Pulseaudio SBC codec defines that audio samples are in PA_SAMPLE_S16LE
format which is little endian. But libsbc library expects audio samples by
default in host endianity which is big endian on big endian system. So SBC
support on big endian system is broken. To fix this problem tell libsbc
library that audio samples are in little endian to match PA_SIMPLE_S16LE
sample format.
Bug: https://bugs.freedesktop.org/show_bug.cgi?id=91359
Remove dead code and replace numeric bitpool values by macro definitions.
Maximal bitpool value in fill_capabilities() was reduced from 64 to 53
(SBC_BITPOOL_HQ_JOINT_STEREO_44100) because default_bitpool() already set
maximal value to 53.
This patch does not change SBC behavior as maximal bitpool was already
limited to 53. So it is just clean up.
This patch introduce new modular API for bluetooth A2DP codecs. Its
benefits are:
* bluez5-util and module-bluez5-device does not contain any codec specific
code, they are codec independent.
* For adding new A2DP codec it is needed just to adjust one table in
a2dp-codec-util.c file. All codec specific functions are in separate
codec file.
* Support for backchannel (microphone voice). Some A2DP codecs (like
FastStream or aptX Low Latency) are bi-directional and can be used for
both music playback and audio call.
* Support for more configurations per codec. This allows to implement low
quality mode of some codec together with high quality.
Current SBC codec implementation was moved from bluez5-util and
module-bluez5-device to its own file and converted to this new A2DP API.
This adds API to allow clients to schedule a callback in the mainloop
thread without the mainloop lock being held. This is meant for a case
where the client might be dealing with locking its own objects in
addition to the mainloop thread itself. In this case, it might need ton
control the locking order of the two, to match the order in other
threads, as it might not always be able to allow for its objects to be
locked after the mainloop thread lock.
The current null-source implementation has several bugs:
1) The latency reported is the negative of the correct latency.
2) The memchunk passed to pa_source_post() is not initialized
with silence.
3) In PA_SOURCE_MESSAGE_SET_STATE the timestamp is always set
when the source transitions to RUNNING state. This should only
happen when the source transitions from SUSPENDED to RUNNING
but also if it changes from SUSPENDED to IDLE.
4) The timing of the thread function is incorrect. It always
uses u->latency_time, regardless of the specified source
latency.
5) The latency_time argument seems pointless because the source
is defined with dynamic latency.
This patch fixes the issues by
1) inverting the sign of the reported latency,
2) initializing the memchunk with silence,
3) changing the logic in PA_SOURCE_MESSAGE_SET_STATE so that
the timestamp is set when needed,
4) using u->block_usec instead of u->latency_time for setting
the rtpoll timer and checking if the timer has elapsed,
5) removing the latency_time option.
So far PulseAudio only supported two different work formats: S16NE if
it's sufficient to represent the input and output formats without loss
of precision and FLOAT32NE in all other cases. For systems that use
S32NE exclusively, this results in unnecessary conversions from S32NE to
FLOAT32NE and back again.
Add S32NE remap operations and make use of them (for the COPY and
TRIVIAL resamplers) if both input and output format are S32NE. This
avoids the back and forth conversions between S32NE and FLOAT32NE,
significantly improving performance for those cases.
pa_init_remap_func() takes care to initialise pa_remap_t.do_remap to
NULL before calling init_remap_func (the CPU-specific remap init
function) and invokes init_remap_c if init_remap_func did not set
pa_remap_t.do_remap to non-NULL. remap_init_test_channels() calls
init_remap_func() directly so it must make sure pa_remap_t.do_remap is
set to NULL. Otherwise we'll end up with a random value in
pa_remap_t.do_remap if there is no CPU-optimised remap function for the
current operation.
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>
This is added to keep backward compatibility. The default value of
this new argument is false. Therefore, triggering by source-output
will be activated only if it is set to true explicitly.
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
Previously, media.role property of only sink-input is used to
determine to trigger and apply ducking or cork to sink-inputs.
On the other hand, some use cases require that source-output
also need to trigger the effect to sink-inputs. Therefore this
patch adds logic to retrieve source-ouputs to find trigger role
by checking media.role property and apply ducking/cork to sink-
inputs that meet conditions.
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
Currently, the ladspa-sink is not suspended when the master sink is suspended.
With this patch, the ladspa-sink will be suspended with suspend cause
PA_SUSPEND_UNAVAILABLE when the master sink is suspended for other reasons
than PA_SUSPEND_IDLE. This fixes issue #15.
Currently, virtual sinks and sources are not suspended when the master sink
or source is suspended. To implement this, the slave must be able to track
the suspend cause of the master.
With this patch, the sink input suspend callback will not only be called
when the sink or source is changing state, but also when the suspend cause
changes. Similar to the set_state_in_*_thread_cb() functions, the suspend
callback receives a state and a suspend cause as additional arguments.
Because the new state and suspend cause of the sink or source have already
been set, the old values are passed to the callback.
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.
After a suspend/resume cycle of a system, it may be possible that module-loopback
accumulates several seconds of audio in the memblockq before the alsa sink becomes
active again. Also it may be possible for other reasons that the actual loopback
latency is too different from the target latency to be adjusted in a reasonable
time by the normal rate controller.
This patch adds the option fast_adjust_threshold_msec to module-loopback. If set,
the latency will be forcefully adjusted to the target latency by dropping or
inserting samples if the actual latency differs more than fast_adjust_threshold_msec
from the target latency.
Also the calculation of the real adjust time would fail when the system was
suspended because that case was not considered. Now the real adjust time
calculation is skipped if the time passed between two calls of adjust_rates()
appears significantly too long.
pa_split_in_place() and pa_split_spaces_in_place() are modifed
to use size_t type instead of integer type.
alsa-ucm.c is revised according to this change.
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
The old minimum version was set in commit 57e3ccaf51 based on what the
commit author happened to have installed at the time. Russell Treleaven
now confirmed that Debian 8's gettext version, 0.19.3, works fine too,
or at least PulseAudio builds without errors. There might be room to
lower the required version even further, but that requires someone to
test older gettext versions.
In a former commit 37358e42c4 ("alsa: Suppress udev detection of sound
card for some units on IEEE 1394 bus"), PulseAudio has udev rules to
suppress handling some units on IEEE 1394 bus for a below issue:
Bug 199365 - repeating bus resets on Firewire bus with Focusrite Saffaire 26/io
https://bugzilla.kernel.org/show_bug.cgi?id=199365
However, I found that the rules match another model; Focusrite Liquid
Saffire 56. For detail, refer to below patch for Linux sound subsystem:
[alsa-devel] [PATCH] ALSA: bebob: use more identical mod_alias for
Saffire Pro 10 I/O against Liquid Saffire 56
https://mailman.alsa-project.org/pipermail/alsa-devel/2019-February/146003.html
For PulseAudio, the udev rule should be improved, because Liquid Saffire 56
(an application of TCAT TCD2200 ASIC, a.k.a Dice Jr.) can be handled by
pulseaudio without the issue.
This commit changes udev rule with model name instead of model_id from
configuration ROM. Below is data on udevd for Liquid Saffire 56, for
your information:
$ udevadm info -q all -p /sys/bus/firewire/devices/fw1.0/sound/card2/
P: /devices/pci0000:00/0000:00:01.2/0000:03:00.2/0000:04:07.0/0000:0a:00.0/0000:0b:00.0/fw1/fw1.0/sound/card2
E: DEVPATH=/devices/pci0000:00/0000:00:01.2/0000:03:00.2/0000:04:07.0/0000:0a:00.0/0000:0b:00.0/fw1/fw1.0/sound/card2
E: ID_BUS=firewire
E: ID_FOR_SEAT=sound-pci-0000_0b_00_0
E: ID_ID=firewire-0x00130e04018001e9
E: ID_MODEL=LIQUID_SAFFIRE_56
E: ID_MODEL_FROM_DATABASE=XIO2213A/B/XIO2221 IEEE-1394b OHCI Controller [Cheetah Express]
E: ID_MODEL_ID=0x000006
E: ID_PATH=pci-0000:0b:00.0
E: ID_PATH_TAG=pci-0000_0b_00_0
E: ID_PCI_CLASS_FROM_DATABASE=Serial bus controller
E: ID_PCI_INTERFACE_FROM_DATABASE=OHCI
E: ID_PCI_SUBCLASS_FROM_DATABASE=FireWire (IEEE 1394)
E: ID_SERIAL=0x00130e04018001e9
E: ID_SERIAL_SHORT=0x00130e04018001e9
E: ID_VENDOR=Focusrite
E: ID_VENDOR_FROM_DATABASE=Texas Instruments
E: ID_VENDOR_ID=0x00130e
E: SOUND_INITIALIZED=1
E: SUBSYSTEM=sound
E: SYSTEMD_WANTS=sound.target
E: TAGS=💺systemd:
E: USEC_INITIALIZED=9802422583
Fixes: 37358e42c4 ("alsa: Suppress udev detection of sound card for some units on IEEE 1394 bus")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Similar to module-tunnel-sink-new, module-virtual-source did not create
a rtpoll for the uplink sink. This lead to a crash when the uplink sink
was used by module loopback, because module-loopback relies on the sink
to provide a rtpoll. Additionally, the sink was not unlinked when the
module was unloaded.
This patch fixes both issues. The rtpoll created is never run by the sink,
so the patch is no real fix but just a workaround to make module-loopback
happy.
pa_card_profile_set_available needs to check if the card is linked
before firing PA_CORE_HOOK_CARD_PROFILE_AVAILABLE_CHANGED, so callbacks
connected to it receive a fully initialized card object.
This fixes a crash introduced by commit 30a551bbc
"switch-on-port-available: Check if we need to change the active
profile".
If one device tries to use PulseAudio to send audio over A2DP to another
device with bluez-alsa, that doesn't work because PulseAudio uses an
incorrect RTP payload type and bluez-alsa checks that the RTP payload
type is correct. According to the A2DP spec, the payload type should be
set to a number between 96 and 127.
Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/591
We split out some of the check-daemon tests that take a long time to
run, and also reduce how long we wait for the daemon to start up. This
should make the CI process quicker.
This allows us to disable automatically updating build system files in
case things change. This is desirable in the common case, but not
necessarily for CI, where we want the ability to take a build directory
as an artifact from one stage to the next (i.e. into a fresh checkout).
