ALSA has been fixed which changes timmestamp behavior.
This leads to inconsistencies in the delay reporting because the time
stamp no longer reflects the time when the delay was updated if the
ALSA report_delay flag is not set. Therefore latencies are not calculated
correctly.
This patch uses snd_pcm_status_set_audio_htstamp_config() to set the
ALSA report_delay flag to 1 before the call to snd_pcm_status(). With
this, time stamps are updated as expected.
In alsa-lib, snd_pcm_hw_params() internally calls snd_pcm_prepare(), thus
user space applications have no need to call snd_pcm_prepare() after calls
of snd_pcm_hw_params(). An explicit calls of snd_pcm_prepare() is expected
in a case to recover PCM substreams.
Current implementation of PulseAudio modules for ALSA playbacking/capturing
results in double calls of snd_pcm_prepare(). The second call for hw plugin
of alsa-lib executes ioctl(2) with SNDRV_PCM_IOCTL_PREPARE command in state
of SNDRV_PCM_STATE_PREPARED for the PCM substream. This has no effects to
the PCM substream as long as corresponding drivers are implemented
correctly.
This commit removes the second call for the reason.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
This isn't a great fix, but we need ALSA API to do this right. In the
mean time, USB devices work fine with timer-based scheduling, so there's
no reason to force a large minimum latency by disabling tsched on them.
...because we will later try with plug:* which will probably succeed,
so this is not an error.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
FSF addresses used in PA sources are no longer valid and rpmlint
generates numerous warnings during packaging because of this.
This patch changes all FSF addresses to FSF web page according to
the GPL how-to: https://www.gnu.org/licenses/gpl-howto.en.html
Done automatically by sed-ing through sources.
Now that we have switched to using the mixer handle only,
there is no use for sending hctl handles around.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Instead of using the hctl interface, we can find controls belonging
to other iface types than "mixer". We do this by introducing a new
mixer class "SND_MIXER_ELEM_PULSEAUDIO" and create snd_mixer_elem's
for all PCM and CARD iface types (as Jacks are of the CARD type and
ELD controls are of the PCM type).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
hwparams_copy needs to be reset (as it is also reset for the third and
fourth try) before the second try.
If the reset is not done and the first try fails:
D: [lt-pulseaudio] alsa-util.c: Maximum hw buffer size is 743 ms
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_period_size_near() failed: Invalid argument
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
D: [lt-pulseaudio] alsa-util.c: Set only period size (to 1102 samples).
We have three failures and finally the fourth (only period size) succeed.
With this patch:
D: [lt-pulseaudio] alsa-util.c: Maximum hw buffer size is 743 ms
I: [lt-pulseaudio] alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Invalid argument
D: [lt-pulseaudio] alsa-util.c: Set period size first (to 1102 samples), buffer size second (to 4408 samples).
We only fail with the first try, the second (period followed by buffer) is
fine.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
When given an explicit device.description in card_properties, prefer
this information over other default prefixes (e.g. 'Built-in Audio')
when constructing sink/source descriptions.
For example, if I manually configure the card description to be
"FooBar", I then expect that the sinks and created by the card also
have "FooBar" in their description instead of generic "Built-in
Audio".
PCM Devices which have the BATCH flag set update the PCM pointer only with
period size granularity. Using timer based scheduling does not have any
advantage in this mode. For one devices which have that flag set usually update
the position pointer in software after getting the period interrupt. So
disabling the period interrupt is not possible for this kind of devices.
Furthermore writing to or reading from the buffer slice for the current period
is not possible since the position inside the buffer is not known. On the other
hand the tsched algorithm seems to get easily confused for this kind of
hardware, which results in garbled audio output. This typically means that timer
based scheduling needs to be manually disabled on systems with such devices.
Auto disabling tsched in this case allows these systems to run with the default
configuration.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
This patch removes all occurrences of double and triple
newlines.
Command used for this:
find . -type d \( -name ffmpeg \) -prune -o \
-regex '\(.*\.[hc]\|.*\.cc\)' \
-a -not -name 'adrian-aec.*' -a -not \
-name reserve.c -a -not -name 'rtkit.*' \
-exec sed -i -e '/^$/{N;s/^\n$//}' {} \;
Two passes were needed to remove triple newlines.
The excluded files are mirrored files from external sources.
This patch replaces every occurrence of ')\n{' with ') {'.
Command used for this:
find . -type d \( -name ffmpeg \) -prune -o \
-regex '\(.*\.[hc]\|.*\.cc\)' \
-a -not -name core-util.c -a -not \
-name adrian-aec.c -a -not -name g711.c \
-exec sed -i -e '/)$/{N;s/)\n{$/) {/}' {} \;
The excluded files are mirrored files from external sources.
Currently, this function only reads the monitor name, but could
be extended to read e g supported formats as well.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
We inadvertantly stopped supporting non-standard rates when the
passthrough work was done. This makes sure that if no standard rates are
supported, we try to fallback to whatever ALSA gives us.
Refactor code to fetch avail, delay and timestamp values
in a single call to snd_pcm_status().
The information reported is exactly the same as before,
however it is extracted in a more atomic manner to
improve timer-based scheduling.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
