port_enum_params should use right param id for SPA_PARAM_Format.
Fix typoed , instead of ; at end of line. Pop frame before putting
state.offset, in case there would be padding added (apparently usually
no).
The free buffer check must happen before writing to check for leftover data in buffers. In case data is left over from previous submission, bitpool mustn't be increased.
Also improved logging by adding bitpool to the log message.
When we get buffers from the driver, check if we have at least as many
as we requested. If we have more, that's ok, we will simply not queue
them.
Previously we would ignore the input number of buffers and use the
allocated amount to fill up the buffer array, which might be too small
and then we crash.
Just like the optional build, make all field parsing optional. This
will leave the fields with their default values if they are not parsed
from the param.
We can then remove our custom functions and use the generic ones in
various places.
Set up the adapter and then set the started field to true before we
Start the follower and converter. It is possible that the follower will
start processing immediately so the started state needs to be set
correctly.
Similarly, first perform the Pause and Suspend calls on the follower
and converter before clearing our buffers and format so that things
are stopped properly.
See #2764
AVDTP in principle allows 62 endpoints, but in practice it appears some
devices (Samsung Galaxy Buds2 Pro, Redmi Buds 3 Lite, probably others)
fail to connect A2DP when the number is somewhere above 24. A2DP
connection works when initiated from the Central, but not when the
device itself does it, so these devices are not fully broken. We should
reduce the number of registered A2DP endpoints to avoid running into
problems with such broken devices.
Some of our source codecs are the same actual codec with the same
configuration, and don't need separate source endpoints.
Allow codecs to not have a registered endpoint (fill_caps == NULL), and
tolerate codecs with the same endpoint name. In codec switch, keep
track separately which of the codecs with the same endpoint name the
local endpoint is currently associated with.
Pick the first format in the list of supported as something closest
to 640x480 instead of the lowest possible resolution.
Applications that don't suggest a default size will then get something
more sensible than a poster frame.
The graph cycle goes: driver timeout -> process output nodes -> process
driver node. Hence, driver should produce buffers in the timeout,
otherwise there's one quantum extra latency.
Make the bluez5 media/sco sources as drivers put a buffer to io before
indicating ready, and as follower do it in process. Also make checks if
io == NULL, and don't set io->status to HAVE_DATA unless there really is
a buffer ready.
Introduce `unregister_media_endpoint()` to unregister the specified
media endpoint object from dbus. Moreover, move the logic that decides
whether or not a particular codec should be registered for a given
direction into `endpoint_should_be_registered()` and use that in
both `(un)register_media_endpoint()`.
Driver timeouts need to be started/stopped when we switch from follower
to driver or vice versa.
The BT sources fail to do this, so fix it. Sinks already do it right.
Use separate timers for driving graph and for flushing, since they don't
have the same period.
Flushing is done based on the time positions of the next sample to be
written, so it will stay in sync with the graph. Because writing too
much data to SCO sockets generally only causes the device to skip ahead,
we don't need to handle the case where the writing has been lagging.
This fixes simultaneous playback to both ALSA and SCO sinks from the
same graph, with SCO as driver, which previously produces broken sound
(e.g. with pw-play --latency 512, linked to the two sinks) ALSA nodes
require regular driver intervals, which was not true previously.
Bigger buffer allows for more fluctuation in transmission rate without
sound glitches.
It doesn't matter much for latency, as under normal conditions we are
not producing data faster than the BT adapter can transmit, so the
buffer generally is almost always empty or full, and in the latter case
we have to reduce the bitrate.
Do not start the playback device until there is data to play. Otherwise
time consuming configuration of other nodes (such as setting hw params
of a capture device) may be done after playback has been started, which
may cause xrun.
