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75 Commits

Author SHA1 Message Date
Romain Vimont
d939b99dde Warn on ignored audio options
For raw audio codec, some audio options are ignored.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:59:36 +01:00
Romain Vimont
bac8119d17 Make --audio-codec=raw the default
Encoding and decoding the audio stream adds latency, which is not ideal
for real-time mirroring.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:47:49 +01:00
Romain Vimont
29e47bdcdc Add --audio-codec=raw option
Add support for raw (PCM S16 LE) audio codec (a decoder is included in
FFmpeg).

Even if it consumes more bandwidth, it provides a lower latency.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:47:49 +01:00
Romain Vimont
73cd59ab25 Add raw audio recorder
Add an alternative AudioRecorder to stream raw packets without encoding.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:47:42 +01:00
Romain Vimont
ed070a067b 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>
2023-03-03 21:47:23 +01:00
Romain Vimont
0924ab9d58 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>
2023-03-03 21:47:20 +01:00
Romain Vimont
14c27d1756 Stop on decoder frame push error
On push, frame sinks report downstream errors to stop upstream
components. Do not ignore the error.
2023-03-03 21:47:20 +01:00
Romain Vimont
501b5bceff 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>
2023-03-03 21:47:18 +01:00
Romain Vimont
e7905e2069 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>
2023-03-03 21:47:15 +01:00
Romain Vimont
446b76bb92 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>
2023-03-03 21:47:13 +01:00
Romain Vimont
93bbcd53cd 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>
2023-03-03 21:47:10 +01:00
Romain Vimont
d19c03d015 Use frame source trait in decoder 2023-03-03 21:47:10 +01:00
Romain Vimont
8dac37fff0 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.
2023-03-03 21:47:10 +01:00
Romain Vimont
0f5205ad7b Use packet source trait in demuxer 2023-03-03 21:47:10 +01:00
Romain Vimont
94ec920eff 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.
2023-03-03 21:47:10 +01:00
Romain Vimont
9ac466ad84 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.
2023-03-03 21:47:10 +01:00
Romain Vimont
9c0f8c7d2f Report video buffer downstream errors
Make the video buffer stop if its consumer could not receive a frame.
2023-03-03 21:47:10 +01:00
Romain Vimont
099445a77a Stop the video buffer on error
If an error occurs from the video buffer thread (typically an
out-of-memory error), then stop.
2023-03-03 21:47:10 +01:00
Romain Vimont
ecc2f3f25c 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.
2023-03-03 21:47:10 +01:00
Romain Vimont
7e4c1328e3 Remove sc_queue
All uses have been replaced by VecDeque.
2023-03-03 21:47:10 +01:00
Romain Vimont
da8fb85c29 Remove cbuf
All uses have been replaced by VecDeque.
2023-03-03 21:47:10 +01:00
Romain Vimont
c2c0910a19 Use VecDeque in aoa_hid
Replace cbuf by VecDeque in aoa_hid
2023-03-03 21:47:10 +01:00
Romain Vimont
4eb8263068 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.
2023-03-03 21:47:10 +01:00
Romain Vimont
4dbd665796 Use VecDeque in controller
Replace cbuf by VecDeque in controller.
2023-03-03 21:47:10 +01:00
Romain Vimont
4daf4c1161 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.
2023-03-03 21:47:10 +01:00
Romain Vimont
fc4ece9d6a 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.
2023-03-03 21:47:10 +01:00
Romain Vimont
abeaa9ae07 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>
2023-03-03 21:47:10 +01:00
Romain Vimont
e9b98ed367 Add sc_allocarray() util
Add a function to allocate an array, which fails safely in the case
where the multiplication would overflow.
2023-03-03 21:47:10 +01:00
Romain Vimont
2e1cdf204a Use reallocarray() in sc_vector
This fails safely in case of overflow.
2023-03-03 21:47:10 +01:00
Romain Vimont
6bb37abd4b Add compat for reallocarray()
This function fails safely in the case where the multiplication would
overflow.
2023-03-03 21:47:10 +01:00
Romain Vimont
7ae7543de2 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.
2023-03-03 21:47:10 +01:00
Romain Vimont
fd0a82ba93 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>
2023-03-03 21:47:07 +01:00
Romain Vimont
81fb51e625 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>
2023-03-03 21:47:04 +01:00
Simon Chan
d7ebacf956 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>
2023-03-03 21:46:49 +01:00
Romain Vimont
5105824d59 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>
2023-03-03 21:46:29 +01:00
Romain Vimont
1fc03c73b1 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>
2023-03-03 21:46:24 +01:00
Romain Vimont
45b17282e1 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>
2023-03-03 21:46:22 +01:00
Romain Vimont
ac0724b388 Pass AVCodecContext to frame sinks
Frame consumers may need details about the frame format.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:46:17 +01:00
Romain Vimont
92537b9d7c Add an audio decoder
PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:46:01 +01:00
Romain Vimont
d661d39765 Give a name to decoder instances
This will be useful in logs.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:45:58 +01:00
Romain Vimont
d2ce33eeae Rename decoder to video_decoder
This prepares the introduction of audio_decoder.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:45:20 +01:00
Romain Vimont
e89ffb9c92 Log display sizes in display list
This is more convenient than just the display id alone.
2023-03-03 21:45:20 +01:00
Romain Vimont
0eeca46d5c Add --list-device-displays 2023-03-03 21:45:20 +01:00
Romain Vimont
e97d556905 Move log message helpers to LogUtils
This class will also contain other log helpers.
2023-03-03 21:45:20 +01:00
Romain Vimont
771a63f9d4 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>
2023-03-03 21:45:17 +01:00
Romain Vimont
4b8c200e7e Add --list-encoders
Add an option to list the device encoders properly.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:45:16 +01:00
Romain Vimont
0134786ce7 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>
2023-03-03 21:45:14 +01:00
Romain Vimont
5275d4fa16 Add --audio-encoder
Similar to --video-encoder, but for audio.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:45:12 +01:00
Romain Vimont
b0c3191682 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>
2023-03-03 21:45:10 +01:00
Romain Vimont
de3e1cc77a Add --audio-codec-options
Similar to --video-codec-options, but for audio.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:45:08 +01:00
Romain Vimont
1c4000c151 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>
2023-03-03 21:45:06 +01:00
Romain Vimont
5dbd6a3971 Add support for AAC audio codec
Add option --audio-codec=aac.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:45:04 +01:00
Romain Vimont
07388d8509 Add --audio-codec
Introduce the selection mechanism. Alternative codecs will be added
later.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:45:02 +01:00
Romain Vimont
ebabae2045 Add --audio-bit-rate
Add an option to configure the audio bit-rate.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:44:59 +01:00
Romain Vimont
779994f06a Disable MethodLength checkstyle on createOptions()
This method will grow as needed to initialize options.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:44:56 +01:00
Romain Vimont
c3dd91cf72 Rename --encoder to --video-encoder
This prepares the introduction of --audio-encoder.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:44:54 +01:00
Romain Vimont
f40d7b78c2 Rename --codec-options to --video-codec-options
This prepares the introduction of --audio-codec-options.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:44:52 +01:00
Romain Vimont
30622c169a Rename --bit-rate to --video-bit-rate
This prepares the introduction of --audio-bit-rate.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:44:49 +01:00
Romain Vimont
1feaeed8ea Rename --codec to --video-codec
This prepares the introduction of --audio-codec.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:44:47 +01:00
Romain Vimont
c649e2a4c1 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>
2023-03-03 21:44:45 +01:00
Romain Vimont
fd019753fb 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>
2023-03-03 21:44:42 +01:00
Romain Vimont
2fda50851c 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>
2023-03-03 21:44:37 +01:00
Romain Vimont
4e67dbd0e2 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>
2023-03-03 21:44:30 +01:00
Romain Vimont
71238dbe53 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>
2023-03-03 21:44:28 +01:00
Romain Vimont
29b479bcf9 Rename video-specific variables in recorder
This paves the way to add audio-specific variables.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:44:27 +01:00
Romain Vimont
10cb52265f 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>
2023-03-03 21:44:22 +01:00
Romain Vimont
94718156f1 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>
2023-03-03 21:44:20 +01:00
Romain Vimont
2301ad3353 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>
2023-03-03 21:44:18 +01:00
Romain Vimont
b72280bcc3 Give a name to demuxer instances
This will be useful in logs.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:44:15 +01:00
Romain Vimont
c813002f0a Rename demuxer to video_demuxer
There will be another demuxer instance for audio.

PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757>
2023-03-03 21:44:13 +01:00
Romain Vimont
af12bdf05b 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>
2023-03-03 21:44:11 +01:00
Romain Vimont
e69871ae8a 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>
2023-03-03 21:44:08 +01:00
Romain Vimont
c7845387cf 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>
2023-03-03 21:44:05 +01:00
Simon Chan
26553129d4 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>
2023-03-03 21:44:03 +01:00
Simon Chan
87bb26fe61 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>
2023-03-03 21:44:01 +01:00
11 changed files with 32 additions and 60 deletions

View File

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

View File

@@ -35,17 +35,7 @@ Default is 0 (no buffering).
.BI "\-\-audio\-codec " name
Select an audio codec (raw, opus or aac).
Default is opus.
.TP
.BI "\-\-audio\-codec\-options " key\fR[:\fItype\fR]=\fIvalue\fR[,...]
Set a list of comma-separated key:type=value options for the device audio encoder.
The possible values for 'type' are 'int' (default), 'long', 'float' and 'string'.
The list of possible codec options is available in the Android documentation
.UR https://d.android.com/reference/android/media/MediaFormat
.UE .
Default is raw.
.TP
.BI "\-\-audio\-encoder " name

View File

@@ -74,7 +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) {

View File

@@ -133,7 +133,7 @@ static const struct sc_option options[] = {
.longopt = "audio-codec",
.argdesc = "name",
.text = "Select an audio codec (raw, opus or aac).\n"
"Default is opus.",
"Default is raw.",
},
{
.longopt_id = OPT_AUDIO_CODEC_OPTIONS,
@@ -1917,20 +1917,9 @@ 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");
}
LOGW("Recording does not support RAW audio codec, automatically "
"switching to --audio-codec=opus");
opts->audio_codec = SC_CODEC_OPUS;
}
if (opts->audio_codec == SC_CODEC_RAW) {

View File

@@ -16,7 +16,7 @@ const struct scrcpy_options scrcpy_options_default = {
#endif
.log_level = SC_LOG_LEVEL_INFO,
.video_codec = SC_CODEC_H264,
.audio_codec = SC_CODEC_OPUS,
.audio_codec = SC_CODEC_RAW,
.record_format = SC_RECORD_FORMAT_AUTO,
.keyboard_input_mode = SC_KEYBOARD_INPUT_MODE_INJECT,
.mouse_input_mode = SC_MOUSE_INPUT_MODE_INJECT,

View File

@@ -238,7 +238,7 @@ execute_server(struct sc_server *server,
ADD_PARAM("video_codec=%s",
sc_server_get_codec_name(params->video_codec));
}
if (params->audio_codec != SC_CODEC_OPUS) {
if (params->audio_codec != SC_CODEC_RAW) {
ADD_PARAM("audio_codec=%s",
sc_server_get_codec_name(params->audio_codec));
}

View File

@@ -37,8 +37,8 @@ public final class AudioEncoder implements AudioRecorder {
private static final int SAMPLE_RATE = AudioCapture.SAMPLE_RATE;
private static final int CHANNELS = AudioCapture.CHANNELS;
private static final int READ_MS = 5; // milliseconds
private static final int READ_SIZE = AudioCapture.millisToBytes(READ_MS);
private static final int BUFFER_MS = 5; // milliseconds
private static final int BUFFER_SIZE = AudioCapture.millisToBytes(BUFFER_MS);
private final Streamer streamer;
private final int bitRate;
@@ -91,7 +91,7 @@ public final class AudioEncoder implements AudioRecorder {
while (!Thread.currentThread().isInterrupted()) {
InputTask task = inputTasks.take();
ByteBuffer buffer = mediaCodec.getInputBuffer(task.index);
int r = capture.read(buffer, READ_SIZE, bufferInfo);
int r = capture.read(buffer, BUFFER_SIZE, bufferInfo);
if (r < 0) {
throw new IOException("Could not read audio: " + r);
}

View File

@@ -11,15 +11,15 @@ public final class AudioRawRecorder implements AudioRecorder {
private Thread thread;
private static final int READ_MS = 5; // milliseconds
private static final int READ_SIZE = AudioCapture.millisToBytes(READ_MS);
private static final int BUFFER_MS = 5; // milliseconds
private static final int BUFFER_SIZE = AudioCapture.millisToBytes(BUFFER_MS);
public AudioRawRecorder(Streamer streamer) {
this.streamer = streamer;
}
private void record() throws IOException, AudioCaptureForegroundException {
final ByteBuffer buffer = ByteBuffer.allocateDirect(READ_SIZE);
final ByteBuffer buffer = ByteBuffer.allocateDirect(BUFFER_SIZE);
final MediaCodec.BufferInfo bufferInfo = new MediaCodec.BufferInfo();
AudioCapture capture = new AudioCapture();
@@ -29,7 +29,7 @@ public final class AudioRawRecorder implements AudioRecorder {
streamer.writeHeader();
while (!Thread.currentThread().isInterrupted()) {
buffer.position(0);
int r = capture.read(buffer, READ_SIZE, bufferInfo);
int r = capture.read(buffer, BUFFER_SIZE, bufferInfo);
if (r < 0) {
throw new IOException("Could not read audio: " + r);
}

View File

@@ -110,11 +110,6 @@ 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

@@ -11,7 +11,7 @@ public class Options {
private boolean audio = true;
private int maxSize;
private VideoCodec videoCodec = VideoCodec.H264;
private AudioCodec audioCodec = AudioCodec.OPUS;
private AudioCodec audioCodec = AudioCodec.RAW;
private int videoBitRate = 8000000;
private int audioBitRate = 196000;
private int maxFps;

View File

@@ -51,34 +51,27 @@ public final class Streamer {
IO.writeFully(fd, code, 0, code.length);
}
public void writePacket(ByteBuffer buffer, long pts, boolean config, boolean keyFrame) throws IOException {
if (config && codec == AudioCodec.OPUS) {
fixOpusConfigPacket(buffer);
public void writePacket(ByteBuffer codecBuffer, MediaCodec.BufferInfo bufferInfo) throws IOException {
if (codec == AudioCodec.OPUS) {
fixOpusConfigPacket(codecBuffer, bufferInfo);
}
if (sendFrameMeta) {
writeFrameMeta(fd, buffer.remaining(), pts, config, keyFrame);
writeFrameMeta(fd, bufferInfo, codecBuffer.remaining());
}
IO.writeFully(fd, buffer);
IO.writeFully(fd, codecBuffer);
}
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 {
private void writeFrameMeta(FileDescriptor fd, MediaCodec.BufferInfo bufferInfo, int packetSize) throws IOException {
headerBuffer.clear();
long ptsAndFlags;
if (config) {
if ((bufferInfo.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG) != 0) {
ptsAndFlags = PACKET_FLAG_CONFIG; // non-media data packet
} else {
ptsAndFlags = pts;
if (keyFrame) {
ptsAndFlags = bufferInfo.presentationTimeUs;
if ((bufferInfo.flags & MediaCodec.BUFFER_FLAG_KEY_FRAME) != 0) {
ptsAndFlags |= PACKET_FLAG_KEY_FRAME;
}
}
@@ -89,7 +82,7 @@ public final class Streamer {
IO.writeFully(fd, headerBuffer);
}
private static void fixOpusConfigPacket(ByteBuffer buffer) throws IOException {
private static void fixOpusConfigPacket(ByteBuffer buffer, MediaCodec.BufferInfo bufferInfo) 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........|
@@ -106,6 +99,11 @@ 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");
}