A video buffer had 2 responsibilities:
- handle the frame delaying mechanism (queuing packets and pushing them
after the expected delay);
- keep only the most recent frame (using a sc_frame_buffer).
In order to be able to reuse only the frame delaying mechanism, extract
it to a separate component, sc_delay_buffer.
The video_buffer thread clears the queue once it is stopped, but new
frames might still be pushed asynchronously.
To avoid the problem, do not push any frame once the video_buffer is
stopped.
The packets queued for buffering were wrapped in a dynamically allocated
structure with a "next" field.
To avoid this additional layer of allocation and indirection, use a
VecDeque.
The packets queued for recording were wrapped in a dynamically allocated
structure with a "next" field.
To avoid this additional layer of allocation and indirection, use a
VecDeque.
Since in scrcpy a video packet passed to avcodec_send_packet() is always
a complete video frame, it is sufficient to call avcodec_receive_frame()
exactly once.
In practice, it also works for audio packets: the decoder produces
exactly 1 frame for 1 input packet.
In theory, it is an implementation detail though, so
avcodec_receive_frame() should be called in a loop.
By default, scrcpy mirrors only the video when audio capture fails on
the device. Add an option to force scrcpy to fail if audio is enabled
but does not work.
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
If there is exactly one producer, then it can assume that the remaining
space in the buffer will only increase until it writes 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 a lock held, and update its
knowledge of the safe empty remaining space.
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
When audio capture fails on the device, scrcpy continues 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.
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
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.
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
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.
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
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.
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
The client does not use the audio stream if there is no display and no
recording (i.e. only V4L2), so disable audio so that the device does not
attempt to capture it.
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
When audio is enabled, open a new socket to send the audio stream from
the device to the client.
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
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.
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
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 (spoiler: one for video, one
for audio), then 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.
On the scrcpy-deps repo, I built FFmpeg 5.1.2 binaries for Windows with
only the features used by scrcpy.
For comparison, here are the sizes of the dll for FFmpeg 5.1.2:
- before: 89M
- after: 4.7M
It also allows to upgrade the old FFmpeg version (4.3.1) used for win32.
Refs <https://github.com/rom1v/scrcpy-deps>
Refs <https://github.com/Genymobile/scrcpy/issues/1753>
Running scrcpy --tcpip on a device already connected via TCP/IP did not
initialize server->serial.
As a consequence, in debug mode, an assertion failed:
scrcpy: ../app/src/server.c:770: run_server: Assertion
`server->serial' failed.
In release mode, scrcpy failed with this error:
adb: -s requires an argument
Scrcpy does not use FFmpeg network features. Initialize network locally
instead (useful only for Windows).
The include block has been moved to fix the following warning:
Please include winsock2.h before windows.h
When a call to a packet or frame sink fails, do not log the error on the
caller side: either the "failure" is expected (explicitly stopped) or it
must be logged by the packet or frame sink implementation.
The PTS received from MediaCodec are expressed relative to an arbitrary
clock origin. We consider the PTS of the first frame to be 0, and the
PTS of every other frame is relative to this first PTS (note that the
PTS is only used for recording, it is ignored for mirroring).
For simplicity, this relative PTS was computed on the server-side.
To prepare support for multiple stream (video and audio), send the
packet with its original PTS, and handle the PTS offset on the
client-side (by the recorder).
Since we can't know in advance which stream will produce the first
packet with the lowest PTS (a packet received later on one stream may
have a PTS lower than a packet received earlier on another stream),
computing the PTS on the server-side would require unnecessary waiting.
