On some systems, the SDL audio callback is not called frequently enough
(for example it requests 5ms of samples every 10ms), because the output
buffer is too small.
By default, we want to use a small value (5ms) to minimize latency and
buffer underrun, but if it does not work well, users need a way to
increase it.
Refs #3793 <https://github.com/Genymobile/scrcpy/issues/3793>
An int was compared with an unsigned:
../app/src/audio_player.c:290:27: warning: comparison of integers of
different signs: 'int' and 'unsigned int' [-Wsign-compare]
if (abs(diff) < ap->sample_rate / 1000) {
~~~~~~~~~ ^ ~~~~~~~~~~~~~~~~~~~~~~
On buffer underflow, the average buffering must be updated, but it is
intended to be accessed only from the receiver thread.
Make the player and the receiver thread communicate the underflow via a
new field (ap->underflow).
Play the decoded audio using SDL.
The audio player frame sink receives the audio frames, resample them
and write them to a byte buffer (introduced by this commit).
On SDL audio callback (from an internal SDL thread), copy samples from
this byte buffer to the SDL audio buffer.
The byte buffer is protected by the SDL_AudioDeviceLock(), but it has
been designed so that the producer and the consumer may write and read
in parallel, provided that they don't access the same slices of the
ring-buffer buffer.
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
Co-authored-by: Simon Chan <1330321+yume-chan@users.noreply.github.com>
shuax
Romain Vimont