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>
For OPUS codec, FFmpeg expects the raw extradata, but MediaCodec wraps
it in some structure.
Fix the config packet to send only the raw extradata.
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
Remove USER_ID from ServiceManager, and replace it by a constant in
FakeContext.
This is the same as android.os.Process.ROOT_UID, but this constant has
been introduced in API 29.
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
The Callbacks interface notifies new packets. But in addition, the
screen encoder will need to write headers on start.
We could add a function onStart(), but for simplicity, just remove the
interface, which brings no value, and call the streamer directly.
Refs 87972e2022
On any error, all previously opened sockets must be closed.
Handle these errors in a single catch-block. Currently, there are only 2
sockets, but this will simplify even more with more sockets.
Note: this commit is better displayed with --ignore-space-change (-b).
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.
User-friendly error messages are printed on specific configuration
exceptions. In that case, do not print the stacktrace.
Also handle the user-friendly error message directly where the error
occurs, and print multiline messages in a single log call, to avoid
confusing interleaving.
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>
As reported by gradle:
> Setting the namespace via a source AndroidManifest.xml's package
> attribute is deprecated.
>
> Please instead set the namespace (or testNamespace) in the module's
> build.gradle file, as described here:
> https://developer.android.com/studio/build/configure-app-module#set-namespace
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
DesktopConnection implements Closeable, so it is implicitly closed after
its try-with-resources block. Closing the DesktopConnection shutdowns
the sockets, so it is necessary in particular to wake up blocking read()
calls from the controller.
But the controller thread was joined before the DesktopConnection was
closed, causing a deadlock. To fix the problem, join the controller
thread only after the DesktopConnection is closed.
Refs 400a1c69b1
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.
On close, the client closes the socket. This wakes up socket blocking
calls on the server-side, by throwing an exception. Since this exception
is expected, it was not logged.
However, other IOExceptions might occur, which must not be ignored. For
that purpose, log only IOException when they are not caused by an EPIPE
error.
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.
MediaCodec may fail spuriously, typically when stopping an encoding and
starting a new one immediately (for example on device rotation).
In that case, retry a few times, in many cases it should work.
Refs #3693 <https://github.com/Genymobile/scrcpy/issues/3693>
Romain Vimont
