Take active rate correction properly into account when dropping data on
overrun resync.
Drop data only for the currently processed stream, after data has been
consumed from it. Make sure the rate correction factor is updated after
this for the next cycle of the stream.
Also fix buffer fill level calculation: the fill level interpolation
should use node rate corr, not clock rate diff, since the calculations
are done in system clock domain. Fix same issue in fractional delay
calculation, and take no resampler prefill into account.
Later, we maybe need some more resampler APIs to avoid such details
leaking in.
Previously, stream could have its old rate correction locked in, and its
fill level would then end up off the target on the next cycle.
The rate matching filter assumes buffer level for cycle j+1 is
buffer(j+1) = buffer(j) + recv(j) - corr(j+1) * duration
but what we are actually doing is instead
buffer(j+1) = buffer(j) + recv(j) - corr(j-1) * duration
because the correction factor that is computed is not used for the next
cycle, but the one following that. Although the filter is still stable
in theory the extra lag causes oscillations to be damped less.
Fix by using the computed correction factor for the next cycle, as
there's no reason why we'd like to have more lag in rate matching.
This then changes c(j-1) -> c(j) in the assumptions, which turns out to
fix the situation. Fix the filter derivation to match. The filter
coefficients stay as they were, and they are actually exactly correct
also for short averaging times.
In practice, it is observed that ISO RX with quantum 4096 converges to
stable rate, whereas previously the matching retained small
oscillations.
The buffer level number includes the current quantum, so it should not
be subtracted. We do this after recovery from glitch, and this throws
rate matching off.
The level after recovery should also include the resampler delay.
Take resampler delay into account when computing the buffer fill level,
including the fractional part.
If decode-buffer is now fed nominal packet reference times in
write_packet(), it converges the total buffer + resampler latency to the
target at sub-sample accuracy.
This is needed for aligning RX of ISO streams in the same group, so that
e.g. stereo pair alignment is achieved even though the streams have
separate resamplers. Resampler phases get aligned via independent rate
matching.
Use kernel BT_PKT_SEQNUM (likely in Linux v6.17) to provide the ISO
packets sequence numbers. Fall back to counting packets if kernel is too
old to support the feature.
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).
Use kernel-provided packet reception timestamps to get less jitter in
packet timings. Mostly matters for ISO/SCO which have regular schedule.
A2DP (L2CAP) doesn't currently do RX timestamps in kernel, but we can as
well use the same mechanism for it.
Interpolate buffer level to current playback position, and change its
definition so it directly corresponds to the total buffer latency. This
is also a bit simpler.
Now that BlueZ supports delay reporting in A2DP sink role, implement
that.
Report value that gives the total latency between packet reception and
audio rendering.
Also make Latency parameter in media-source to be not just a dummy
value.
Use TX timestamps to get accurate reading of queue length and latency on
kernel + controller side.
This is new kernel BT feature, so requires kernel with the necessary
patches, available currently only in bluetooth-next/master branch.
Enabling Poll Errqueue kernel experimental Bluetooth feature is also
required for this.
Use the latency information to mitigate controller issues where ISO
streams are desynchronized due to tx problems or spontaneously when some
packets that should have been sent are left sitting in the queue, and
transmission is off by a multiple of the ISO interval. This state is
visible in the latency information, so if we see streams in a group have
persistently different latencies, drop packets to resynchronize them.
Also make corrections if the kernel/controller queues get too long, so
that we don't have too big latency there.
Since BlueZ watches the same socket for errors, and TX timestamps arrive
via the socket error queue, we need to set BT_POLL_ERRQUEUE in addition
to SO_TIMESTAMPING so that BlueZ doesn't think TX timestamps are errors.
Link: https://github.com/bluez/bluez/issues/515
Link: https://lore.kernel.org/linux-bluetooth/cover.1710440392.git.pav@iki.fi/
Link: https://lore.kernel.org/linux-bluetooth/f57e065bb571d633f811610d273711c7047af335.1712499936.git.pav@iki.fi/
The maximum receive buffer target of 6 packets may be too small when
there's huge jitter in reception. Increase it so that we may use all
buffer available if needed (2*quantum_limit = 370 ms @ 44100).
For SCO, explicitly set maximum buffer to 40 ms, so that latency cannot
grow too large there. For A2DP duplex, set it to 80 ms for same reason.
These are close to the old 6*packet limit.
For BAP server audio sink, set buffering target so that we try to match
the target presentation delay. Also adjust requested node latency to be
smaller than the delay.
Also fix BAP transport presentation delay value parsing, and parse also
the other BAP transport properties. Of these, transport latency value
needs to be taken into account in the total sink latency.
If no packets have been received and spa_bt_decode_buffer_process is
called, this->packet_size.max == INT32_MIN, which can give overflows.
Guard against this condition, although it should be harmless.
a2dp-source as driver does not produce regularly spaced graph cycles,
because A2DP is not isochronous. This causes e.g. crackling for alsa
etc. that expect regular timings. It also does not rate match.
Change a2dp-source to trigger graph on regular intervals. Change recv to
only accumulate data to a buffer, and put data to buffers in process().
Rate match with DLL, keeping average buffer level constant. Keep track
of jitter to determine a safe target value.
