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.
Co-authored-by: Simon Chan <1330321+yume-chan@users.noreply.github.com>
If there is exactly one producer, then it can assume that the remaining
space in the buffer will only increase until it write something.
This assumption may allow the producer to write to the buffer (up to a
known safe size) without any synchronization mechanism, thus allowing
to read and write different parts of the buffer in parallel.
The producer can then commit the write with lock held, and update its
knowledge of the safe empty remaining space.
When audio capture fails on the device, scrcpy continue mirroring the
video stream. This allows to enable audio by default only when
supported.
However, if an audio configuration occurs (for example the user
explicitly selected an unknown audio encoder), this must be treated as
an error and scrcpy must exit.
If no bit-rate is passed, let the server use the default value (8Mbps).
This avoids to define a default value on both sides, and to pass the
default bit-rate as an argument when starting the server.
By default, audio is enabled (--no-audio must be explicitly passed to
disable it).
However, some devices may not support audio capture (typically devices
below Android 11, or Android 11 when the shell application is not
foreground on start).
In that case, make the server notify the client to dynamically disable
audio forwarding so that it does not wait indefinitely for an audio
stream.
Also disable audio on unknown codec or missing decoder on the
client-side, for the same reasons.
For video streams (at least H.264 and H.265), the config packet
containing SPS/PPS must be prepended to the next packet (the following
keyframe).
For audio streams (at least OPUS), they must not be merged.
When audio is enabled, open a new socket to send the audio stream from
the device to the client.
Co-authored-by: Romain Vimont <rom@rom1v.com>
Signed-off-by: Romain Vimont <rom@rom1v.com>
Audio will be enabled by default (when supported). Add an option to
disable it.
Co-authored-by: Romain Vimont <rom@rom1v.com>
Signed-off-by: Romain Vimont <rom@rom1v.com>
The recorder opened the target file from the packet sink open()
callback, called by the demuxer. Only then the recorder thread was
started.
One golden rule for the recorder is to never block the demuxer for I/O,
because it would impact mirroring. This rule is respected on recording
packets, but not for the initial recorder opening.
Therefore, start the recorder thread from sc_recorder_init(), open the
file immediately from the recorder thread, then make it wait for the
stream to start (on packet sink open()).
Now that the recorder can report errors directly (rather than making the
demuxer call fail), it is possible to report file opening error even
before the packet sink is open.
The recorder has two initialization phases: one to initialize the
concrete recorder object, and one to open its packet_sink trait.
Initialize mutex and condvar as part of the object initialization.
If there were several packet_sink traits, the mutex and condvar would
still be initialized only once.
Stop scrcpy on recorder errors.
It was previously indirectly stopped by the demuxer, which failed to
push packets to a recorder in error. Report it directly instead:
- it avoids to wait for the next demuxer call;
- it will allow to open the target file from a separate thread and stop
immediately on any I/O error.
