Pass zero-length packets to the codec. BAP/ISO may use these to indicate
missing data.
Fix A2DP codecs to not parse input with spa_return_val_if_fail, that's
meant for assertions. Just return -EINVAL directly, it's normal that
input data may contain garbage.
If packet sequence number jumps ahead, or we would underflow, use
codec-provided packet loss concealment to produce some audio data.
When we produce it during underflow, skip the corresponding number of
sequence numbers of future packets.
If codec doesn't have PLC, keep the previous behavior (pad with zeros,
buffering pauses to wait for data).
LC3 and Opus have built-in support for packet loss concealment.
Add codec interface for that, and implement for LC3.
Extend media_codec interface so that packets not aligned with socket
reads can be handled, as in HFP. This is required for correct sequence
number counting, and for being able to run codec PLC *before* decoding
the next correctly received packet.
Update rate matching only once per process(). This ensures all nodes in
the group update their rate matching in the same way.
Also account for audio data in ISO output buffer in the reference time.
The calculations is in system clock domain, so when converting from
samples/duration to time rate difference should be accounted.
This does not have much effect in practice.
The rate matching calculations are done in the system clock domain. If
the driver ticks at a different rate, the correction factor needs to be
adjusted by the rate_diff.
This fixes ISO streams getting out of sync with each other when target
delay changes. This happens because typically one of them is the driver
and the other follower. Driver adjust clock rate, and follower does its
own adjustment *on top of that* so it rate matches more or less at
double speed. (The DLL of the follower to some degree corrects for
this, but can't do that when hitting RATE_CTL_DIFF_MAX and moreover it
acts with a delay.)
A2DP v1.4 uses the rfa bits for adding 5.1 and 7.1 configurations.
Clear those bits properly when sending configuration, in case remote
device sets them.
Significantly better CPU performance in lieu of canceller quality. Not
implemented for 0.x series, as there's a lot more to enable there (such
as routing modes), and I am hoping to drop support for those versions
before too long.
Ensure we have at least a `.` after `audioconvert.filter-graph`, so we
don't try to read past the end if it does not exist.
Also document in the param name that an index is expected.
When dialing an incorrect phone number some phones (e.g. iOS 18.5)
replies with OK but never send +CIEV updates, so there's no way to
know that the dial is not in progress and the call object should be
removed.
This change waits for +CIEV event to create the call object.
In USB Audio Class 2 (UAC2) setups, pitch control is handled by
feedback endpoints. The host adjusts its data rate accordingly.
When pitch control is active (pitch_elem), applying the default
delay-locked loop (DLL) bandwidth can lead to instability and
oscillations around the target rate.
This patch adds a new parameter, api.alsa.dll-bandwidth-max, to
configure the maximum DLL bandwidth. It introduces a new field
in the ALSA state to store this value.
By default, it uses SPA_DLL_BW_MAX, but when pitch control is in
use, setting it to a lower value (e.g. 0.02) helps ensure better
stability, based on empirical testing.
In some cases, it is possible that the follower shares a clock with the
driver, but the driver rate is not known when the follower is assigned
to the driver. If this happens, then state->driver_rate is 0, and when
setting the format, we might think that we need to resample (because
follower rate != driver rate). This can cause us to incorrectly halve
the period size for the node.
This was introduced in commit 0b67c10a9c,
which forces reevaluation of matching status on driver change.
To avoid this, let us also probe for the driver rate when updating the
matching status, so we can make the update more accurate.
Fixes warning:
[186/359] Compiling C object spa/plugins/v4l2/libspa-v4l2.so.p/v4l2-source.c.o
In file included from ../spa/plugins/v4l2/v4l2-source.c:164:
../spa/plugins/v4l2/v4l2-utils.c: In function ‘spa_v4l2_enum_format’:
../spa/plugins/v4l2/v4l2-utils.c:1103:22: warning: unused variable ‘drop_next’ [-Wunused-variable]
1103 | bool drop_next = false;
| ^~~~~~~~~
Some devices might have nonfunctional 'Pro Audio' sound. This patch adds a
new 'api.acp.disable-pro-audio' option to disable pro-audio profile entirely.
Because we can now destroy sources (and free the source structure) by
simply holding the lock, there is a window where we might access the
freed source.
When we in iterate release the lock and go into the epoll, another
thread might acquire the lock and delete the fd from epoll. This might
happen right after epoll detected activity on the fd. When iterate
manages to acquire the lock again, it will process to dispatch the
active fd and deref the ep.data pointer, which is now pointing to freed
memory.
Fix this by incrementing a removed_count whenever we remove a source.
Check the counter if it was the same as before the epoll otherwise we
can't assume all sources are alive still. Return in that case as if
there were no fds to poll. The caller should reenter the iterate at some
point and we will return all the fds with activity, minus the one that
got destroyed. We need to give control to the caller because part of the
removal could be to stop the loop iteration all together.
When we are the owners of the loop lock and we are not in the loop
thread itself, release all locks so that the loop can start processing
our invoke items and we get a chance to make progress. After that
re-acquire the locks.
This can happen when you change some of the core loop_locked() calls to
blocking _invoke functions that are called with the loop locked.
We have all core blocking invoke functions removed now so this is not
actually going to be used but just in case an application tries to
blocking invoke while locking the loop, this will now at least do
something else than deadlock.
This assumes that the modifier is always 'linear'. That is not quite
correct for all V4L2 formats. But PipeWire only uses input devices and
other modifiers are very unlikely.
This makes it possible to use DMABUFs with the GStreamer pipewiresrc.
This reverts commit b57b87abbb.
It turns out this device is subtily different and doesn't work with the
current profile configurations. A UCM profile was added to alsa-ucm-conf
instead.
Change media-source to use sco-io for HFP codecs.
Replace sco-source with media-source.
sco-source is mostly copypaste from media-source, only differed in the
IO handling.
Change media-sink to use sco-io for HFP codecs.
Move SCO fragmentation to sco-io side.
Replace sco-sink with media-sink.
sco-sink is mostly copypaste from media-sink, and only differed in the
fragmentation detail, which can as well be handled on sco-io side.
Don't try to write data in too large blocks.
This controls the maximum amount of data to send at once. sco-io will
buffer and fragment packets to the right size.
Previously in sco-sink, SO_SNDBUF was not set, so there could be a
longer queue in the socket.
E.g. LDAC doesn't have that. Add the missing guard that was accidentally
dropped in the rewrite.
Also explicitly filter out non-A2DP/BAP codecs that can't be used in
ensure_media_codec
We don't actually have to store the ramp parameters so allocate them on
the stack and then use them to generate the sequence.
Make it possible to generate a sequence into a custom buffer as well.
Make sure we use the right rate (the graph rate) to calculate the number
of samples when converting from time to samples.
