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115 Commits

Author SHA1 Message Date
Romain Vimont
20226308d6 Warn on ignored audio options
For raw audio codec, some audio options are ignored.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:28:20 +01:00
Romain Vimont
bf6772715c Add --audio-codec=raw option
Add support for raw (PCM S16 LE) audio codec (a raw decoder is included
in FFmpeg).

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
4bfb27b197 Add raw audio recorder
Add an alternative AudioRecorder to stream raw packets without encoding.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
b08684709a Extract audio recorder interface
In order to support both encoded and raw audio stream, extract a
interface (very minimal, but sufficient to just start and stop).

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
f065fee916 Extract audio capture
The audio capture was implemented in AudioEncoder.

In order to reuse it without encoding, extract it to a separate class.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
0259d69968 Stop on decoder frame push error
On push, frame sinks report downstream errors to stop upstream
components. Do not ignore the error.
2023-03-06 09:26:29 +01:00
Romain Vimont
61b9b41e80 Add --audio-buffer
Expose an option to add a buffering delay (in milliseconds) before
playing audio.

This is similar to the options --display-buffer and --v4l2-buffer for
video frames.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
5828b85138 Optionally do not delay the first frame
A delay buffer delayed all the frames except the first one, to open the
scrcpy window immediately and get a picture.

Make this feature optional, so that the delay buffer might also be used
for audio.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
296ea22bbf Accept clock estimation with a single point
If there is only one point, assume the slope is 1.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
2f0f4649e8 Use delay buffer as a frame source/sink
The components needing delayed frames (sc_screen and sc_v4l2_sink)
managed a sc_video_buffer instance, which itself embedded a
sc_frame_buffer instance (to keep only the most recent frame).

In theory, these components should not be aware of delaying: they should
just receive AVFrames later, and only handle a sc_frame_buffer.

Therefore, refactor sc_delay_buffer as a frame source (it consumes)
frames) and a frame sink (it produces frames, after some delay), and
plug an instance in the pipeline only when a delay is requested.

This also removes the need for a specific sc_video_buffer.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
d835a7a2e6 Use frame source trait in decoder 2023-03-06 09:26:29 +01:00
Romain Vimont
51b89a53fb Introduce frame source trait
There was a frame sink trait, implemented by components able to receive
AVFrames, but each frame source had to manually send to frame sinks.

In order to mutualise sink management, add a frame sink trait.
2023-03-06 09:26:29 +01:00
Romain Vimont
603b2158ed Use packet source trait in demuxer 2023-03-06 09:26:29 +01:00
Romain Vimont
0784f266a8 Introduce packet source trait
There was a packet sink trait, implemented by components able to
receive AVPackets, but each packet source had to manually send to packet
sinks.

In order to mutualise sink management, add a packet source trait.
2023-03-06 09:26:29 +01:00
Romain Vimont
6c11c9510d Extract sc_delay_buffer
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 reuse only the frame delaying mechanism, extract it to a
separate component, sc_delay_buffer.
2023-03-06 09:26:29 +01:00
Romain Vimont
1ebfe0efce Report video buffer downstream errors
Make the video buffer stop if its consumer could not receive a frame.
2023-03-06 09:26:29 +01:00
Romain Vimont
631dc6f357 Stop the video buffer on error
If an error occurs from the video buffer thread (typically an
out-of-memory error), then stop.
2023-03-06 09:26:29 +01:00
Romain Vimont
e9984266fa Fix possible race condition on video_buffer end
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.
2023-03-06 09:26:29 +01:00
Romain Vimont
5f2cf1e8c0 Remove sc_queue
All uses have been replaced by VecDeque.
2023-03-06 09:26:29 +01:00
Romain Vimont
6e00a55f03 Remove cbuf
All uses have been replaced by VecDeque.
2023-03-06 09:26:29 +01:00
Romain Vimont
92ca8c83d6 Use VecDeque in aoa_hid
Replace cbuf by VecDeque in aoa_hid
2023-03-06 09:26:29 +01:00
Romain Vimont
b2997fe398 Use VecDeque in file_pusher
Replace cbuf by VecDeque in file_pusher.

As a side-effect, the new implementation does not limit the queue to an
arbitrary value.
2023-03-06 09:26:29 +01:00
Romain Vimont
92f17f1e53 Use VecDeque in controller
Replace cbuf by VecDeque in controller.
2023-03-06 09:26:29 +01:00
Romain Vimont
4b18089c6c Use VecDeque in video_buffer
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.
2023-03-06 09:26:29 +01:00
Romain Vimont
4cfd04adf7 Use VecDeque in recorder
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.
2023-03-06 09:26:29 +01:00
Romain Vimont
faa2f22cd6 Introduce VecDeque
Introduce a double-ended queue implemented with a growable ring buffer.

Inspired from the Rust VecDeque type:
<https://doc.rust-lang.org/std/collections/struct.VecDeque.html>
2023-03-06 09:26:29 +01:00
Romain Vimont
cef9bf51a9 Add sc_allocarray() util
Add a function to allocate an array, which fails safely in the case
where the multiplication would overflow.
2023-03-06 09:26:29 +01:00
Romain Vimont
c30bb893f8 Use reallocarray() in sc_vector
This fails safely in case of overflow.
2023-03-06 09:26:29 +01:00
Romain Vimont
35debac4d1 Add compat for reallocarray()
This function fails safely in the case where the multiplication would
overflow.
2023-03-06 09:26:29 +01:00
Romain Vimont
2dd75bf88a Call avcodec_receive_frame() in a loop
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.
2023-03-06 09:26:29 +01:00
Romain Vimont
a8895b06d2 Add --require-audio
By default, scrcpy mirrors only the video when audio capture fails on
the device. Add a flag to force scrcpy to fail if audio is enabled but
does not work.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
427245ee20 Add compat support for FFmpeg < 5.1
The new chlayout API has been introduced in FFmpeg 5.1. Use the old
channel_layout API on older versions.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Simon Chan
2a6df62ce9 Add workaround to capture audio on Android 11
On Android 11, it is possible to start the capture only when the running
app is in foreground. But scrcpy is not an app, it's a Java application
started from shell.

As a workaround, start an existing Android shell existing activity just
to start the capture, then close it immediately.

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>
2023-03-06 09:26:29 +01:00
Romain Vimont
30d1244962 Add audio player
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>
2023-03-06 09:26:29 +01:00
Romain Vimont
4e6941e632 Add two-step write feature to bytebuf
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.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
85f069b39b Introduce bytebuf util
Add a ring-buffer for bytes. It will be useful for buffering audio.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
77eb14f6e1 Pass AVCodecContext to frame sinks
Frame consumers may need details about the frame format.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
8025238654 Add an audio decoder
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
f57b62c286 Give a name to decoder instances
This will be useful in logs.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
b994288afa Rename decoder to video_decoder
This prepares the introduction of audio_decoder.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
fe396544a6 Log display sizes in display list
This is more convenient than just the display id alone.
2023-03-06 09:26:29 +01:00
Romain Vimont
e5a5d4b9cc Add --list-displays 2023-03-06 09:26:29 +01:00
Romain Vimont
5c7af685a4 Move log message helpers to LogUtils
This class will also contain other log helpers.
2023-03-06 09:26:29 +01:00
Romain Vimont
b93a8b911b Quit on audio configuration failure
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.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
afbeeeb678 Add --list-encoders
Add an option to list the device encoders properly.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
f5f2edd4b9 Move await_for_server() logs
Print the logs on the caller side. This will allow to call the function
in another context without printing the logs.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
dce8cf5108 Add --audio-encoder
Similar to --video-encoder, but for audio.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
4768d3f8b8 Extract unknown encoder error message
This will allow to reuse the same code for audio encoder selection.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
d558c1c8e7 Add --audio-codec-options
Similar to --video-codec-options, but for audio.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
3cd76e0bac Extract application of codec options
This will allow to reuse the same code for audio codec options.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
0cd6e4d5dd Add support for AAC audio codec
Add option --audio-codec=aac.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
56c65bd4c6 Add --audio-codec
Introduce the selection mechanism. Alternative codecs will be added
later.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
78dcc01dcf Add --audio-bit-rate
Add an option to configure the audio bit-rate.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
b9e0803806 Disable MethodLength checkstyle on createOptions()
This method will grow as needed to initialize options.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
879f086e6a Rename --encoder to --video-encoder
This prepares the introduction of --audio-encoder.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
0828631b16 Rename --codec-options to --video-codec-options
This prepares the introduction of --audio-codec-options.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
7653158e5a Rename --bit-rate to --video-bit-rate
This prepares the introduction of --audio-bit-rate.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
42082f622f Rename --codec to --video-codec
This prepares the introduction of --audio-codec.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
5a02005fc3 Remove default bit-rate on client side
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>
2023-03-06 09:26:29 +01:00
Romain Vimont
8c680d391d Record at least video packets on stop
If the recorder is stopped while it has not received any audio packet
yet, make sure the video stream is correctly recorded.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
a9ea8b7c1f Disable audio before Android 11
The permission "android.permission.RECORD_AUDIO" has been added for
shell in Android 11.

Moreover, on lower versions, it may make the server segfault on the
device (happened on a Nexus 5 with Android 6.0.1).

Refs <4feeee8891%5E%21/>
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
4be7925667 Disable audio on initialization error
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>
2023-03-06 09:26:29 +01:00
Romain Vimont
43d6d0d4bc Add record audio support
Make the recorder accept two input sources (video and audio), and mux
them into a single file.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
7a4fe1e8f6 Rename video-specific variables in recorder
This paves the way to add audio-specific variables.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
75cd0ea3b5 Do not merge config audio packets
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>
2023-03-06 09:26:29 +01:00
Romain Vimont
1f9523dd67 Add an audio demuxer
Add a demuxer which will read the stream from the audio socket.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
f0b74e2ed8 Force --no-audio if no display and no recording
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>
2023-03-06 09:26:29 +01:00
Romain Vimont
6da741177f Give a name to demuxer instances
This will be useful in logs.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
f8231417aa Rename demuxer to video_demuxer
There will be another demuxer instance for audio.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
f230db9476 Extract OPUS extradata
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>
2023-03-06 09:26:29 +01:00
Romain Vimont
24a904800f Use a streamer to send the audio stream
Send each encoded audio packet using a streamer.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:26:29 +01:00
Romain Vimont
9bc0998c09 Encode recorded audio on the device
For now, the encoded packets are just logged into the console.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-06 09:25:46 +01:00
Romain Vimont
cd7bdabc84 Make streamer more generic
Expose a method to write a packet from raw metadata (without
BufferInfo).
2023-03-05 21:08:42 +01:00
Simon Chan
901fdc6bf6 Capture device audio
Create an AudioRecorder to capture the audio source REMOTE_SUBMIX.

For now, the captured packets are just logged into the console.

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>
2023-03-05 18:13:25 +01:00
Simon Chan
2d7630882a Add a new socket for audio stream
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>
2023-03-05 18:13:25 +01:00
Simon Chan
0457253655 Add --no-audio option
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>
2023-03-03 21:43:57 +01:00
Romain Vimont
484f38be1c Use FakeContext for Application instance
This will expose the correct package name and UID to the application
context.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:43:54 +01:00
Romain Vimont
013bf96cd0 Use shell package name for workarounds
For consistency.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:43:50 +01:00
Romain Vimont
3c090b3d9e Use ROOT_UID from FakeContext
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>
2023-03-03 21:43:44 +01:00
Romain Vimont
31068ee607 Use PACKAGE_NAME from FakeContext
Remove duplicated constant.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:43:41 +01:00
Romain Vimont
e23366fb4e Use AttributionSource from FakeContext
FakeContext already provides an AttributeSource instance.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>

Co-authored-by: Simon Chan <1330321+yume-chan@users.noreply.github.com>
2023-03-03 21:43:33 +01:00
Simon Chan
b5ae5bf6bb Add a fake Android Context
Since scrcpy-server is not an Android application (it's a java
executable), it has no Context.

Some features will require a Context instance to get the package name
and the UID. Add a FakeContext for this purpose.

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>
2023-03-03 21:43:11 +01:00
Romain Vimont
e068fe43cf Improve error message for unknown encoder
The provided encoder name depends on the selected codec. Improve the
error message and the suggestions.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:43:06 +01:00
Romain Vimont
883a998c10 Rename "codec" variable to "mediaCodec"
This will allow to use "codec" for the Codec type.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:43:04 +01:00
Romain Vimont
3c670dc52a Make streamer independent of codec type
Rename VideoStreamer to Streamer, and extract a Codec interface which
will also support audio codecs.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:42:47 +01:00
Romain Vimont
af1f00bece Pass all args to ScreenEncoder constructor
There is no good reason to pass some of them in the constructor and some
others as parameters of the streamScreen() method.
2023-03-03 21:09:21 +01:00
Romain Vimont
7853c4c303 Move screen encoder initialization
This prepares further refactors.
2023-03-03 21:09:21 +01:00
Romain Vimont
bbb025bcf3 Write streamer header from ScreenEncoder
The screen encoder is responsible to write data to the video streamer.
2023-03-03 21:09:21 +01:00
Romain Vimont
9e4d7f59d7 Use VideoStreamer directly from ScreenEncoder
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
2023-03-03 21:09:21 +01:00
Romain Vimont
89c3de5498 Simplify error handling on socket creation
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).
2023-03-03 21:09:21 +01:00
Romain Vimont
0d8644d3ff Reorder initialization
Initialize components in the pipeline order: demuxer first, decoder and
recorder second.
2023-03-03 21:09:21 +01:00
Romain Vimont
f0660df102 Refactor recorder logic
Process the initial config packet (necessary to write the header)
separately.
2023-03-03 21:09:21 +01:00
Romain Vimont
ddd9c8b4a8 Move last packet recording
Write the last packet at the end.
2023-03-03 21:09:21 +01:00
Romain Vimont
ed14c56be4 Add start() function for recorder
For consistency with the other components, do not start the internal
thread from an init() function.
2023-03-03 21:09:21 +01:00
Romain Vimont
0af71d2bd8 Open recording file from the recorder thread
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.
2023-03-03 21:09:21 +01:00
Romain Vimont
e1deb7077c Inline packet_sink impl in recorder
Remove useless wrappers.
2023-03-03 21:09:21 +01:00
Romain Vimont
92b5c297b4 Initialize recorder fields from init()
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.
2023-03-03 21:09:21 +01:00
Romain Vimont
92bc1a37ae Report recorder errors
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.
2023-03-03 21:09:21 +01:00
Romain Vimont
37c9c3cb50 Move previous packet to a local variable
It is only used from run_recorder().
2023-03-03 21:09:21 +01:00
Romain Vimont
83c20a10db Move pts_origin to a local variable
It is only used from run_recorder().
2023-03-03 21:09:21 +01:00
Romain Vimont
c6cd4ff8fe Change PTS origin type from uint64_t to int64_t
It is initialized from AVPacket.pts, which is an int64_t.
2023-03-03 21:09:21 +01:00
Romain Vimont
8eef96012b Fix --encoder documentation
Mention that it depends on the codec provided by --codec (which is not
necessarily H264 anymore).
2023-03-03 21:09:21 +01:00
Romain Vimont
3619efa7d4 Do not print stacktraces when unnecessary
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.
2023-03-03 21:09:21 +01:00
Romain Vimont
4fecf4e49e Fix --no-clipboard-autosync bash completion
Fix typo.
2023-03-03 21:09:21 +01:00
Romain Vimont
3015224135 Split server stop() and join()
For consistency with the other components, call stop() and join()
separately.

This allows to stop all components, then join them all.
2023-03-03 21:09:21 +01:00
Romain Vimont
2381a61798 Print FFmpeg logs
FFmpeg logs are redirected to a specific SDL log category.

Initialize the log level for this category to print them as expected.
2023-03-03 21:09:21 +01:00
Romain Vimont
3ab2840fd5 Move FFmpeg callback initialization
Configure FFmpeg log redirection on start from a log helper.
2023-03-03 21:09:21 +01:00
Romain Vimont
2bdee9b31c Upgrade FFmpeg custom builds for Windows
Use a build which includes the pcm_s16le decoder, to support RAW audio.

Refs <https://github.com/rom1v/scrcpy-deps/commits/6.0-scrcpy-2>
2023-03-03 21:09:21 +01:00
Romain Vimont
8131d1f00e Upgrade FFmpeg (6.0) for Windows
Use the latest version (specifically built for scrcpy).

Refs <https://www.ffmpeg.org/download.html#release_6.0>
2023-03-03 21:09:20 +01:00
Romain Vimont
17b9d149cf Use minimal prebuilt FFmpeg for Windows
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>
2023-03-03 21:08:42 +01:00
Romain Vimont
c8b0e3682c Simplify libusb prebuilt scripts
In theory, include/ might be slightly different for win32 and win64
builds. Use each one separately to simplify.
2023-03-03 21:06:03 +01:00
Romain Vimont
14a85fd61e Silence lint warning about constant in API 29
MediaFormat.MIMETYPE_VIDEO_AV1 has been added in API 29, but it is not
a problem to inline the constant in older versions.
2023-03-03 11:13:48 +01:00
Romain Vimont
5bf52a98ed Remove manifest package name
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
2023-03-03 11:13:48 +01:00
Romain Vimont
a252194161 Upgrade gradle build tools to 7.4.0
Plugin version 7.4.0.
Gradle version 7.5.

Refs <https://developer.android.com/studio/releases/gradle-plugin#updating-gradle>
2023-03-03 11:13:48 +01:00
Romain Vimont
b5d41ad4f6 Fix useless garbage initialization
The variable `p` was initialized with a garbage value (a `const char **`
casted to `char *`). Fortunately, it was never read.

Refs <https://github.com/Genymobile/scrcpy/issues/3765>
2023-03-03 11:12:31 +01:00
27 changed files with 422 additions and 302 deletions

View File

@@ -718,7 +718,7 @@ scrcpy --display=1
The list of display ids can be retrieved by:
```bash
adb shell dumpsys display # search "mDisplayId=" in the output
scrcpy --list-displays
```
The secondary display may only be controlled if the device runs at least Android

View File

@@ -45,6 +45,7 @@ _scrcpy() {
-r --record=
--record-format=
--render-driver=
--require-audio
--rotation=
-s --serial=
--shortcut-mod=
@@ -77,7 +78,7 @@ _scrcpy() {
return
;;
--audio-codec)
COMPREPLY=($(compgen -W 'opus aac' -- "$cur"))
COMPREPLY=($(compgen -W 'raw opus aac' -- "$cur"))
return
;;
--lock-video-orientation)

View File

@@ -10,7 +10,7 @@ local arguments
arguments=(
'--always-on-top[Make scrcpy window always on top \(above other windows\)]'
'--audio-bit-rate=[Encode the audio at the given bit-rate]'
'--audio-codec=[Select the audio codec]:codec:(opus aac)'
'--audio-codec=[Select the audio codec]:codec:(raw opus aac)'
'--audio-codec-options=[Set a list of comma-separated key\:type=value options for the device audio encoder]'
'--audio-encoder=[Use a specific MediaCodec audio encoder]'
{-b,--video-bit-rate=}'[Encode the video at the given bit-rate]'
@@ -51,6 +51,7 @@ arguments=(
{-r,--record=}'[Record screen to file]:record file:_files'
'--record-format=[Force recording format]:format:(mp4 mkv)'
'--render-driver=[Request SDL to use the given render driver]:driver name:(direct3d opengl opengles2 opengles metal software)'
'--require-audio=[Make scrcpy fail if audio is enabled but does not work]'
'--rotation=[Set the initial display rotation]:rotation values:(0 1 2 3)'
{-s,--serial=}'[The device serial number \(mandatory for multiple devices only\)]:serial:($("${ADB-adb}" devices | awk '\''$2 == "device" {print $1}'\''))'
'--shortcut-mod=[\[key1,key2+key3,...\] Specify the modifiers to use for scrcpy shortcuts]:shortcut mod:(lctrl rctrl lalt ralt lsuper rsuper)'

View File

@@ -136,26 +136,19 @@ else
ffmpeg_bin_dir = meson.current_source_dir() + '/prebuilt-deps/data/' + prebuilt_ffmpeg + '/bin'
ffmpeg_include_dir = 'prebuilt-deps/data/' + prebuilt_ffmpeg + '/include'
# ffmpeg versions are different for win32 and win64 builds
ffmpeg_avcodec = meson.get_cross_property('ffmpeg_avcodec')
ffmpeg_avformat = meson.get_cross_property('ffmpeg_avformat')
ffmpeg_avutil = meson.get_cross_property('ffmpeg_avutil')
ffmpeg_swresample = meson.get_cross_property('ffmpeg_swresample')
ffmpeg = declare_dependency(
dependencies: [
cc.find_library(ffmpeg_avcodec, dirs: ffmpeg_bin_dir),
cc.find_library(ffmpeg_avformat, dirs: ffmpeg_bin_dir),
cc.find_library(ffmpeg_avutil, dirs: ffmpeg_bin_dir),
cc.find_library(ffmpeg_swresample, dirs: ffmpeg_bin_dir),
cc.find_library('avcodec-60', dirs: ffmpeg_bin_dir),
cc.find_library('avformat-60', dirs: ffmpeg_bin_dir),
cc.find_library('avutil-58', dirs: ffmpeg_bin_dir),
cc.find_library('swresample-4', dirs: ffmpeg_bin_dir),
],
include_directories: include_directories(ffmpeg_include_dir)
)
prebuilt_libusb = meson.get_cross_property('prebuilt_libusb')
prebuilt_libusb_root = meson.get_cross_property('prebuilt_libusb_root')
libusb_bin_dir = meson.current_source_dir() + '/prebuilt-deps/data/' + prebuilt_libusb
libusb_include_dir = 'prebuilt-deps/data/' + prebuilt_libusb_root + '/include'
libusb_bin_dir = meson.current_source_dir() + '/prebuilt-deps/data/' + prebuilt_libusb + '/bin'
libusb_include_dir = 'prebuilt-deps/data/' + prebuilt_libusb + '/include'
libusb = declare_dependency(
dependencies: [

View File

@@ -1,45 +0,0 @@
#!/usr/bin/env bash
set -e
DIR=$(dirname ${BASH_SOURCE[0]})
cd "$DIR"
. common
mkdir -p "$PREBUILT_DATA_DIR"
cd "$PREBUILT_DATA_DIR"
DEP_DIR=ffmpeg-win32-4.3.1
FILENAME_SHARED=ffmpeg-4.3.1-win32-shared.zip
SHA256SUM_SHARED=357af9901a456f4dcbacd107e83a934d344c9cb07ddad8aaf80612eeab7d26d2
FILENAME_DEV=ffmpeg-4.3.1-win32-dev.zip
SHA256SUM_DEV=230efb08e9bcf225bd474da29676c70e591fc94d8790a740ca801408fddcb78b
if [[ -d "$DEP_DIR" ]]
then
echo "$DEP_DIR" found
exit 0
fi
get_file "https://github.com/Genymobile/scrcpy/releases/download/v1.16/$FILENAME_SHARED" \
"$FILENAME_SHARED" "$SHA256SUM_SHARED"
get_file "https://github.com/Genymobile/scrcpy/releases/download/v1.16/$FILENAME_DEV" \
"$FILENAME_DEV" "$SHA256SUM_DEV"
mkdir "$DEP_DIR"
cd "$DEP_DIR"
ZIP_PREFIX_SHARED=ffmpeg-4.3.1-win32-shared
unzip "../$FILENAME_SHARED" \
"$ZIP_PREFIX_SHARED"/bin/avutil-56.dll \
"$ZIP_PREFIX_SHARED"/bin/avcodec-58.dll \
"$ZIP_PREFIX_SHARED"/bin/avformat-58.dll \
"$ZIP_PREFIX_SHARED"/bin/swresample-3.dll \
"$ZIP_PREFIX_SHARED"/bin/swscale-5.dll
ZIP_PREFIX_DEV=ffmpeg-4.3.1-win32-dev
unzip "../$FILENAME_DEV" \
"$ZIP_PREFIX_DEV/include/*"
mv "$ZIP_PREFIX_SHARED"/* .
mv "$ZIP_PREFIX_DEV"/* .
rmdir "$ZIP_PREFIX_SHARED" "$ZIP_PREFIX_DEV"

View File

@@ -1,36 +0,0 @@
#!/usr/bin/env bash
set -e
DIR=$(dirname ${BASH_SOURCE[0]})
cd "$DIR"
. common
mkdir -p "$PREBUILT_DATA_DIR"
cd "$PREBUILT_DATA_DIR"
VERSION=5.1.2
DEP_DIR=ffmpeg-win64-$VERSION
FILENAME=ffmpeg-$VERSION-full_build-shared.7z
SHA256SUM=d9eb97b72d7cfdae4d0f7eaea59ccffb8c364d67d88018ea715d5e2e193f00e9
if [[ -d "$DEP_DIR" ]]
then
echo "$DEP_DIR" found
exit 0
fi
get_file "https://github.com/GyanD/codexffmpeg/releases/download/$VERSION/$FILENAME" \
"$FILENAME" "$SHA256SUM"
mkdir "$DEP_DIR"
cd "$DEP_DIR"
ZIP_PREFIX=ffmpeg-$VERSION-full_build-shared
7z x "../$FILENAME" \
"$ZIP_PREFIX"/bin/avutil-57.dll \
"$ZIP_PREFIX"/bin/avcodec-59.dll \
"$ZIP_PREFIX"/bin/avformat-59.dll \
"$ZIP_PREFIX"/bin/swresample-4.dll \
"$ZIP_PREFIX"/bin/swscale-6.dll \
"$ZIP_PREFIX"/include
mv "$ZIP_PREFIX"/* .
rmdir "$ZIP_PREFIX"

View File

@@ -0,0 +1,30 @@
#!/usr/bin/env bash
set -e
DIR=$(dirname ${BASH_SOURCE[0]})
cd "$DIR"
. common
mkdir -p "$PREBUILT_DATA_DIR"
cd "$PREBUILT_DATA_DIR"
VERSION=6.0-scrcpy-2
DEP_DIR="ffmpeg-$VERSION"
FILENAME="$DEP_DIR".7z
SHA256SUM=98ef97f8607c97a5c4f9c5a0a991b78f105d002a3619145011d16ffb92501b14
if [[ -d "$DEP_DIR" ]]
then
echo "$DEP_DIR" found
exit 0
fi
get_file "https://github.com/rom1v/scrcpy-deps/releases/download/$VERSION/$FILENAME" \
"$FILENAME" "$SHA256SUM"
mkdir "$DEP_DIR"
cd "$DEP_DIR"
ZIP_PREFIX=ffmpeg
7z x "../$FILENAME"
mv "$ZIP_PREFIX"/* .
rmdir "$ZIP_PREFIX"

View File

@@ -22,13 +22,12 @@ get_file "https://github.com/libusb/libusb/releases/download/v1.0.26/$FILENAME"
mkdir "$DEP_DIR"
cd "$DEP_DIR"
# include/ is the same in all folders of the archive
7z x "../$FILENAME" \
libusb-1.0.26-binaries/libusb-MinGW-Win32/bin/msys-usb-1.0.dll \
libusb-1.0.26-binaries/libusb-MinGW-Win32/include/ \
libusb-1.0.26-binaries/libusb-MinGW-x64/bin/msys-usb-1.0.dll \
libusb-1.0.26-binaries/libusb-MinGW-x64/include/
mv libusb-1.0.26-binaries/libusb-MinGW-Win32/bin MinGW-Win32
mv libusb-1.0.26-binaries/libusb-MinGW-x64/bin MinGW-x64
mv libusb-1.0.26-binaries/libusb-MinGW-x64/include .
mv libusb-1.0.26-binaries/libusb-MinGW-Win32 .
mv libusb-1.0.26-binaries/libusb-MinGW-x64 .
rm -rf libusb-1.0.26-binaries

View File

@@ -33,7 +33,7 @@ Default is 0 (no buffering).
.TP
.BI "\-\-audio\-codec " name
Select an audio codec (opus or aac).
Select an audio codec (raw, opus or aac).
Default is opus.

View File

@@ -62,7 +62,10 @@ sc_audio_player_sdl_callback(void *userdata, uint8_t *stream, int len_int) {
" samples", bytes_to_samples(ap, len - read));
#endif
memset(stream + read, 0, len - read);
ap->underflow += bytes_to_samples(ap, len - read);
// If the first frame has not been received yet, it's not an underflow
if (ap->received) {
ap->underflow += bytes_to_samples(ap, len - read);
}
}
ap->last_consumed = sc_tick_now();
@@ -71,7 +74,7 @@ sc_audio_player_sdl_callback(void *userdata, uint8_t *stream, int len_int) {
static uint8_t *
sc_audio_player_get_swr_buf(struct sc_audio_player *ap, size_t min_samples) {
size_t min_buf_size = samples_to_bytes(ap, min_samples);
if (min_buf_size < ap->swr_buf_alloc_size) {
if (min_buf_size > ap->swr_buf_alloc_size) {
size_t new_size = min_buf_size + 4096;
uint8_t *buf = realloc(ap->swr_buf, new_size);
if (!buf) {
@@ -176,6 +179,7 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
ap->last_consumed = 0;
ap->underflow = 0;
ap->received = 0;
SDL_PauseAudioDevice(ap->device, 0);
@@ -233,7 +237,7 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
size_t samples_written = MIN(ret, dst_nb_samples);
size_t swr_buf_size = samples_to_bytes(ap, samples_written);
#ifndef SC_AUDIO_PLAYER_NDEBUG
LOGI("[Audio] %" SC_PRIsizet " samples written to buffer", samples_written);
LOGD("[Audio] %" SC_PRIsizet " samples written to buffer", samples_written);
#endif
// Since this function is the only writer, the current available space is
@@ -252,8 +256,8 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
if (ap->last_consumed) {
sc_tick now = sc_tick_now();
assert(now >= ap->last_consumed);
extrapolated = (sc_tick_now() - ap->last_consumed) * ap->sample_rate
/ SC_TICK_FREQ;
extrapolated = (now - ap->last_consumed) * ap->sample_rate
/ SC_TICK_FREQ;
}
size_t read_avail = sc_bytebuf_read_available(&ap->buf);
@@ -270,7 +274,7 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
sc_average_push(&ap->avg_buffering, buffering);
#ifndef SC_AUDIO_PLAYER_NDEBUG
LOGD("[AUDIO] buffered_samples=%" SC_PRIsizet
LOGD("[Audio] buffered_samples=%" SC_PRIsizet
" underflow=%" SC_PRIsizet
" extrapolated=%" SC_PRIsizet
" buffering=%f avg_buffering=%f",
@@ -311,10 +315,13 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
size_t avg = sc_average_get(&ap->avg_buffering);
if (avg > SC_TARGET_BUFFERED_SAMPLES) {
size_t diff = SC_TARGET_BUFFERED_SAMPLES - avg;
if (diff < ap->underflow) {
if (ap->underflow > diff) {
// Partially absorb underflow for clock compensation (only keep
// the diff with the target buffering level).
ap->underflow = diff;
ap->underflow -= diff;
} else {
// Totally absorb underflow for clock compensation
ap->underflow = 0;
}
size_t skip_samples = MIN(ap->underflow, buffered_samples);
@@ -333,6 +340,7 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
}
ap->previous_write_avail = sc_bytebuf_write_available(&ap->buf);
ap->received = true;
SDL_UnlockAudioDevice(ap->device);
@@ -344,7 +352,7 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
float avg = sc_average_get(&ap->avg_buffering);
int diff = SC_TARGET_BUFFERED_SAMPLES - avg;
#ifndef SC_AUDIO_PLAYER_NDEBUG
LOGI("[Audio] Average buffering=%f, compensation %d", avg, diff);
LOGD("[Audio] Average buffering=%f, compensation %d", avg, diff);
#endif
// Compensate the diff over 3 seconds (but will be recomputed after
// 1 second)

View File

@@ -45,6 +45,7 @@ struct sc_audio_player {
// Number of silence samples inserted to be compensated
size_t underflow;
bool received;
const struct sc_audio_player_callbacks *cbs;
void *cbs_userdata;

View File

@@ -132,7 +132,7 @@ static const struct sc_option options[] = {
.longopt_id = OPT_AUDIO_CODEC,
.longopt = "audio-codec",
.argdesc = "name",
.text = "Select an audio codec (opus or aac).\n"
.text = "Select an audio codec (raw, opus or aac).\n"
"Default is opus.",
},
{
@@ -1506,6 +1506,10 @@ parse_video_codec(const char *optarg, enum sc_codec *codec) {
static bool
parse_audio_codec(const char *optarg, enum sc_codec *codec) {
if (!strcmp(optarg, "raw")) {
*codec = SC_CODEC_RAW;
return true;
}
if (!strcmp(optarg, "opus")) {
*codec = SC_CODEC_OPUS;
return true;
@@ -1514,7 +1518,7 @@ parse_audio_codec(const char *optarg, enum sc_codec *codec) {
*codec = SC_CODEC_AAC;
return true;
}
LOGE("Unsupported audio codec: %s (expected opus)", optarg);
LOGE("Unsupported audio codec: %s (expected raw, opus or aac)", optarg);
return false;
}
@@ -1912,6 +1916,35 @@ parse_args_with_getopt(struct scrcpy_cli_args *args, int argc, char *argv[],
}
}
if (opts->record_filename && opts->audio_codec == SC_CODEC_RAW) {
LOGW("Recording does not support RAW audio codec");
return false;
}
if (opts->audio_codec == SC_CODEC_RAW) {
if (opts->audio_bit_rate) {
LOGW("--audio-bit-rate is ignored for raw audio codec");
}
if (opts->audio_codec_options) {
LOGW("--audio-codec-options is ignored for raw audio codec");
}
if (opts->audio_encoder) {
LOGW("--audio-encoder is ignored for raw audio codec");
}
}
if (opts->audio_codec == SC_CODEC_RAW) {
if (opts->audio_bit_rate) {
LOGW("--audio-bit-rate is ignored for raw audio codec");
}
if (opts->audio_codec_options) {
LOGW("--audio-codec-options is ignored for raw audio codec");
}
if (opts->audio_encoder) {
LOGW("--audio-encoder is ignored for raw audio codec");
}
}
if (!opts->control) {
if (opts->turn_screen_off) {
LOGE("Could not request to turn screen off if control is disabled");

View File

@@ -23,6 +23,7 @@ sc_demuxer_to_avcodec_id(uint32_t codec_id) {
#define SC_CODEC_ID_H264 UINT32_C(0x68323634) // "h264" in ASCII
#define SC_CODEC_ID_H265 UINT32_C(0x68323635) // "h265" in ASCII
#define SC_CODEC_ID_AV1 UINT32_C(0x00617631) // "av1" in ASCII
#define SC_CODEC_ID_RAW UINT32_C(0x00726177) // "raw" in ASCII
#define SC_CODEC_ID_OPUS UINT32_C(0x6f707573) // "opus" in ASCII
#define SC_CODEC_ID_AAC UINT32_C(0x00616163) // "aac in ASCII"
switch (codec_id) {
@@ -32,6 +33,8 @@ sc_demuxer_to_avcodec_id(uint32_t codec_id) {
return AV_CODEC_ID_HEVC;
case SC_CODEC_ID_AV1:
return AV_CODEC_ID_AV1;
case SC_CODEC_ID_RAW:
return AV_CODEC_ID_PCM_S16LE;
case SC_CODEC_ID_OPUS:
return AV_CODEC_ID_OPUS;
case SC_CODEC_ID_AAC:

View File

@@ -27,6 +27,7 @@ enum sc_codec {
SC_CODEC_H264,
SC_CODEC_H265,
SC_CODEC_AV1,
SC_CODEC_RAW,
SC_CODEC_OPUS,
SC_CODEC_AAC,
};

View File

@@ -169,6 +169,8 @@ sc_server_get_codec_name(enum sc_codec codec) {
return "h265";
case SC_CODEC_AV1:
return "av1";
case SC_CODEC_RAW:
return "raw";
case SC_CODEC_OPUS:
return "opus";
case SC_CODEC_AAC:
@@ -214,7 +216,7 @@ execute_server(struct sc_server *server,
unsigned dyn_idx = count; // from there, the strings are allocated
#define ADD_PARAM(fmt, ...) { \
char *p = (char *) &cmd[count]; \
char *p; \
if (asprintf(&p, fmt, ## __VA_ARGS__) == -1) { \
goto end; \
} \

View File

@@ -16,11 +16,6 @@ cpu = 'i686'
endian = 'little'
[properties]
ffmpeg_avcodec = 'avcodec-58'
ffmpeg_avformat = 'avformat-58'
ffmpeg_avutil = 'avutil-56'
ffmpeg_swresample = 'swresample-3'
prebuilt_ffmpeg = 'ffmpeg-win32-4.3.1'
prebuilt_ffmpeg = 'ffmpeg-6.0-scrcpy-2/win32'
prebuilt_sdl2 = 'SDL2-2.26.1/i686-w64-mingw32'
prebuilt_libusb_root = 'libusb-1.0.26'
prebuilt_libusb = 'libusb-1.0.26/MinGW-Win32'
prebuilt_libusb = 'libusb-1.0.26/libusb-MinGW-Win32'

View File

@@ -16,11 +16,6 @@ cpu = 'x86_64'
endian = 'little'
[properties]
ffmpeg_avcodec = 'avcodec-59'
ffmpeg_avformat = 'avformat-59'
ffmpeg_avutil = 'avutil-57'
ffmpeg_swresample = 'swresample-4'
prebuilt_ffmpeg = 'ffmpeg-win64-5.1.2'
prebuilt_ffmpeg = 'ffmpeg-6.0-scrcpy-2/win64'
prebuilt_sdl2 = 'SDL2-2.26.1/x86_64-w64-mingw32'
prebuilt_libusb_root = 'libusb-1.0.26'
prebuilt_libusb = 'libusb-1.0.26/MinGW-x64'
prebuilt_libusb = 'libusb-1.0.26/libusb-MinGW-x64'

View File

@@ -11,7 +11,7 @@
.PHONY: default clean \
test \
build-server \
prepare-deps-win32 prepare-deps-win64 \
prepare-deps \
build-win32 build-win64 \
dist-win32 dist-win64 \
zip-win32 zip-win64 \
@@ -62,19 +62,13 @@ build-server:
meson setup "$(SERVER_BUILD_DIR)" --buildtype release -Dcompile_app=false )
ninja -C "$(SERVER_BUILD_DIR)"
prepare-deps-win32:
prepare-deps:
@app/prebuilt-deps/prepare-adb.sh
@app/prebuilt-deps/prepare-sdl.sh
@app/prebuilt-deps/prepare-ffmpeg-win32.sh
@app/prebuilt-deps/prepare-ffmpeg.sh
@app/prebuilt-deps/prepare-libusb.sh
prepare-deps-win64:
@app/prebuilt-deps/prepare-adb.sh
@app/prebuilt-deps/prepare-sdl.sh
@app/prebuilt-deps/prepare-ffmpeg-win64.sh
@app/prebuilt-deps/prepare-libusb.sh
build-win32: prepare-deps-win32
build-win32: prepare-deps
[ -d "$(WIN32_BUILD_DIR)" ] || ( mkdir "$(WIN32_BUILD_DIR)" && \
meson setup "$(WIN32_BUILD_DIR)" \
--cross-file cross_win32.txt \
@@ -83,7 +77,7 @@ build-win32: prepare-deps-win32
-Dportable=true )
ninja -C "$(WIN32_BUILD_DIR)"
build-win64: prepare-deps-win64
build-win64: prepare-deps
[ -d "$(WIN64_BUILD_DIR)" ] || ( mkdir "$(WIN64_BUILD_DIR)" && \
meson setup "$(WIN64_BUILD_DIR)" \
--cross-file cross_win64.txt \
@@ -100,16 +94,16 @@ dist-win32: build-server build-win32
cp app/data/scrcpy-noconsole.vbs "$(DIST)/$(WIN32_TARGET_DIR)"
cp app/data/icon.png "$(DIST)/$(WIN32_TARGET_DIR)"
cp app/data/open_a_terminal_here.bat "$(DIST)/$(WIN32_TARGET_DIR)"
cp app/prebuilt-deps/data/ffmpeg-win32-4.3.1/bin/avutil-56.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-win32-4.3.1/bin/avcodec-58.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-win32-4.3.1/bin/avformat-58.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-win32-4.3.1/bin/swresample-3.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-win32-4.3.1/bin/swscale-5.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-2/win32/bin/avutil-58.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-2/win32/bin/avcodec-60.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-2/win32/bin/avformat-60.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-2/win32/bin/swresample-4.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-2/win32/bin/zlib1.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/platform-tools-33.0.3/adb.exe "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/platform-tools-33.0.3/AdbWinApi.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/platform-tools-33.0.3/AdbWinUsbApi.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/SDL2-2.26.1/i686-w64-mingw32/bin/SDL2.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/libusb-1.0.26/MinGW-Win32/msys-usb-1.0.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/libusb-1.0.26/libusb-MinGW-Win32/bin/msys-usb-1.0.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
dist-win64: build-server build-win64
mkdir -p "$(DIST)/$(WIN64_TARGET_DIR)"
@@ -119,16 +113,16 @@ dist-win64: build-server build-win64
cp app/data/scrcpy-noconsole.vbs "$(DIST)/$(WIN64_TARGET_DIR)"
cp app/data/icon.png "$(DIST)/$(WIN64_TARGET_DIR)"
cp app/data/open_a_terminal_here.bat "$(DIST)/$(WIN64_TARGET_DIR)"
cp app/prebuilt-deps/data/ffmpeg-win64-5.1.2/bin/avutil-57.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-win64-5.1.2/bin/avcodec-59.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-win64-5.1.2/bin/avformat-59.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-win64-5.1.2/bin/swresample-4.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-win64-5.1.2/bin/swscale-6.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-2/win64/bin/avutil-58.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-2/win64/bin/avcodec-60.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-2/win64/bin/avformat-60.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-2/win64/bin/swresample-4.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-2/win64/bin/zlib1.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/platform-tools-33.0.3/adb.exe "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/platform-tools-33.0.3/AdbWinApi.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/platform-tools-33.0.3/AdbWinUsbApi.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/SDL2-2.26.1/x86_64-w64-mingw32/bin/SDL2.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/libusb-1.0.26/MinGW-x64/msys-usb-1.0.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/libusb-1.0.26/libusb-MinGW-x64/bin/msys-usb-1.0.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
zip-win32: dist-win32
cd "$(DIST)"; \

View File

@@ -0,0 +1,148 @@
package com.genymobile.scrcpy;
import com.genymobile.scrcpy.wrappers.ServiceManager;
import android.annotation.SuppressLint;
import android.annotation.TargetApi;
import android.content.ComponentName;
import android.content.Intent;
import android.media.AudioFormat;
import android.media.AudioRecord;
import android.media.AudioTimestamp;
import android.media.MediaCodec;
import android.media.MediaRecorder;
import android.os.Build;
import android.os.SystemClock;
import java.io.IOException;
import java.nio.ByteBuffer;
public final class AudioCapture {
public static final int SAMPLE_RATE = 48000;
public static final int CHANNEL_CONFIG = AudioFormat.CHANNEL_IN_STEREO;
public static final int CHANNELS = 2;
public static final int FORMAT = AudioFormat.ENCODING_PCM_16BIT;
public static final int BYTES_PER_SAMPLE = 2;
private AudioRecord recorder;
private final AudioTimestamp timestamp = new AudioTimestamp();
private long previousPts = 0;
private long nextPts = 0;
public static int millisToBytes(int millis) {
return SAMPLE_RATE * CHANNELS * BYTES_PER_SAMPLE * millis / 1000;
}
private static AudioFormat createAudioFormat() {
AudioFormat.Builder builder = new AudioFormat.Builder();
builder.setEncoding(FORMAT);
builder.setSampleRate(SAMPLE_RATE);
builder.setChannelMask(CHANNEL_CONFIG);
return builder.build();
}
@TargetApi(Build.VERSION_CODES.M)
@SuppressLint({"WrongConstant", "MissingPermission"})
private static AudioRecord createAudioRecord() {
AudioRecord.Builder builder = new AudioRecord.Builder();
if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.S) {
// On older APIs, Workarounds.fillAppInfo() must be called beforehand
builder.setContext(FakeContext.get());
}
builder.setAudioSource(MediaRecorder.AudioSource.REMOTE_SUBMIX);
builder.setAudioFormat(createAudioFormat());
int minBufferSize = AudioRecord.getMinBufferSize(SAMPLE_RATE, CHANNEL_CONFIG, FORMAT);
// This buffer size does not impact latency
builder.setBufferSizeInBytes(8 * minBufferSize);
return builder.build();
}
private static void startWorkaroundAndroid11() {
if (Build.VERSION.SDK_INT == Build.VERSION_CODES.R) {
// Android 11 requires Apps to be at foreground to record audio.
// Normally, each App has its own user ID, so Android checks whether the requesting App has the user ID that's at the foreground.
// But scrcpy server is NOT an App, it's a Java application started from Android shell, so it has the same user ID (2000) with Android
// shell ("com.android.shell").
// If there is an Activity from Android shell running at foreground, then the permission system will believe scrcpy is also in the
// foreground.
if (Build.VERSION.SDK_INT == Build.VERSION_CODES.R) {
Intent intent = new Intent(Intent.ACTION_MAIN);
intent.addFlags(Intent.FLAG_ACTIVITY_NEW_TASK);
intent.addCategory(Intent.CATEGORY_LAUNCHER);
intent.setComponent(new ComponentName(FakeContext.PACKAGE_NAME, "com.android.shell.HeapDumpActivity"));
ServiceManager.getActivityManager().startActivityAsUserWithFeature(intent);
// Wait for activity to start
SystemClock.sleep(150);
}
}
}
private static void stopWorkaroundAndroid11() {
if (Build.VERSION.SDK_INT == Build.VERSION_CODES.R) {
ServiceManager.getActivityManager().forceStopPackage(FakeContext.PACKAGE_NAME);
}
}
public void start() throws AudioCaptureForegroundException {
startWorkaroundAndroid11();
try {
recorder = createAudioRecord();
recorder.startRecording();
} catch (UnsupportedOperationException e) {
if (Build.VERSION.SDK_INT == Build.VERSION_CODES.R) {
Ln.e("Failed to start audio capture");
Ln.e("On Android 11, it is only possible to capture in foreground, make sure that the device is unlocked when starting scrcpy.");
throw new AudioCaptureForegroundException();
}
throw e;
} finally {
stopWorkaroundAndroid11();
}
}
public void stop() {
if (recorder != null) {
// Will call .stop() if necessary, without throwing an IllegalStateException
recorder.release();
}
}
@TargetApi(Build.VERSION_CODES.N)
public int read(ByteBuffer directBuffer, int size, MediaCodec.BufferInfo outBufferInfo) throws IOException {
int r = recorder.read(directBuffer, size);
if (r < 0) {
return r;
}
long pts;
int ret = recorder.getTimestamp(timestamp, AudioTimestamp.TIMEBASE_MONOTONIC);
if (ret == AudioRecord.SUCCESS) {
pts = timestamp.nanoTime / 1000;
} else {
if (nextPts == 0) {
Ln.w("Could not get any audio timestamp");
}
// compute from previous timestamp and packet size
pts = nextPts;
}
long durationMs = r * 1000 / CHANNELS / SAMPLE_RATE;
nextPts = pts + durationMs;
if (previousPts != 0 && pts < previousPts) {
// Audio PTS may come from two sources:
// - recorder.getTimestamp() if the call works;
// - an estimation from the previous PTS and the packet size as a fallback.
//
// Therefore, the property that PTS are monotonically increasing is no guaranteed in corner cases, so enforce it.
pts = previousPts + 1;
}
previousPts = pts;
outBufferInfo.set(0, r, pts, 0);
return r;
}
}

View File

@@ -0,0 +1,7 @@
package com.genymobile.scrcpy;
/**
* Exception thrown if audio capture failed on Android 11 specifically because the running App (shell) was not in foreground.
*/
public class AudioCaptureForegroundException extends Exception {
}

View File

@@ -3,6 +3,7 @@ package com.genymobile.scrcpy;
import android.media.MediaFormat;
public enum AudioCodec implements Codec {
RAW(0x00_72_61_77, "raw", MediaFormat.MIMETYPE_AUDIO_RAW),
OPUS(0x6f_70_75_73, "opus", MediaFormat.MIMETYPE_AUDIO_OPUS),
AAC(0x00_61_61_63, "aac", MediaFormat.MIMETYPE_AUDIO_AAC);

View File

@@ -1,22 +1,12 @@
package com.genymobile.scrcpy;
import com.genymobile.scrcpy.wrappers.ServiceManager;
import android.annotation.SuppressLint;
import android.annotation.TargetApi;
import android.content.ComponentName;
import android.content.Intent;
import android.media.AudioFormat;
import android.media.AudioRecord;
import android.media.AudioTimestamp;
import android.media.MediaCodec;
import android.media.MediaFormat;
import android.media.MediaRecorder;
import android.os.Build;
import android.os.Handler;
import android.os.HandlerThread;
import android.os.Looper;
import android.os.SystemClock;
import java.io.IOException;
import java.nio.ByteBuffer;
@@ -24,7 +14,7 @@ import java.util.List;
import java.util.concurrent.ArrayBlockingQueue;
import java.util.concurrent.BlockingQueue;
public final class AudioEncoder {
public final class AudioEncoder implements AudioRecorder {
private static class InputTask {
private final int index;
@@ -44,14 +34,11 @@ public final class AudioEncoder {
}
}
private static final int SAMPLE_RATE = 48000;
private static final int CHANNEL_CONFIG = AudioFormat.CHANNEL_IN_STEREO;
private static final int CHANNELS = 2;
private static final int FORMAT = AudioFormat.ENCODING_PCM_16BIT;
private static final int BYTES_PER_SAMPLE = 2;
private static final int SAMPLE_RATE = AudioCapture.SAMPLE_RATE;
private static final int CHANNELS = AudioCapture.CHANNELS;
private static final int BUFFER_MS = 5; // milliseconds
private static final int BUFFER_SIZE = SAMPLE_RATE * CHANNELS * BYTES_PER_SAMPLE * BUFFER_MS / 1000;
private static final int READ_MS = 5; // milliseconds
private static final int READ_SIZE = AudioCapture.millisToBytes(READ_MS);
private final Streamer streamer;
private final int bitRate;
@@ -78,29 +65,6 @@ public final class AudioEncoder {
this.encoderName = encoderName;
}
private static AudioFormat createAudioFormat() {
AudioFormat.Builder builder = new AudioFormat.Builder();
builder.setEncoding(FORMAT);
builder.setSampleRate(SAMPLE_RATE);
builder.setChannelMask(CHANNEL_CONFIG);
return builder.build();
}
@TargetApi(Build.VERSION_CODES.M)
@SuppressLint({"WrongConstant", "MissingPermission"})
private static AudioRecord createAudioRecord() {
AudioRecord.Builder builder = new AudioRecord.Builder();
if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.S) {
// On older APIs, Workarounds.fillAppInfo() must be called beforehand
builder.setContext(FakeContext.get());
}
builder.setAudioSource(MediaRecorder.AudioSource.REMOTE_SUBMIX);
builder.setAudioFormat(createAudioFormat());
int minBufferSize = AudioRecord.getMinBufferSize(SAMPLE_RATE, CHANNEL_CONFIG, FORMAT);
builder.setBufferSizeInBytes(minBufferSize);
return builder.build();
}
private static MediaFormat createFormat(String mimeType, int bitRate, List<CodecOption> codecOptions) {
MediaFormat format = new MediaFormat();
format.setString(MediaFormat.KEY_MIME, mimeType);
@@ -121,47 +85,18 @@ public final class AudioEncoder {
}
@TargetApi(Build.VERSION_CODES.N)
private void inputThread(MediaCodec mediaCodec, AudioRecord recorder) throws IOException, InterruptedException {
final AudioTimestamp timestamp = new AudioTimestamp();
long previousPts = 0;
long nextPts = 0;
private void inputThread(MediaCodec mediaCodec, AudioCapture capture) throws IOException, InterruptedException {
final MediaCodec.BufferInfo bufferInfo = new MediaCodec.BufferInfo();
while (!Thread.currentThread().isInterrupted()) {
InputTask task = inputTasks.take();
ByteBuffer buffer = mediaCodec.getInputBuffer(task.index);
int r = recorder.read(buffer, BUFFER_SIZE);
int r = capture.read(buffer, READ_SIZE, bufferInfo);
if (r < 0) {
throw new IOException("Could not read audio: " + r);
}
long pts;
int ret = recorder.getTimestamp(timestamp, AudioTimestamp.TIMEBASE_MONOTONIC);
if (ret == AudioRecord.SUCCESS) {
pts = timestamp.nanoTime / 1000;
} else {
if (nextPts == 0) {
Ln.w("Could not get any audio timestamp");
}
// compute from previous timestamp and packet size
pts = nextPts;
}
long durationMs = r * 1000 / CHANNELS / SAMPLE_RATE;
nextPts = pts + durationMs;
if (previousPts != 0 && pts < previousPts) {
// Audio PTS may come from two sources:
// - recorder.getTimestamp() if the call works;
// - an estimation from the previous PTS and the packet size as a fallback.
//
// Therefore, the property that PTS are monotonically increasing is no guaranteed in corner cases, so enforce it.
pts = previousPts + 1;
}
mediaCodec.queueInputBuffer(task.index, 0, r, pts, 0);
previousPts = pts;
mediaCodec.queueInputBuffer(task.index, bufferInfo.offset, bufferInfo.size, bufferInfo.presentationTimeUs, bufferInfo.flags);
}
}
@@ -183,7 +118,7 @@ public final class AudioEncoder {
thread = new Thread(() -> {
try {
encode();
} catch (ConfigurationException e) {
} catch (ConfigurationException | AudioCaptureForegroundException e) {
// Do not print stack trace, a user-friendly error-message has already been logged
} catch (IOException e) {
Ln.e("Audio encoding error", e);
@@ -222,34 +157,8 @@ public final class AudioEncoder {
}
}
private static void startWorkaroundAndroid11() {
if (Build.VERSION.SDK_INT == Build.VERSION_CODES.R) {
// Android 11 requires Apps to be at foreground to record audio.
// Normally, each App has its own user ID, so Android checks whether the requesting App has the user ID that's at the foreground.
// But Scrcpy server is NOT an App, it's a Java application started from Android shell, so it has the same user ID (2000) with Android
// shell ("com.android.shell").
// If there is an Activity from Android shell running at foreground, then the permission system will believe Scrcpy is also in the
// foreground.
if (Build.VERSION.SDK_INT == Build.VERSION_CODES.R) {
Intent intent = new Intent(Intent.ACTION_MAIN);
intent.addFlags(Intent.FLAG_ACTIVITY_NEW_TASK);
intent.addCategory(Intent.CATEGORY_LAUNCHER);
intent.setComponent(new ComponentName(FakeContext.PACKAGE_NAME, "com.android.shell.HeapDumpActivity"));
ServiceManager.getActivityManager().startActivityAsUserWithFeature(intent);
// Wait for activity to start
SystemClock.sleep(150);
}
}
}
private static void stopWorkaroundAndroid11() {
if (Build.VERSION.SDK_INT == Build.VERSION_CODES.R) {
ServiceManager.getActivityManager().forceStopPackage(FakeContext.PACKAGE_NAME);
}
}
@TargetApi(Build.VERSION_CODES.M)
public void encode() throws IOException, ConfigurationException {
public void encode() throws IOException, ConfigurationException, AudioCaptureForegroundException {
if (Build.VERSION.SDK_INT < Build.VERSION_CODES.R) {
Ln.w("Audio disabled: it is not supported before Android 11");
streamer.writeDisableStream(false);
@@ -257,10 +166,9 @@ public final class AudioEncoder {
}
MediaCodec mediaCodec = null;
AudioRecord recorder = null;
AudioCapture capture = new AudioCapture();
boolean mediaCodecStarted = false;
boolean recorderStarted = false;
try {
Codec codec = streamer.getCodec();
mediaCodec = createMediaCodec(codec, encoderName);
@@ -272,28 +180,13 @@ public final class AudioEncoder {
mediaCodec.setCallback(new EncoderCallback(), new Handler(mediaCodecThread.getLooper()));
mediaCodec.configure(format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
startWorkaroundAndroid11();
try {
recorder = createAudioRecord();
recorder.startRecording();
} catch (UnsupportedOperationException e) {
if (Build.VERSION.SDK_INT == Build.VERSION_CODES.R) {
Ln.e("Failed to start audio capture");
Ln.e("On Android 11, it is only possible to capture in foreground, make sure that the device is unlocked when starting scrcpy.");
// Continue to mirror without audio (configurationError is left to false)
streamer.writeDisableStream(false);
return;
}
} finally {
stopWorkaroundAndroid11();
}
recorderStarted = true;
capture.start();
final MediaCodec mediaCodecRef = mediaCodec;
final AudioRecord recorderRef = recorder;
final AudioCapture captureRef = capture;
inputThread = new Thread(() -> {
try {
inputThread(mediaCodecRef, recorderRef);
inputThread(mediaCodecRef, captureRef);
} catch (IOException | InterruptedException e) {
Ln.e("Audio capture error", e);
} finally {
@@ -366,11 +259,8 @@ public final class AudioEncoder {
}
mediaCodec.release();
}
if (recorder != null) {
if (recorderStarted) {
recorder.stop();
}
recorder.release();
if (capture != null) {
capture.stop();
}
}
}

View File

@@ -0,0 +1,75 @@
package com.genymobile.scrcpy;
import android.media.MediaCodec;
import java.io.IOException;
import java.nio.ByteBuffer;
public final class AudioRawRecorder implements AudioRecorder {
private final Streamer streamer;
private Thread thread;
private static final int READ_MS = 5; // milliseconds
private static final int READ_SIZE = AudioCapture.millisToBytes(READ_MS);
public AudioRawRecorder(Streamer streamer) {
this.streamer = streamer;
}
private void record() throws IOException, AudioCaptureForegroundException {
final ByteBuffer buffer = ByteBuffer.allocateDirect(READ_SIZE);
final MediaCodec.BufferInfo bufferInfo = new MediaCodec.BufferInfo();
AudioCapture capture = new AudioCapture();
try {
capture.start();
streamer.writeHeader();
while (!Thread.currentThread().isInterrupted()) {
buffer.position(0);
int r = capture.read(buffer, READ_SIZE, bufferInfo);
if (r < 0) {
throw new IOException("Could not read audio: " + r);
}
buffer.limit(r);
streamer.writePacket(buffer, bufferInfo);
}
} catch (Throwable e) {
// Notify the client that the audio could not be captured
streamer.writeDisableStream(false);
throw e;
} finally {
capture.stop();
}
}
public void start() {
thread = new Thread(() -> {
try {
record();
} catch (AudioCaptureForegroundException e) {
// Do not print stack trace, a user-friendly error-message has already been logged
} catch (IOException e) {
Ln.e("Audio recording error", e);
} finally {
Ln.d("Audio recorder stopped");
}
});
thread.start();
}
public void stop() {
if (thread != null) {
thread.interrupt();
}
}
public void join() throws InterruptedException {
if (thread != null) {
thread.join();
}
}
}

View File

@@ -0,0 +1,12 @@
package com.genymobile.scrcpy;
/**
* A component able to record audio asynchronously
*
* The implementation is responsible to send packets.
*/
public interface AudioRecorder {
void start();
void stop();
void join() throws InterruptedException;
}

View File

@@ -110,6 +110,11 @@ public final class DesktopConnection implements Closeable {
videoSocket.shutdownInput();
videoSocket.shutdownOutput();
videoSocket.close();
if (audioSocket != null) {
audioSocket.shutdownInput();
audioSocket.shutdownOutput();
audioSocket.close();
}
if (controlSocket != null) {
controlSocket.shutdownInput();
controlSocket.shutdownOutput();

View File

@@ -92,7 +92,7 @@ public final class Server {
}
Controller controller = null;
AudioEncoder audioEncoder = null;
AudioRecorder audioRecorder = null;
try (DesktopConnection connection = DesktopConnection.open(scid, tunnelForward, audio, control, sendDummyByte)) {
if (options.getSendDeviceMeta()) {
@@ -109,10 +109,15 @@ public final class Server {
}
if (audio) {
Streamer audioStreamer = new Streamer(connection.getAudioFd(), options.getAudioCodec(), options.getSendCodecId(),
options.getSendFrameMeta());
audioEncoder = new AudioEncoder(audioStreamer, options.getAudioBitRate(), options.getAudioCodecOptions(), options.getAudioEncoder());
audioEncoder.start();
AudioCodec audioCodec = options.getAudioCodec();
Streamer audioStreamer = new Streamer(connection.getAudioFd(), audioCodec, options.getSendCodecId(), options.getSendFrameMeta());
if (audioCodec == AudioCodec.RAW) {
audioRecorder = new AudioRawRecorder(audioStreamer);
} else {
audioRecorder = new AudioEncoder(audioStreamer, options.getAudioBitRate(), options.getAudioCodecOptions(),
options.getAudioEncoder());
}
audioRecorder.start();
}
Streamer videoStreamer = new Streamer(connection.getVideoFd(), options.getVideoCodec(), options.getSendCodecId(),
@@ -131,8 +136,8 @@ public final class Server {
} finally {
Ln.d("Screen streaming stopped");
initThread.interrupt();
if (audioEncoder != null) {
audioEncoder.stop();
if (audioRecorder != null) {
audioRecorder.stop();
}
if (controller != null) {
controller.stop();
@@ -140,8 +145,8 @@ public final class Server {
try {
initThread.join();
if (audioEncoder != null) {
audioEncoder.join();
if (audioRecorder != null) {
audioRecorder.join();
}
if (controller != null) {
controller.join();

View File

@@ -51,27 +51,34 @@ public final class Streamer {
IO.writeFully(fd, code, 0, code.length);
}
public void writePacket(ByteBuffer codecBuffer, MediaCodec.BufferInfo bufferInfo) throws IOException {
if (codec == AudioCodec.OPUS) {
fixOpusConfigPacket(codecBuffer, bufferInfo);
public void writePacket(ByteBuffer buffer, long pts, boolean config, boolean keyFrame) throws IOException {
if (config && codec == AudioCodec.OPUS) {
fixOpusConfigPacket(buffer);
}
if (sendFrameMeta) {
writeFrameMeta(fd, bufferInfo, codecBuffer.remaining());
writeFrameMeta(fd, buffer.remaining(), pts, config, keyFrame);
}
IO.writeFully(fd, codecBuffer);
IO.writeFully(fd, buffer);
}
private void writeFrameMeta(FileDescriptor fd, MediaCodec.BufferInfo bufferInfo, int packetSize) throws IOException {
public void writePacket(ByteBuffer codecBuffer, MediaCodec.BufferInfo bufferInfo) throws IOException {
long pts = bufferInfo.presentationTimeUs;
boolean config = (bufferInfo.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG) != 0;
boolean keyFrame = (bufferInfo.flags & MediaCodec.BUFFER_FLAG_KEY_FRAME) != 0;
writePacket(codecBuffer, pts, config, keyFrame);
}
private void writeFrameMeta(FileDescriptor fd, int packetSize, long pts, boolean config, boolean keyFrame) throws IOException {
headerBuffer.clear();
long ptsAndFlags;
if ((bufferInfo.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG) != 0) {
if (config) {
ptsAndFlags = PACKET_FLAG_CONFIG; // non-media data packet
} else {
ptsAndFlags = bufferInfo.presentationTimeUs;
if ((bufferInfo.flags & MediaCodec.BUFFER_FLAG_KEY_FRAME) != 0) {
ptsAndFlags = pts;
if (keyFrame) {
ptsAndFlags |= PACKET_FLAG_KEY_FRAME;
}
}
@@ -82,7 +89,7 @@ public final class Streamer {
IO.writeFully(fd, headerBuffer);
}
private static void fixOpusConfigPacket(ByteBuffer buffer, MediaCodec.BufferInfo bufferInfo) throws IOException {
private static void fixOpusConfigPacket(ByteBuffer buffer) throws IOException {
// Here is an example of the config packet received for an OPUS stream:
//
// 00000000 41 4f 50 55 53 48 44 52 13 00 00 00 00 00 00 00 |AOPUSHDR........|
@@ -99,11 +106,6 @@ public final class Streamer {
//
// <https://developer.android.com/reference/android/media/MediaCodec#CSD>
boolean isConfig = (bufferInfo.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG) != 0;
if (!isConfig) {
return;
}
if (buffer.remaining() < 16) {
throw new IOException("Not enough data in OPUS config packet");
}