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3 Commits

Author SHA1 Message Date
Romain Vimont
3d93a099b7 Add log 2023-11-03 11:51:41 +01:00
Romain Vimont
29c305ca5f Initialize FakeContext first 2023-11-03 11:51:32 +01:00
Romain Vimont
65d6bdb237 Attempt to fix #4392
<https://github.com/Genymobile/scrcpy/issues/4392#issuecomment-1790954884>
2023-11-03 09:58:23 +01:00
25 changed files with 159 additions and 273 deletions

View File

@@ -97,7 +97,7 @@ _scrcpy() {
return
;;
--audio-codec)
COMPREPLY=($(compgen -W 'opus aac flac raw' -- "$cur"))
COMPREPLY=($(compgen -W 'opus aac raw' -- "$cur"))
return
;;
--video-source)
@@ -125,7 +125,7 @@ _scrcpy() {
return
;;
--record-format)
COMPREPLY=($(compgen -W 'mp4 mkv m4a mka opus aac flac wav' -- "$cur"))
COMPREPLY=($(compgen -W 'mkv mp4' -- "$cur"))
return
;;
--render-driver)

View File

@@ -11,7 +11,7 @@ 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-buffer=[Configure the audio buffering delay (in milliseconds)]'
'--audio-codec=[Select the audio codec]:codec:(opus aac flac raw)'
'--audio-codec=[Select the audio codec]:codec:(opus aac raw)'
'--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]'
'--audio-source=[Select the audio source]:source:(output mic)'
@@ -65,7 +65,7 @@ arguments=(
'--push-target=[Set the target directory for pushing files to the device by drag and drop]'
{-r,--record=}'[Record screen to file]:record file:_files'
'--raw-key-events[Inject key events for all input keys, and ignore text events]'
'--record-format=[Force recording format]:format:(mp4 mkv m4a mka opus aac flac wav)'
'--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)'

View File

@@ -6,11 +6,11 @@ cd "$DIR"
mkdir -p "$PREBUILT_DATA_DIR"
cd "$PREBUILT_DATA_DIR"
VERSION=6.1-scrcpy-2
VERSION=6.0-scrcpy-4
DEP_DIR="ffmpeg-$VERSION"
FILENAME="$DEP_DIR".7z
SHA256SUM=7f25f638dc24a0f5d4af07a088b6a604cf33548900bbfd2f6ce0bae050b7664d
SHA256SUM=39274b321491ce83e76cab5d24e7cbe3f402d3ccf382f739b13be5651c146b60
if [[ -d "$DEP_DIR" ]]
then

View File

@@ -35,7 +35,7 @@ Default is 50.
.TP
.BI "\-\-audio\-codec " name
Select an audio codec (opus, aac, flac or raw).
Select an audio codec (opus, aac or raw).
Default is opus.
@@ -347,7 +347,7 @@ Record screen to
The format is determined by the
.B \-\-record\-format
option if set, or by the file extension.
option if set, or by the file extension (.mp4 or .mkv).
.TP
.B \-\-raw\-key\-events
@@ -355,7 +355,7 @@ Inject key events for all input keys, and ignore text events.
.TP
.BI "\-\-record\-format " format
Force recording format (mp4, mkv, m4a, mka, opus, aac, flac or wav).
Force recording format (either mp4 or mkv).
.TP
.BI "\-\-render\-driver " name

View File

@@ -152,7 +152,7 @@ static const struct sc_option options[] = {
.longopt_id = OPT_AUDIO_CODEC,
.longopt = "audio-codec",
.argdesc = "name",
.text = "Select an audio codec (opus, aac, flac or raw).\n"
.text = "Select an audio codec (opus, aac or raw).\n"
"Default is opus.",
},
{
@@ -583,7 +583,7 @@ static const struct sc_option options[] = {
.argdesc = "file.mp4",
.text = "Record screen to file.\n"
"The format is determined by the --record-format option if "
"set, or by the file extension.",
"set, or by the file extension (.mp4 or .mkv).",
},
{
.longopt_id = OPT_RAW_KEY_EVENTS,
@@ -594,8 +594,7 @@ static const struct sc_option options[] = {
.longopt_id = OPT_RECORD_FORMAT,
.longopt = "record-format",
.argdesc = "format",
.text = "Force recording format (mp4, mkv, m4a, mka, opus, aac, flac "
"or wav).",
.text = "Force recording format (either mp4 or mkv).",
},
{
.longopt_id = OPT_RENDER_DRIVER,
@@ -1627,12 +1626,6 @@ get_record_format(const char *name) {
if (!strcmp(name, "aac")) {
return SC_RECORD_FORMAT_AAC;
}
if (!strcmp(name, "flac")) {
return SC_RECORD_FORMAT_FLAC;
}
if (!strcmp(name, "wav")) {
return SC_RECORD_FORMAT_WAV;
}
return 0;
}
@@ -1702,15 +1695,11 @@ parse_audio_codec(const char *optarg, enum sc_codec *codec) {
*codec = SC_CODEC_AAC;
return true;
}
if (!strcmp(optarg, "flac")) {
*codec = SC_CODEC_FLAC;
return true;
}
if (!strcmp(optarg, "raw")) {
*codec = SC_CODEC_RAW;
return true;
}
LOGE("Unsupported audio codec: %s (expected opus, aac, flac or raw)", optarg);
LOGE("Unsupported audio codec: %s (expected opus, aac or raw)", optarg);
return false;
}
@@ -2268,19 +2257,6 @@ parse_args_with_getopt(struct scrcpy_cli_args *args, int argc, char *argv[],
opts->require_audio = true;
}
if (opts->audio_playback && opts->audio_buffer == -1) {
if (opts->audio_codec == SC_CODEC_FLAC) {
// Use 50 ms audio buffer by default, but use a higher value for FLAC,
// which is not low latency (the default encoder produces blocks of
// 4096 samples, which represent ~85.333ms).
LOGI("FLAC audio: audio buffer increased to 120 ms (use "
"--audio-buffer to set a custom value)");
opts->audio_buffer = SC_TICK_FROM_MS(120);
} else {
opts->audio_buffer = SC_TICK_FROM_MS(50);
}
}
#ifdef HAVE_V4L2
if (v4l2) {
if (opts->lock_video_orientation ==
@@ -2376,6 +2352,11 @@ parse_args_with_getopt(struct scrcpy_cli_args *args, int argc, char *argv[],
}
}
if (opts->audio_codec == SC_CODEC_RAW) {
LOGW("Recording does not support RAW audio codec");
return false;
}
if (opts->video
&& sc_record_format_is_audio_only(opts->record_format)) {
LOGE("Audio container does not support video stream");
@@ -2395,30 +2376,6 @@ parse_args_with_getopt(struct scrcpy_cli_args *args, int argc, char *argv[],
"(try with --audio-codec=aac)");
return false;
}
if (opts->record_format == SC_RECORD_FORMAT_FLAC
&& opts->audio_codec != SC_CODEC_FLAC) {
LOGE("Recording to FLAC file requires a FLAC audio stream "
"(try with --audio-codec=flac)");
return false;
}
if (opts->record_format == SC_RECORD_FORMAT_WAV
&& opts->audio_codec != SC_CODEC_RAW) {
LOGE("Recording to WAV file requires a RAW audio stream "
"(try with --audio-codec=raw)");
return false;
}
if ((opts->record_format == SC_RECORD_FORMAT_MP4 ||
opts->record_format == SC_RECORD_FORMAT_M4A)
&& opts->audio_codec == SC_CODEC_RAW) {
LOGE("Recording to MP4 container does not support RAW audio");
return false;
}
}
if (opts->audio_codec == SC_CODEC_FLAC && opts->audio_bit_rate) {
LOGW("--audio-bit-rate is ignored for FLAC audio codec");
}
if (opts->audio_codec == SC_CODEC_RAW) {

View File

@@ -25,8 +25,7 @@ sc_demuxer_to_avcodec_id(uint32_t codec_id) {
#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_OPUS UINT32_C(0x6f707573) // "opus" in ASCII
#define SC_CODEC_ID_AAC UINT32_C(0x00616163) // "aac" in ASCII
#define SC_CODEC_ID_FLAC UINT32_C(0x666c6163) // "flac" in ASCII
#define SC_CODEC_ID_AAC UINT32_C(0x00616163) // "aac in ASCII"
#define SC_CODEC_ID_RAW UINT32_C(0x00726177) // "raw" in ASCII
switch (codec_id) {
case SC_CODEC_ID_H264:
@@ -44,8 +43,6 @@ sc_demuxer_to_avcodec_id(uint32_t codec_id) {
return AV_CODEC_ID_OPUS;
case SC_CODEC_ID_AAC:
return AV_CODEC_ID_AAC;
case SC_CODEC_ID_FLAC:
return AV_CODEC_ID_FLAC;
case SC_CODEC_ID_RAW:
return AV_CODEC_ID_PCM_S16LE;
default:
@@ -210,11 +207,6 @@ run_demuxer(void *data) {
codec_ctx->channels = 2;
#endif
codec_ctx->sample_rate = 48000;
if (raw_codec_id == SC_CODEC_ID_FLAC) {
// The sample_fmt is not set by the FLAC decoder
codec_ctx->sample_fmt = AV_SAMPLE_FMT_S16;
}
}
if (avcodec_open2(codec_ctx, codec, NULL) < 0) {

View File

@@ -46,7 +46,7 @@ const struct scrcpy_options scrcpy_options_default = {
.window_height = 0,
.display_id = 0,
.display_buffer = 0,
.audio_buffer = -1, // depends on the audio format,
.audio_buffer = SC_TICK_FROM_MS(50),
.audio_output_buffer = SC_TICK_FROM_MS(5),
.time_limit = 0,
#ifdef HAVE_V4L2

View File

@@ -25,8 +25,6 @@ enum sc_record_format {
SC_RECORD_FORMAT_MKA,
SC_RECORD_FORMAT_OPUS,
SC_RECORD_FORMAT_AAC,
SC_RECORD_FORMAT_FLAC,
SC_RECORD_FORMAT_WAV,
};
static inline bool
@@ -34,9 +32,7 @@ sc_record_format_is_audio_only(enum sc_record_format fmt) {
return fmt == SC_RECORD_FORMAT_M4A
|| fmt == SC_RECORD_FORMAT_MKA
|| fmt == SC_RECORD_FORMAT_OPUS
|| fmt == SC_RECORD_FORMAT_AAC
|| fmt == SC_RECORD_FORMAT_FLAC
|| fmt == SC_RECORD_FORMAT_WAV;
|| fmt == SC_RECORD_FORMAT_AAC;
}
enum sc_codec {
@@ -45,7 +41,6 @@ enum sc_codec {
SC_CODEC_AV1,
SC_CODEC_OPUS,
SC_CODEC_AAC,
SC_CODEC_FLAC,
SC_CODEC_RAW,
};

View File

@@ -69,10 +69,6 @@ sc_recorder_get_format_name(enum sc_record_format format) {
return "matroska";
case SC_RECORD_FORMAT_OPUS:
return "opus";
case SC_RECORD_FORMAT_FLAC:
return "flac";
case SC_RECORD_FORMAT_WAV:
return "wav";
default:
return NULL;
}
@@ -105,7 +101,7 @@ sc_recorder_write_stream(struct sc_recorder *recorder,
AVStream *stream = recorder->ctx->streams[st->index];
sc_recorder_rescale_packet(stream, packet);
if (st->last_pts != AV_NOPTS_VALUE && packet->pts <= st->last_pts) {
LOGD("Fixing PTS non monotonically increasing in stream %d "
LOGW("Fixing PTS non monotonically increasing in stream %d "
"(%" PRIi64 " >= %" PRIi64 ")",
st->index, st->last_pts, packet->pts);
packet->pts = ++st->last_pts;
@@ -170,14 +166,13 @@ sc_recorder_close_output_file(struct sc_recorder *recorder) {
}
static inline bool
sc_recorder_must_wait_for_config_packets(struct sc_recorder *recorder) {
sc_recorder_has_empty_queues(struct sc_recorder *recorder) {
if (recorder->video && sc_vecdeque_is_empty(&recorder->video_queue)) {
// The video queue is empty
return true;
}
if (recorder->audio && recorder->audio_expects_config_packet
&& sc_vecdeque_is_empty(&recorder->audio_queue)) {
if (recorder->audio && sc_vecdeque_is_empty(&recorder->audio_queue)) {
// The audio queue is empty (when audio is enabled)
return true;
}
@@ -193,7 +188,7 @@ sc_recorder_process_header(struct sc_recorder *recorder) {
while (!recorder->stopped &&
((recorder->video && !recorder->video_init)
|| (recorder->audio && !recorder->audio_init)
|| sc_recorder_must_wait_for_config_packets(recorder))) {
|| sc_recorder_has_empty_queues(recorder))) {
sc_cond_wait(&recorder->cond, &recorder->mutex);
}
@@ -212,8 +207,7 @@ sc_recorder_process_header(struct sc_recorder *recorder) {
}
AVPacket *audio_pkt = NULL;
if (recorder->audio_expects_config_packet &&
!sc_vecdeque_is_empty(&recorder->audio_queue)) {
if (!sc_vecdeque_is_empty(&recorder->audio_queue)) {
assert(recorder->audio);
audio_pkt = sc_vecdeque_pop(&recorder->audio_queue);
}
@@ -601,10 +595,6 @@ sc_recorder_audio_packet_sink_open(struct sc_packet_sink *sink,
recorder->audio_stream.index = stream->index;
// A config packet is provided for all formats supported except raw audio
recorder->audio_expects_config_packet =
ctx->codec_id != AV_CODEC_ID_PCM_S16LE;
recorder->audio_init = true;
sc_cond_signal(&recorder->cond);
sc_mutex_unlock(&recorder->mutex);
@@ -717,8 +707,6 @@ sc_recorder_init(struct sc_recorder *recorder, const char *filename,
recorder->video_init = false;
recorder->audio_init = false;
recorder->audio_expects_config_packet = false;
sc_recorder_stream_init(&recorder->video_stream);
sc_recorder_stream_init(&recorder->audio_stream);

View File

@@ -50,8 +50,6 @@ struct sc_recorder {
bool video_init;
bool audio_init;
bool audio_expects_config_packet;
struct sc_recorder_stream video_stream;
struct sc_recorder_stream audio_stream;

View File

@@ -417,14 +417,10 @@ scrcpy(struct scrcpy_options *options) {
if (options->video_playback) {
sdl_set_hints(options->render_driver);
}
// Initialize the video subsystem even if --no-video or --no-video-playback
// is passed so that clipboard synchronization still works.
// <https://github.com/Genymobile/scrcpy/issues/4418>
if (SDL_Init(SDL_INIT_VIDEO)) {
LOGE("Could not initialize SDL video: %s", SDL_GetError());
goto end;
if (SDL_Init(SDL_INIT_VIDEO)) {
LOGE("Could not initialize SDL video: %s", SDL_GetError());
goto end;
}
}
if (options->audio_playback) {

View File

@@ -178,8 +178,6 @@ sc_server_get_codec_name(enum sc_codec codec) {
return "opus";
case SC_CODEC_AAC:
return "aac";
case SC_CODEC_FLAC:
return "flac";
case SC_CODEC_RAW:
return "raw";
default:

View File

@@ -16,6 +16,6 @@ cpu = 'i686'
endian = 'little'
[properties]
prebuilt_ffmpeg = 'ffmpeg-6.1-scrcpy-2/win32'
prebuilt_ffmpeg = 'ffmpeg-6.0-scrcpy-4/win32'
prebuilt_sdl2 = 'SDL2-2.28.4/i686-w64-mingw32'
prebuilt_libusb = 'libusb-1.0.26/libusb-MinGW-Win32'

View File

@@ -16,6 +16,6 @@ cpu = 'x86_64'
endian = 'little'
[properties]
prebuilt_ffmpeg = 'ffmpeg-6.1-scrcpy-2/win64'
prebuilt_ffmpeg = 'ffmpeg-6.0-scrcpy-4/win64'
prebuilt_sdl2 = 'SDL2-2.28.4/x86_64-w64-mingw32'
prebuilt_libusb = 'libusb-1.0.26/libusb-MinGW-x64'

View File

@@ -62,13 +62,12 @@ scrcpy --audio-source=mic --no-video --no-playback --record=file.opus
## Codec
The audio codec can be selected. The possible values are `opus` (default),
`aac`, `flac` and `raw` (uncompressed PCM 16-bit LE):
The audio codec can be selected. The possible values are `opus` (default), `aac`
and `raw` (uncompressed PCM 16-bit LE):
```bash
scrcpy --audio-codec=opus # default
scrcpy --audio-codec=aac
scrcpy --audio-codec=flac
scrcpy --audio-codec=raw
```
@@ -81,14 +80,7 @@ then your device has no Opus encoder: try `scrcpy --audio-codec=aac`.
For advanced usage, to pass arbitrary parameters to the [`MediaFormat`],
check `--audio-codec-options` in the manpage or in `scrcpy --help`.
For example, to change the [FLAC compression level]:
```bash
scrcpy --audio-codec=flac --audio-codec-options=flac-compression-level=8
```
[`MediaFormat`]: https://developer.android.com/reference/android/media/MediaFormat
[FLAC compression level]: https://developer.android.com/reference/android/media/MediaFormat#KEY_FLAC_COMPRESSION_LEVEL
## Encoder

View File

@@ -18,9 +18,7 @@ To record only the audio:
```bash
scrcpy --no-video --record=file.opus
scrcpy --no-video --audio-codec=aac --record=file.aac
scrcpy --no-video --audio-codec=flac --record=file.flac
scrcpy --no-video --audio-codec=raw --record=file.wav
# .m4a/.mp4 and .mka/.mkv are also supported for opus, aac and flac
# .m4a/.mp4 and .mka/.mkv are also supported for both opus and aac
```
Timestamps are captured on the device, so [packet delay variation] does not
@@ -33,17 +31,14 @@ course, not if you capture your scrcpy window and audio output on the computer).
## Format
The video and audio streams are encoded on the device, but are muxed on the
client side. Several formats (containers) are supported:
- MP4 (`.mp4`, `.m4a`, `.aac`)
- Matroska (`.mkv`, `.mka`)
- OPUS (`.opus`)
- FLAC (`.flac`)
- WAV (`.wav`)
client side. Two formats (containers) are supported:
- Matroska (`.mkv`)
- MP4 (`.mp4`)
The container is automatically selected based on the filename.
It is also possible to explicitly select a container (in that case the filename
needs not end with a known extension):
needs not end with `.mkv` or `.mp4`):
```
scrcpy --record=file --record-format=mkv

View File

@@ -94,10 +94,10 @@ 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-6.1-scrcpy-2/win32/bin/avutil-58.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.1-scrcpy-2/win32/bin/avcodec-60.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.1-scrcpy-2/win32/bin/avformat-60.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.1-scrcpy-2/win32/bin/swresample-4.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-4/win32/bin/avutil-58.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-4/win32/bin/avcodec-60.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-4/win32/bin/avformat-60.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-4/win32/bin/swresample-4.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/platform-tools-34.0.5/adb.exe "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/platform-tools-34.0.5/AdbWinApi.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
cp app/prebuilt-deps/data/platform-tools-34.0.5/AdbWinUsbApi.dll "$(DIST)/$(WIN32_TARGET_DIR)/"
@@ -112,10 +112,10 @@ 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-6.1-scrcpy-2/win64/bin/avutil-58.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.1-scrcpy-2/win64/bin/avcodec-60.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.1-scrcpy-2/win64/bin/avformat-60.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.1-scrcpy-2/win64/bin/swresample-4.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-4/win64/bin/avutil-58.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-4/win64/bin/avcodec-60.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-4/win64/bin/avformat-60.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/ffmpeg-6.0-scrcpy-4/win64/bin/swresample-4.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/platform-tools-34.0.5/adb.exe "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/platform-tools-34.0.5/AdbWinApi.dll "$(DIST)/$(WIN64_TARGET_DIR)/"
cp app/prebuilt-deps/data/platform-tools-34.0.5/AdbWinUsbApi.dll "$(DIST)/$(WIN64_TARGET_DIR)/"

View File

@@ -24,19 +24,11 @@ public final class AudioCapture {
public static final int ENCODING = AudioFormat.ENCODING_PCM_16BIT;
public static final int BYTES_PER_SAMPLE = 2;
// Never read more than 1024 samples, even if the buffer is bigger (that would increase latency).
// A lower value is useless, since the system captures audio samples by blocks of 1024 (so for example if we read by blocks of 256 samples, we
// receive 4 successive blocks without waiting, then we wait for the 4 next ones).
public static final int MAX_READ_SIZE = 1024 * CHANNELS * BYTES_PER_SAMPLE;
private static final long ONE_SAMPLE_US = (1000000 + SAMPLE_RATE - 1) / SAMPLE_RATE; // 1 sample in microseconds (used for fixing PTS)
private final int audioSource;
private AudioRecord recorder;
private final AudioTimestamp timestamp = new AudioTimestamp();
private long previousRecorderTimestamp = -1;
private long previousPts = 0;
private long nextPts = 0;
@@ -44,6 +36,10 @@ public final class AudioCapture {
this.audioSource = audioSource.value();
}
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(ENCODING);
@@ -139,8 +135,8 @@ public final class AudioCapture {
}
@TargetApi(Build.VERSION_CODES.N)
public int read(ByteBuffer directBuffer, MediaCodec.BufferInfo outBufferInfo) {
int r = recorder.read(directBuffer, MAX_READ_SIZE);
public int read(ByteBuffer directBuffer, int size, MediaCodec.BufferInfo outBufferInfo) {
int r = recorder.read(directBuffer, size);
if (r <= 0) {
return r;
}
@@ -148,9 +144,8 @@ public final class AudioCapture {
long pts;
int ret = recorder.getTimestamp(timestamp, AudioTimestamp.TIMEBASE_MONOTONIC);
if (ret == AudioRecord.SUCCESS && timestamp.nanoTime != previousRecorderTimestamp) {
if (ret == AudioRecord.SUCCESS) {
pts = timestamp.nanoTime / 1000;
previousRecorderTimestamp = timestamp.nanoTime;
} else {
if (nextPts == 0) {
Ln.w("Could not get any audio timestamp");
@@ -162,13 +157,13 @@ public final class AudioCapture {
long durationUs = r * 1000000 / (CHANNELS * BYTES_PER_SAMPLE * SAMPLE_RATE);
nextPts = pts + durationUs;
if (previousPts != 0 && pts < previousPts + ONE_SAMPLE_US) {
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 + ONE_SAMPLE_US;
pts = previousPts + 1;
}
previousPts = pts;

View File

@@ -5,7 +5,6 @@ import android.media.MediaFormat;
public enum AudioCodec implements Codec {
OPUS(0x6f_70_75_73, "opus", MediaFormat.MIMETYPE_AUDIO_OPUS),
AAC(0x00_61_61_63, "aac", MediaFormat.MIMETYPE_AUDIO_AAC),
FLAC(0x66_6c_61_63, "flac", MediaFormat.MIMETYPE_AUDIO_FLAC),
RAW(0x00_72_61_77, "raw", MediaFormat.MIMETYPE_AUDIO_RAW);
private final int id; // 4-byte ASCII representation of the name

View File

@@ -37,6 +37,9 @@ public final class AudioEncoder implements AsyncProcessor {
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 final AudioCapture capture;
private final Streamer streamer;
private final int bitRate;
@@ -90,7 +93,7 @@ public final class AudioEncoder implements AsyncProcessor {
while (!Thread.currentThread().isInterrupted()) {
InputTask task = inputTasks.take();
ByteBuffer buffer = mediaCodec.getInputBuffer(task.index);
int r = capture.read(buffer, bufferInfo);
int r = capture.read(buffer, READ_SIZE, bufferInfo);
if (r <= 0) {
throw new IOException("Could not read audio: " + r);
}

View File

@@ -13,6 +13,9 @@ public final class AudioRawRecorder implements AsyncProcessor {
private Thread thread;
private static final int READ_MS = 5; // milliseconds
private static final int READ_SIZE = AudioCapture.millisToBytes(READ_MS);
public AudioRawRecorder(AudioCapture capture, Streamer streamer) {
this.capture = capture;
this.streamer = streamer;
@@ -25,22 +28,16 @@ public final class AudioRawRecorder implements AsyncProcessor {
return;
}
final ByteBuffer buffer = ByteBuffer.allocateDirect(AudioCapture.MAX_READ_SIZE);
final ByteBuffer buffer = ByteBuffer.allocateDirect(READ_SIZE);
final MediaCodec.BufferInfo bufferInfo = new MediaCodec.BufferInfo();
try {
try {
capture.start();
} catch (Throwable t) {
// Notify the client that the audio could not be captured
streamer.writeDisableStream(false);
throw t;
}
capture.start();
streamer.writeAudioHeader();
while (!Thread.currentThread().isInterrupted()) {
buffer.position(0);
int r = capture.read(buffer, bufferInfo);
int r = capture.read(buffer, READ_SIZE, bufferInfo);
if (r < 0) {
throw new IOException("Could not read audio: " + r);
}
@@ -48,11 +45,10 @@ public final class AudioRawRecorder implements AsyncProcessor {
streamer.writePacket(buffer, bufferInfo);
}
} catch (IOException e) {
// Broken pipe is expected on close, because the socket is closed by the client
if (!IO.isBrokenPipe(e)) {
Ln.e("Audio capture error", e);
}
} catch (Throwable e) {
// Notify the client that the audio could not be captured
streamer.writeDisableStream(false);
throw e;
} finally {
capture.stop();
}
@@ -66,8 +62,8 @@ public final class AudioRawRecorder implements AsyncProcessor {
record();
} catch (AudioCaptureForegroundException e) {
// Do not print stack trace, a user-friendly error-message has already been logged
} catch (Throwable t) {
Ln.e("Audio recording error", t);
} catch (IOException e) {
Ln.e("Audio recording error", e);
fatalError = true;
} finally {
Ln.d("Audio recorder stopped");

View File

@@ -2,12 +2,11 @@ package com.genymobile.scrcpy;
import android.annotation.TargetApi;
import android.content.AttributionSource;
import android.content.Context;
import android.content.ContextWrapper;
import android.content.MutableContextWrapper;
import android.os.Build;
import android.os.Process;
public final class FakeContext extends ContextWrapper {
public final class FakeContext extends MutableContextWrapper {
public static final String PACKAGE_NAME = "com.android.shell";
public static final int ROOT_UID = 0; // Like android.os.Process.ROOT_UID, but before API 29
@@ -19,7 +18,7 @@ public final class FakeContext extends ContextWrapper {
}
private FakeContext() {
super(Workarounds.getSystemContext());
super(Workarounds.retrieveSystemContext());
}
@Override
@@ -45,9 +44,4 @@ public final class FakeContext extends ContextWrapper {
public int getDeviceId() {
return 0;
}
@Override
public Context getApplicationContext() {
return this;
}
}

View File

@@ -93,26 +93,19 @@ public final class LogUtils {
builder.append("\n (none)");
} else {
for (String id : cameraIds) {
builder.append("\n --camera-id=").append(id);
builder.append("\n --video-source=camera --camera-id=").append(id);
CameraCharacteristics characteristics = cameraManager.getCameraCharacteristics(id);
int facing = characteristics.get(CameraCharacteristics.LENS_FACING);
builder.append(" (").append(getCameraFacingName(facing)).append(", ");
Rect activeSize = characteristics.get(CameraCharacteristics.SENSOR_INFO_ACTIVE_ARRAY_SIZE);
builder.append(activeSize.width()).append("x").append(activeSize.height());
builder.append(activeSize.width()).append("x").append(activeSize.height()).append(", ");
try {
// Capture frame rates for low-FPS mode are the same for every resolution
Range<Integer>[] lowFpsRanges = characteristics.get(CameraCharacteristics.CONTROL_AE_AVAILABLE_TARGET_FPS_RANGES);
SortedSet<Integer> uniqueLowFps = getUniqueSet(lowFpsRanges);
builder.append(", fps=").append(uniqueLowFps);
} catch (Exception e) {
// Some devices may provide invalid ranges, causing an IllegalArgumentException "lower must be less than or equal to upper"
Ln.w("Could not get available frame rates for camera " + id, e);
}
builder.append(')');
// Capture frame rates for low-FPS mode are the same for every resolution
Range<Integer>[] lowFpsRanges = characteristics.get(CameraCharacteristics.CONTROL_AE_AVAILABLE_TARGET_FPS_RANGES);
SortedSet<Integer> uniqueLowFps = getUniqueSet(lowFpsRanges);
builder.append("fps=").append(uniqueLowFps).append(')');
if (includeSizes) {
StreamConfigurationMap configs = characteristics.get(CameraCharacteristics.SCALER_STREAM_CONFIGURATION_MAP);

View File

@@ -5,14 +5,14 @@ import android.media.MediaCodec;
import java.io.FileDescriptor;
import java.io.IOException;
import java.nio.ByteBuffer;
import java.nio.ByteOrder;
import java.util.Arrays;
public final class Streamer {
private static final long PACKET_FLAG_CONFIG = 1L << 63;
private static final long PACKET_FLAG_KEY_FRAME = 1L << 62;
private static final long AOPUSHDR = 0x5244485355504F41L; // "AOPUSHDR" in ASCII (little-endian)
private final FileDescriptor fd;
private final Codec codec;
private final boolean sendCodecMeta;
@@ -30,7 +30,6 @@ public final class Streamer {
public Codec getCodec() {
return codec;
}
public void writeAudioHeader() throws IOException {
if (sendCodecMeta) {
ByteBuffer buffer = ByteBuffer.allocate(4);
@@ -63,12 +62,8 @@ public final class Streamer {
}
public void writePacket(ByteBuffer buffer, long pts, boolean config, boolean keyFrame) throws IOException {
if (config) {
if (codec == AudioCodec.OPUS) {
fixOpusConfigPacket(buffer);
} else if (codec == AudioCodec.FLAC) {
fixFlacConfigPacket(buffer);
}
if (config && codec == AudioCodec.OPUS) {
fixOpusConfigPacket(buffer);
}
if (sendFrameMeta) {
@@ -125,14 +120,11 @@ public final class Streamer {
throw new IOException("Not enough data in OPUS config packet");
}
final byte[] opusHeaderId = {'A', 'O', 'P', 'U', 'S', 'H', 'D', 'R'};
byte[] idBuffer = new byte[8];
buffer.get(idBuffer);
if (!Arrays.equals(idBuffer, opusHeaderId)) {
long id = buffer.getLong();
if (id != AOPUSHDR) {
throw new IOException("OPUS header not found");
}
// The size is in native byte-order
long sizeLong = buffer.getLong();
if (sizeLong < 0 || sizeLong >= 0x7FFFFFFF) {
throw new IOException("Invalid block size in OPUS header: " + sizeLong);
@@ -146,41 +138,4 @@ public final class Streamer {
// Set the buffer to point to the OPUS header slice
buffer.limit(buffer.position() + size);
}
private static void fixFlacConfigPacket(ByteBuffer buffer) throws IOException {
// 00000000 66 4c 61 43 00 00 00 22 |fLaC..." |
// -------------- BELOW IS THE PART WE MUST PUT AS EXTRADATA -------------------
// 00000000 10 00 10 00 00 00 00 00 | ........|
// 00000010 00 00 0b b8 02 f0 00 00 00 00 00 00 00 00 00 00 |................|
// 00000020 00 00 00 00 00 00 00 00 00 00 |.......... |
// ------------------------------------------------------------------------------
// 00000020 84 00 00 28 20 00 | ...( .|
// 00000030 00 00 72 65 66 65 72 65 6e 63 65 20 6c 69 62 46 |..reference libF|
// 00000040 4c 41 43 20 31 2e 33 2e 32 20 32 30 32 32 31 30 |LAC 1.3.2 202210|
// 00000050 32 32 00 00 00 00 |22....|
//
// <https://developer.android.com/reference/android/media/MediaCodec#CSD>
if (buffer.remaining() < 8) {
throw new IOException("Not enough data in FLAC config packet");
}
final byte[] flacHeaderId = {'f', 'L', 'a', 'C'};
byte[] idBuffer = new byte[4];
buffer.get(idBuffer);
if (!Arrays.equals(idBuffer, flacHeaderId)) {
throw new IOException("FLAC header not found");
}
// The size is in big-endian
buffer.order(ByteOrder.BIG_ENDIAN);
int size = buffer.getInt();
if (buffer.remaining() < size) {
throw new IOException("Not enough data in FLAC header (invalid size: " + size + ")");
}
// Set the buffer to point to the FLAC header slice
buffer.limit(buffer.position() + size);
}
}

View File

@@ -7,6 +7,7 @@ import android.content.AttributionSource;
import android.content.Context;
import android.content.ContextWrapper;
import android.content.pm.ApplicationInfo;
import android.content.pm.PackageManager;
import android.media.AudioAttributes;
import android.media.AudioManager;
import android.media.AudioRecord;
@@ -21,34 +22,24 @@ import java.lang.reflect.Method;
public final class Workarounds {
private static final Class<?> ACTIVITY_THREAD_CLASS;
private static final Object ACTIVITY_THREAD;
static {
prepareMainLooper();
try {
// ActivityThread activityThread = new ActivityThread();
ACTIVITY_THREAD_CLASS = Class.forName("android.app.ActivityThread");
Constructor<?> activityThreadConstructor = ACTIVITY_THREAD_CLASS.getDeclaredConstructor();
activityThreadConstructor.setAccessible(true);
ACTIVITY_THREAD = activityThreadConstructor.newInstance();
// ActivityThread.sCurrentActivityThread = activityThread;
Field sCurrentActivityThreadField = ACTIVITY_THREAD_CLASS.getDeclaredField("sCurrentActivityThread");
sCurrentActivityThreadField.setAccessible(true);
sCurrentActivityThreadField.set(null, ACTIVITY_THREAD);
} catch (Exception e) {
throw new AssertionError(e);
}
}
private static Class<?> activityThreadClass;
private static Object activityThread;
private Workarounds() {
// not instantiable
}
public static void apply(boolean audio, boolean camera) {
Workarounds.prepareMainLooper();
try {
fillActivityThread();
FakeContext.get();
} catch (Exception e) {
throw new AssertionError(e);
}
boolean mustFillAppInfo = false;
boolean mustFillBaseContext = false;
boolean mustFillAppContext = false;
if (Build.BRAND.equalsIgnoreCase("meizu")) {
@@ -69,6 +60,7 @@ public final class Workarounds {
// - <https://github.com/Genymobile/scrcpy/issues/4015#issuecomment-1595382142>
// - <https://github.com/Genymobile/scrcpy/issues/3805#issuecomment-1596148031>
mustFillAppInfo = true;
mustFillBaseContext = true;
mustFillAppContext = true;
}
@@ -81,12 +73,15 @@ public final class Workarounds {
if (camera) {
mustFillAppInfo = true;
mustFillAppContext = true;
mustFillBaseContext = true;
}
if (mustFillAppInfo) {
Workarounds.fillAppInfo();
}
if (mustFillBaseContext) {
Workarounds.fillBaseContext();
}
if (mustFillAppContext) {
Workarounds.fillAppContext();
}
@@ -105,9 +100,27 @@ public final class Workarounds {
Looper.prepareMainLooper();
}
@SuppressLint("PrivateApi,DiscouragedPrivateApi")
private static void fillActivityThread() throws Exception {
if (activityThread == null) {
// ActivityThread activityThread = new ActivityThread();
activityThreadClass = Class.forName("android.app.ActivityThread");
Constructor<?> activityThreadConstructor = activityThreadClass.getDeclaredConstructor();
activityThreadConstructor.setAccessible(true);
activityThread = activityThreadConstructor.newInstance();
// ActivityThread.sCurrentActivityThread = activityThread;
Field sCurrentActivityThreadField = activityThreadClass.getDeclaredField("sCurrentActivityThread");
sCurrentActivityThreadField.setAccessible(true);
sCurrentActivityThreadField.set(null, activityThread);
}
}
@SuppressLint("PrivateApi,DiscouragedPrivateApi")
private static void fillAppInfo() {
try {
fillActivityThread();
// ActivityThread.AppBindData appBindData = new ActivityThread.AppBindData();
Class<?> appBindDataClass = Class.forName("android.app.ActivityThread$AppBindData");
Constructor<?> appBindDataConstructor = appBindDataClass.getDeclaredConstructor();
@@ -117,15 +130,26 @@ public final class Workarounds {
ApplicationInfo applicationInfo = new ApplicationInfo();
applicationInfo.packageName = FakeContext.PACKAGE_NAME;
Application application = new Application() {
@Override
public String getOpPackageName() {
return FakeContext.PACKAGE_NAME;
}
};
Field initialApplicationField = activityThreadClass.getDeclaredField("mInitialApplication");
initialApplicationField.setAccessible(true);
initialApplicationField.set(activityThread, application);
// appBindData.appInfo = applicationInfo;
Field appInfoField = appBindDataClass.getDeclaredField("appInfo");
appInfoField.setAccessible(true);
appInfoField.set(appBindData, applicationInfo);
// activityThread.mBoundApplication = appBindData;
Field mBoundApplicationField = ACTIVITY_THREAD_CLASS.getDeclaredField("mBoundApplication");
Field mBoundApplicationField = activityThreadClass.getDeclaredField("mBoundApplication");
mBoundApplicationField.setAccessible(true);
mBoundApplicationField.set(ACTIVITY_THREAD, appBindData);
mBoundApplicationField.set(activityThread, appBindData);
} catch (Throwable throwable) {
// this is a workaround, so failing is not an error
Ln.d("Could not fill app info: " + throwable.getMessage());
@@ -135,29 +159,33 @@ public final class Workarounds {
@SuppressLint("PrivateApi,DiscouragedPrivateApi")
private static void fillAppContext() {
try {
Application app = new Application();
fillActivityThread();
Application app = Application.class.newInstance();
Field baseField = ContextWrapper.class.getDeclaredField("mBase");
baseField.setAccessible(true);
baseField.set(app, FakeContext.get());
// activityThread.mInitialApplication = app;
Field mInitialApplicationField = ACTIVITY_THREAD_CLASS.getDeclaredField("mInitialApplication");
Field mInitialApplicationField = activityThreadClass.getDeclaredField("mInitialApplication");
mInitialApplicationField.setAccessible(true);
mInitialApplicationField.set(ACTIVITY_THREAD, app);
mInitialApplicationField.set(activityThread, app);
} catch (Throwable throwable) {
// this is a workaround, so failing is not an error
Ln.d("Could not fill app context: " + throwable.getMessage());
}
}
static Context getSystemContext() {
private static void fillBaseContext() {
try {
Method getSystemContextMethod = ACTIVITY_THREAD_CLASS.getDeclaredMethod("getSystemContext");
return (Context) getSystemContextMethod.invoke(ACTIVITY_THREAD);
fillActivityThread();
Method getSystemContextMethod = activityThreadClass.getDeclaredMethod("getSystemContext");
Context context = (Context) getSystemContextMethod.invoke(activityThread);
FakeContext.get().setBaseContext(context);
} catch (Throwable throwable) {
// this is a workaround, so failing is not an error
Ln.d("Could not get system context: " + throwable.getMessage());
return null;
Ln.d("Could not fill base context: " + throwable.getMessage());
}
}
@@ -296,4 +324,16 @@ public final class Workarounds {
throw new RuntimeException("Cannot create AudioRecord");
}
}
static Context retrieveSystemContext() {
try {
Method getSystemContextMethod = activityThreadClass.getDeclaredMethod("getSystemContext");
Context ctx = (Context) getSystemContextMethod.invoke(activityThread);
Ln.i("===== " + ctx);
return ctx;
} catch (Exception e) {
Ln.e("Cannot retrieve system context", e);
return null;
}
}
}