tdesktop/Telegram/SourceFiles/media/media_audio_capture.cpp

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/*
This file is part of Telegram Desktop,
the official desktop application for the Telegram messaging service.
For license and copyright information please follow this link:
https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
*/
#include "media/media_audio_capture.h"
#include "media/media_audio_ffmpeg_loader.h"
#include <AL/al.h>
#include <AL/alc.h>
#include <AL/alext.h>
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#include <numeric>
namespace Media {
namespace Capture {
namespace {
constexpr auto kCaptureFrequency = Player::kDefaultFrequency;
constexpr auto kCaptureSkipDuration = TimeMs(400);
constexpr auto kCaptureFadeInDuration = TimeMs(300);
Instance *CaptureInstance = nullptr;
bool ErrorHappened(ALCdevice *device) {
ALenum errCode;
if ((errCode = alcGetError(device)) != ALC_NO_ERROR) {
LOG(("Audio Capture Error: %1, %2").arg(errCode).arg((const char *)alcGetString(device, errCode)));
return true;
}
return false;
}
} // namespace
void Start() {
Assert(CaptureInstance == nullptr);
CaptureInstance = new Instance();
instance()->check();
}
void Finish() {
delete base::take(CaptureInstance);
}
Instance::Instance() : _inner(new Inner(&_thread)) {
CaptureInstance = this;
connect(this, SIGNAL(start()), _inner, SLOT(onStart()));
connect(this, SIGNAL(stop(bool)), _inner, SLOT(onStop(bool)));
connect(_inner, SIGNAL(done(QByteArray, VoiceWaveform, qint32)), this, SIGNAL(done(QByteArray, VoiceWaveform, qint32)));
connect(_inner, SIGNAL(updated(quint16, qint32)), this, SIGNAL(updated(quint16, qint32)));
connect(_inner, SIGNAL(error()), this, SIGNAL(error()));
connect(&_thread, SIGNAL(started()), _inner, SLOT(onInit()));
connect(&_thread, SIGNAL(finished()), _inner, SLOT(deleteLater()));
_thread.start();
}
void Instance::check() {
_available = false;
if (auto defaultDevice = alcGetString(0, ALC_CAPTURE_DEFAULT_DEVICE_SPECIFIER)) {
if (auto device = alcCaptureOpenDevice(defaultDevice, kCaptureFrequency, AL_FORMAT_MONO16, kCaptureFrequency / 5)) {
auto error = ErrorHappened(device);
alcCaptureCloseDevice(device);
_available = !error;
}
}
}
Instance::~Instance() {
_inner = nullptr;
_thread.quit();
_thread.wait();
}
Instance *instance() {
return CaptureInstance;
}
struct Instance::Inner::Private {
ALCdevice *device = nullptr;
AVOutputFormat *fmt = nullptr;
uchar *ioBuffer = nullptr;
AVIOContext *ioContext = nullptr;
AVFormatContext *fmtContext = nullptr;
AVStream *stream = nullptr;
AVCodec *codec = nullptr;
AVCodecContext *codecContext = nullptr;
bool opened = false;
int srcSamples = 0;
int dstSamples = 0;
int maxDstSamples = 0;
int dstSamplesSize = 0;
int fullSamples = 0;
uint8_t **srcSamplesData = nullptr;
uint8_t **dstSamplesData = nullptr;
SwrContext *swrContext = nullptr;
int32 lastUpdate = 0;
uint16 levelMax = 0;
QByteArray data;
int32 dataPos = 0;
int64 waveformMod = 0;
int64 waveformEach = (kCaptureFrequency / 100);
uint16 waveformPeak = 0;
QVector<uchar> waveform;
static int _read_data(void *opaque, uint8_t *buf, int buf_size) {
auto l = reinterpret_cast<Private*>(opaque);
int32 nbytes = qMin(l->data.size() - l->dataPos, int32(buf_size));
if (nbytes <= 0) {
return 0;
}
memcpy(buf, l->data.constData() + l->dataPos, nbytes);
l->dataPos += nbytes;
return nbytes;
}
static int _write_data(void *opaque, uint8_t *buf, int buf_size) {
auto l = reinterpret_cast<Private*>(opaque);
if (buf_size <= 0) return 0;
if (l->dataPos + buf_size > l->data.size()) l->data.resize(l->dataPos + buf_size);
memcpy(l->data.data() + l->dataPos, buf, buf_size);
l->dataPos += buf_size;
return buf_size;
}
static int64_t _seek_data(void *opaque, int64_t offset, int whence) {
auto l = reinterpret_cast<Private*>(opaque);
int32 newPos = -1;
switch (whence) {
case SEEK_SET: newPos = offset; break;
case SEEK_CUR: newPos = l->dataPos + offset; break;
case SEEK_END: newPos = l->data.size() + offset; break;
case AVSEEK_SIZE: {
// Special whence for determining filesize without any seek.
return l->data.size();
} break;
}
if (newPos < 0) {
return -1;
}
l->dataPos = newPos;
return l->dataPos;
}
};
Instance::Inner::Inner(QThread *thread) : d(new Private()) {
moveToThread(thread);
_timer.moveToThread(thread);
connect(&_timer, SIGNAL(timeout()), this, SLOT(onTimeout()));
}
Instance::Inner::~Inner() {
onStop(false);
delete d;
}
void Instance::Inner::onInit() {
}
void Instance::Inner::onStart() {
// Start OpenAL Capture
const ALCchar *dName = alcGetString(0, ALC_CAPTURE_DEFAULT_DEVICE_SPECIFIER);
DEBUG_LOG(("Audio Info: Capture device name '%1'").arg(dName));
d->device = alcCaptureOpenDevice(dName, kCaptureFrequency, AL_FORMAT_MONO16, kCaptureFrequency / 5);
if (!d->device) {
LOG(("Audio Error: capture device not present!"));
emit error();
return;
}
alcCaptureStart(d->device);
if (ErrorHappened(d->device)) {
alcCaptureCloseDevice(d->device);
d->device = nullptr;
emit error();
return;
}
// Create encoding context
d->ioBuffer = (uchar*)av_malloc(AVBlockSize);
d->ioContext = avio_alloc_context(d->ioBuffer, AVBlockSize, 1, static_cast<void*>(d), &Private::_read_data, &Private::_write_data, &Private::_seek_data);
int res = 0;
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
AVOutputFormat *fmt = 0;
while ((fmt = av_oformat_next(fmt))) {
if (fmt->name == qstr("opus")) {
break;
}
}
if (!fmt) {
LOG(("Audio Error: Unable to find opus AVOutputFormat for capture"));
onStop(false);
emit error();
return;
}
if ((res = avformat_alloc_output_context2(&d->fmtContext, fmt, 0, 0)) < 0) {
LOG(("Audio Error: Unable to avformat_alloc_output_context2 for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
onStop(false);
emit error();
return;
}
d->fmtContext->pb = d->ioContext;
d->fmtContext->flags |= AVFMT_FLAG_CUSTOM_IO;
d->opened = true;
// Add audio stream
d->codec = avcodec_find_encoder(fmt->audio_codec);
if (!d->codec) {
LOG(("Audio Error: Unable to avcodec_find_encoder for capture"));
onStop(false);
emit error();
return;
}
d->stream = avformat_new_stream(d->fmtContext, d->codec);
if (!d->stream) {
LOG(("Audio Error: Unable to avformat_new_stream for capture"));
onStop(false);
emit error();
return;
}
d->stream->id = d->fmtContext->nb_streams - 1;
d->codecContext = avcodec_alloc_context3(d->codec);
if (!d->codecContext) {
LOG(("Audio Error: Unable to avcodec_alloc_context3 for capture"));
onStop(false);
emit error();
return;
}
av_opt_set_int(d->codecContext, "refcounted_frames", 1, 0);
d->codecContext->sample_fmt = AV_SAMPLE_FMT_FLTP;
d->codecContext->bit_rate = 64000;
d->codecContext->channel_layout = AV_CH_LAYOUT_MONO;
d->codecContext->sample_rate = kCaptureFrequency;
d->codecContext->channels = 1;
if (d->fmtContext->oformat->flags & AVFMT_GLOBALHEADER) {
d->codecContext->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
// Open audio stream
if ((res = avcodec_open2(d->codecContext, d->codec, nullptr)) < 0) {
LOG(("Audio Error: Unable to avcodec_open2 for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
onStop(false);
emit error();
return;
}
// Alloc source samples
d->srcSamples = (d->codecContext->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE) ? 10000 : d->codecContext->frame_size;
//if ((res = av_samples_alloc_array_and_samples(&d->srcSamplesData, 0, d->codecContext->channels, d->srcSamples, d->codecContext->sample_fmt, 0)) < 0) {
// LOG(("Audio Error: Unable to av_samples_alloc_array_and_samples for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
// onStop(false);
// emit error();
// return;
//}
// Using _captured directly
// Prepare resampling
d->swrContext = swr_alloc();
if (!d->swrContext) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
av_opt_set_int(d->swrContext, "in_channel_count", d->codecContext->channels, 0);
av_opt_set_int(d->swrContext, "in_sample_rate", d->codecContext->sample_rate, 0);
av_opt_set_sample_fmt(d->swrContext, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int(d->swrContext, "out_channel_count", d->codecContext->channels, 0);
av_opt_set_int(d->swrContext, "out_sample_rate", d->codecContext->sample_rate, 0);
av_opt_set_sample_fmt(d->swrContext, "out_sample_fmt", d->codecContext->sample_fmt, 0);
if ((res = swr_init(d->swrContext)) < 0) {
LOG(("Audio Error: Unable to swr_init for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
onStop(false);
emit error();
return;
}
d->maxDstSamples = d->srcSamples;
if ((res = av_samples_alloc_array_and_samples(&d->dstSamplesData, 0, d->codecContext->channels, d->maxDstSamples, d->codecContext->sample_fmt, 0)) < 0) {
LOG(("Audio Error: Unable to av_samples_alloc_array_and_samples for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
onStop(false);
emit error();
return;
}
d->dstSamplesSize = av_samples_get_buffer_size(0, d->codecContext->channels, d->maxDstSamples, d->codecContext->sample_fmt, 0);
if ((res = avcodec_parameters_from_context(d->stream->codecpar, d->codecContext)) < 0) {
LOG(("Audio Error: Unable to avcodec_parameters_from_context for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
onStop(false);
emit error();
return;
}
// Write file header
if ((res = avformat_write_header(d->fmtContext, 0)) < 0) {
LOG(("Audio Error: Unable to avformat_write_header for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
onStop(false);
emit error();
return;
}
_timer.start(50);
_captured.clear();
_captured.reserve(AudioVoiceMsgBufferSize);
DEBUG_LOG(("Audio Capture: started!"));
}
void Instance::Inner::onStop(bool needResult) {
if (!_timer.isActive()) return; // in onStop() already
_timer.stop();
if (d->device) {
alcCaptureStop(d->device);
onTimeout(); // get last data
}
// Write what is left
if (!_captured.isEmpty()) {
auto fadeSamples = kCaptureFadeInDuration * kCaptureFrequency / 1000;
auto capturedSamples = static_cast<int>(_captured.size() / sizeof(short));
if ((_captured.size() % sizeof(short)) || (d->fullSamples + capturedSamples < kCaptureFrequency) || (capturedSamples < fadeSamples)) {
d->fullSamples = 0;
d->dataPos = 0;
d->data.clear();
d->waveformMod = 0;
d->waveformPeak = 0;
d->waveform.clear();
} else {
float64 coef = 1. / fadeSamples, fadedFrom = 0;
for (short *ptr = ((short*)_captured.data()) + capturedSamples, *end = ptr - fadeSamples; ptr != end; ++fadedFrom) {
--ptr;
*ptr = qRound(fadedFrom * coef * *ptr);
}
if (capturedSamples % d->srcSamples) {
int32 s = _captured.size();
_captured.resize(s + (d->srcSamples - (capturedSamples % d->srcSamples)) * sizeof(short));
memset(_captured.data() + s, 0, _captured.size() - s);
}
int32 framesize = d->srcSamples * d->codecContext->channels * sizeof(short), encoded = 0;
while (_captured.size() >= encoded + framesize) {
processFrame(encoded, framesize);
encoded += framesize;
}
writeFrame(nullptr); // drain the codec
if (encoded != _captured.size()) {
d->fullSamples = 0;
d->dataPos = 0;
d->data.clear();
d->waveformMod = 0;
d->waveformPeak = 0;
d->waveform.clear();
}
}
}
DEBUG_LOG(("Audio Capture: stopping (need result: %1), size: %2, samples: %3").arg(Logs::b(needResult)).arg(d->data.size()).arg(d->fullSamples));
_captured = QByteArray();
// Finish stream
if (d->device) {
av_write_trailer(d->fmtContext);
}
QByteArray result = d->fullSamples ? d->data : QByteArray();
VoiceWaveform waveform;
qint32 samples = d->fullSamples;
if (samples && !d->waveform.isEmpty()) {
int64 count = d->waveform.size(), sum = 0;
if (count >= Player::kWaveformSamplesCount) {
QVector<uint16> peaks;
peaks.reserve(Player::kWaveformSamplesCount);
uint16 peak = 0;
for (int32 i = 0; i < count; ++i) {
uint16 sample = uint16(d->waveform.at(i)) * 256;
if (peak < sample) {
peak = sample;
}
sum += Player::kWaveformSamplesCount;
if (sum >= count) {
sum -= count;
peaks.push_back(peak);
peak = 0;
}
}
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auto sum = std::accumulate(peaks.cbegin(), peaks.cend(), 0LL);
peak = qMax(int32(sum * 1.8 / peaks.size()), 2500);
waveform.resize(peaks.size());
for (int32 i = 0, l = peaks.size(); i != l; ++i) {
waveform[i] = char(qMin(31U, uint32(qMin(peaks.at(i), peak)) * 31 / peak));
}
}
}
if (d->device) {
alcCaptureStop(d->device);
alcCaptureCloseDevice(d->device);
d->device = nullptr;
if (d->codecContext) {
avcodec_free_context(&d->codecContext);
d->codecContext = nullptr;
}
if (d->srcSamplesData) {
if (d->srcSamplesData[0]) {
av_freep(&d->srcSamplesData[0]);
}
av_freep(&d->srcSamplesData);
}
if (d->dstSamplesData) {
if (d->dstSamplesData[0]) {
av_freep(&d->dstSamplesData[0]);
}
av_freep(&d->dstSamplesData);
}
d->fullSamples = 0;
if (d->swrContext) {
swr_free(&d->swrContext);
d->swrContext = nullptr;
}
if (d->opened) {
avformat_close_input(&d->fmtContext);
d->opened = false;
}
if (d->ioContext) {
av_freep(&d->ioContext->buffer);
av_freep(&d->ioContext);
d->ioBuffer = nullptr;
} else if (d->ioBuffer) {
av_freep(&d->ioBuffer);
}
if (d->fmtContext) {
avformat_free_context(d->fmtContext);
d->fmtContext = nullptr;
}
d->fmt = nullptr;
d->stream = nullptr;
d->codec = nullptr;
d->lastUpdate = 0;
d->levelMax = 0;
d->dataPos = 0;
d->data.clear();
d->waveformMod = 0;
d->waveformPeak = 0;
d->waveform.clear();
}
if (needResult) emit done(result, waveform, samples);
}
void Instance::Inner::onTimeout() {
if (!d->device) {
_timer.stop();
return;
}
ALint samples;
alcGetIntegerv(d->device, ALC_CAPTURE_SAMPLES, sizeof(samples), &samples);
if (ErrorHappened(d->device)) {
onStop(false);
emit error();
return;
}
if (samples > 0) {
// Get samples from OpenAL
auto s = _captured.size();
auto news = s + static_cast<int>(samples * sizeof(short));
if (news / AudioVoiceMsgBufferSize > s / AudioVoiceMsgBufferSize) {
_captured.reserve(((news / AudioVoiceMsgBufferSize) + 1) * AudioVoiceMsgBufferSize);
}
_captured.resize(news);
alcCaptureSamples(d->device, (ALCvoid *)(_captured.data() + s), samples);
if (ErrorHappened(d->device)) {
onStop(false);
emit error();
return;
}
// Count new recording level and update view
auto skipSamples = kCaptureSkipDuration * kCaptureFrequency / 1000;
auto fadeSamples = kCaptureFadeInDuration * kCaptureFrequency / 1000;
auto levelindex = d->fullSamples + static_cast<int>(s / sizeof(short));
for (auto ptr = (const short*)(_captured.constData() + s), end = (const short*)(_captured.constData() + news); ptr < end; ++ptr, ++levelindex) {
if (levelindex > skipSamples) {
uint16 value = qAbs(*ptr);
if (levelindex < skipSamples + fadeSamples) {
value = qRound(value * float64(levelindex - skipSamples) / fadeSamples);
}
if (d->levelMax < value) {
d->levelMax = value;
}
}
}
qint32 samplesFull = d->fullSamples + _captured.size() / sizeof(short), samplesSinceUpdate = samplesFull - d->lastUpdate;
if (samplesSinceUpdate > AudioVoiceMsgUpdateView * kCaptureFrequency / 1000) {
emit updated(d->levelMax, samplesFull);
d->lastUpdate = samplesFull;
d->levelMax = 0;
}
// Write frames
int32 framesize = d->srcSamples * d->codecContext->channels * sizeof(short), encoded = 0;
while (uint32(_captured.size()) >= encoded + framesize + fadeSamples * sizeof(short)) {
processFrame(encoded, framesize);
encoded += framesize;
}
// Collapse the buffer
if (encoded > 0) {
int32 goodSize = _captured.size() - encoded;
memmove(_captured.data(), _captured.constData() + encoded, goodSize);
_captured.resize(goodSize);
}
} else {
DEBUG_LOG(("Audio Capture: no samples to capture."));
}
}
void Instance::Inner::processFrame(int32 offset, int32 framesize) {
// Prepare audio frame
if (framesize % sizeof(short)) { // in the middle of a sample
LOG(("Audio Error: Bad framesize in writeFrame() for capture, framesize %1, %2").arg(framesize));
onStop(false);
emit error();
return;
}
auto samplesCnt = static_cast<int>(framesize / sizeof(short));
int res = 0;
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
auto srcSamplesDataChannel = (short*)(_captured.data() + offset);
auto srcSamplesData = &srcSamplesDataChannel;
// memcpy(d->srcSamplesData[0], _captured.constData() + offset, framesize);
auto skipSamples = static_cast<int>(kCaptureSkipDuration * kCaptureFrequency / 1000);
auto fadeSamples = static_cast<int>(kCaptureFadeInDuration * kCaptureFrequency / 1000);
if (d->fullSamples < skipSamples + fadeSamples) {
int32 fadedCnt = qMin(samplesCnt, skipSamples + fadeSamples - d->fullSamples);
float64 coef = 1. / fadeSamples, fadedFrom = d->fullSamples - skipSamples;
short *ptr = srcSamplesDataChannel, *zeroEnd = ptr + qMin(samplesCnt, qMax(0, skipSamples - d->fullSamples)), *end = ptr + fadedCnt;
for (; ptr != zeroEnd; ++ptr, ++fadedFrom) {
*ptr = 0;
}
for (; ptr != end; ++ptr, ++fadedFrom) {
*ptr = qRound(fadedFrom * coef * *ptr);
}
}
d->waveform.reserve(d->waveform.size() + (samplesCnt / d->waveformEach) + 1);
for (short *ptr = srcSamplesDataChannel, *end = ptr + samplesCnt; ptr != end; ++ptr) {
uint16 value = qAbs(*ptr);
if (d->waveformPeak < value) {
d->waveformPeak = value;
}
if (++d->waveformMod == d->waveformEach) {
d->waveformMod -= d->waveformEach;
d->waveform.push_back(uchar(d->waveformPeak / 256));
d->waveformPeak = 0;
}
}
// Convert to final format
d->dstSamples = av_rescale_rnd(swr_get_delay(d->swrContext, d->codecContext->sample_rate) + d->srcSamples, d->codecContext->sample_rate, d->codecContext->sample_rate, AV_ROUND_UP);
if (d->dstSamples > d->maxDstSamples) {
d->maxDstSamples = d->dstSamples;
av_freep(&d->dstSamplesData[0]);
if ((res = av_samples_alloc(d->dstSamplesData, 0, d->codecContext->channels, d->dstSamples, d->codecContext->sample_fmt, 1)) < 0) {
LOG(("Audio Error: Unable to av_samples_alloc for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
onStop(false);
emit error();
return;
}
d->dstSamplesSize = av_samples_get_buffer_size(0, d->codecContext->channels, d->maxDstSamples, d->codecContext->sample_fmt, 0);
}
if ((res = swr_convert(d->swrContext, d->dstSamplesData, d->dstSamples, (const uint8_t **)srcSamplesData, d->srcSamples)) < 0) {
LOG(("Audio Error: Unable to swr_convert for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
onStop(false);
emit error();
return;
}
// Write audio frame
AVFrame *frame = av_frame_alloc();
frame->nb_samples = d->dstSamples;
frame->pts = av_rescale_q(d->fullSamples, AVRational { 1, d->codecContext->sample_rate }, d->codecContext->time_base);
avcodec_fill_audio_frame(frame, d->codecContext->channels, d->codecContext->sample_fmt, d->dstSamplesData[0], d->dstSamplesSize, 0);
writeFrame(frame);
d->fullSamples += samplesCnt;
av_frame_free(&frame);
}
void Instance::Inner::writeFrame(AVFrame *frame) {
int res = 0;
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
res = avcodec_send_frame(d->codecContext, frame);
if (res == AVERROR(EAGAIN)) {
int packetsWritten = writePackets();
if (packetsWritten < 0) {
if (frame && packetsWritten == AVERROR_EOF) {
LOG(("Audio Error: EOF in packets received when EAGAIN was got in avcodec_send_frame()"));
onStop(false);
emit error();
}
return;
} else if (!packetsWritten) {
LOG(("Audio Error: No packets received when EAGAIN was got in avcodec_send_frame()"));
onStop(false);
emit error();
return;
}
res = avcodec_send_frame(d->codecContext, frame);
}
if (res < 0) {
LOG(("Audio Error: Unable to avcodec_send_frame for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
onStop(false);
emit error();
return;
}
if (!frame) { // drain
if ((res = writePackets()) != AVERROR_EOF) {
LOG(("Audio Error: not EOF in packets received when draining the codec, result %1").arg(res));
onStop(false);
emit error();
}
}
}
int Instance::Inner::writePackets() {
AVPacket pkt;
memset(&pkt, 0, sizeof(pkt)); // data and size must be 0;
int res = 0;
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
int written = 0;
do {
av_init_packet(&pkt);
if ((res = avcodec_receive_packet(d->codecContext, &pkt)) < 0) {
if (res == AVERROR(EAGAIN)) {
return written;
} else if (res == AVERROR_EOF) {
return res;
}
LOG(("Audio Error: Unable to avcodec_receive_packet for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
onStop(false);
emit error();
return res;
}
av_packet_rescale_ts(&pkt, d->codecContext->time_base, d->stream->time_base);
pkt.stream_index = d->stream->index;
if ((res = av_interleaved_write_frame(d->fmtContext, &pkt)) < 0) {
LOG(("Audio Error: Unable to av_interleaved_write_frame for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
onStop(false);
emit error();
return -1;
}
++written;
av_packet_unref(&pkt);
} while (true);
return written;
}
} // namespace Capture
} // namespace Media