710 lines
22 KiB
C++
710 lines
22 KiB
C++
/*
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This file is part of Telegram Desktop,
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the official desktop version of Telegram messaging app, see https://telegram.org
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Telegram Desktop is free software: you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation, either version 3 of the License, or
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(at your option) any later version.
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It is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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In addition, as a special exception, the copyright holders give permission
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to link the code of portions of this program with the OpenSSL library.
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Full license: https://github.com/telegramdesktop/tdesktop/blob/master/LICENSE
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Copyright (c) 2014-2017 John Preston, https://desktop.telegram.org
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*/
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#include "media/media_audio_capture.h"
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#include "media/media_audio_ffmpeg_loader.h"
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#include <AL/al.h>
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#include <AL/alc.h>
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#include <AL/alext.h>
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#include <numeric>
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namespace Media {
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namespace Capture {
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namespace {
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constexpr auto kCaptureFrequency = Player::kDefaultFrequency;
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constexpr auto kCaptureSkipDuration = TimeMs(400);
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constexpr auto kCaptureFadeInDuration = TimeMs(300);
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Instance *CaptureInstance = nullptr;
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bool ErrorHappened(ALCdevice *device) {
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ALenum errCode;
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if ((errCode = alcGetError(device)) != ALC_NO_ERROR) {
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LOG(("Audio Capture Error: %1, %2").arg(errCode).arg((const char *)alcGetString(device, errCode)));
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return true;
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}
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return false;
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}
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} // namespace
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void Start() {
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t_assert(CaptureInstance == nullptr);
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CaptureInstance = new Instance();
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instance()->check();
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}
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void Finish() {
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delete base::take(CaptureInstance);
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}
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Instance::Instance() : _inner(new Inner(&_thread)) {
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CaptureInstance = this;
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connect(this, SIGNAL(start()), _inner, SLOT(onStart()));
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connect(this, SIGNAL(stop(bool)), _inner, SLOT(onStop(bool)));
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connect(_inner, SIGNAL(done(QByteArray, VoiceWaveform, qint32)), this, SIGNAL(done(QByteArray, VoiceWaveform, qint32)));
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connect(_inner, SIGNAL(updated(quint16, qint32)), this, SIGNAL(updated(quint16, qint32)));
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connect(_inner, SIGNAL(error()), this, SIGNAL(error()));
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connect(&_thread, SIGNAL(started()), _inner, SLOT(onInit()));
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connect(&_thread, SIGNAL(finished()), _inner, SLOT(deleteLater()));
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_thread.start();
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}
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void Instance::check() {
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_available = false;
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if (auto defaultDevice = alcGetString(0, ALC_CAPTURE_DEFAULT_DEVICE_SPECIFIER)) {
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if (auto device = alcCaptureOpenDevice(defaultDevice, kCaptureFrequency, AL_FORMAT_MONO16, kCaptureFrequency / 5)) {
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auto error = ErrorHappened(device);
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alcCaptureCloseDevice(device);
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_available = !error;
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}
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}
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}
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Instance::~Instance() {
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_inner = nullptr;
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_thread.quit();
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_thread.wait();
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}
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Instance *instance() {
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return CaptureInstance;
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}
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struct Instance::Inner::Private {
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ALCdevice *device = nullptr;
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AVOutputFormat *fmt = nullptr;
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uchar *ioBuffer = nullptr;
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AVIOContext *ioContext = nullptr;
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AVFormatContext *fmtContext = nullptr;
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AVStream *stream = nullptr;
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AVCodec *codec = nullptr;
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AVCodecContext *codecContext = nullptr;
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bool opened = false;
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int srcSamples = 0;
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int dstSamples = 0;
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int maxDstSamples = 0;
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int dstSamplesSize = 0;
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int fullSamples = 0;
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uint8_t **srcSamplesData = nullptr;
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uint8_t **dstSamplesData = nullptr;
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SwrContext *swrContext = nullptr;
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int32 lastUpdate = 0;
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uint16 levelMax = 0;
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QByteArray data;
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int32 dataPos = 0;
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int64 waveformMod = 0;
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int64 waveformEach = (kCaptureFrequency / 100);
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uint16 waveformPeak = 0;
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QVector<uchar> waveform;
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static int _read_data(void *opaque, uint8_t *buf, int buf_size) {
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auto l = reinterpret_cast<Private*>(opaque);
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int32 nbytes = qMin(l->data.size() - l->dataPos, int32(buf_size));
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if (nbytes <= 0) {
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return 0;
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}
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memcpy(buf, l->data.constData() + l->dataPos, nbytes);
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l->dataPos += nbytes;
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return nbytes;
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}
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static int _write_data(void *opaque, uint8_t *buf, int buf_size) {
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auto l = reinterpret_cast<Private*>(opaque);
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if (buf_size <= 0) return 0;
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if (l->dataPos + buf_size > l->data.size()) l->data.resize(l->dataPos + buf_size);
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memcpy(l->data.data() + l->dataPos, buf, buf_size);
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l->dataPos += buf_size;
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return buf_size;
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}
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static int64_t _seek_data(void *opaque, int64_t offset, int whence) {
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auto l = reinterpret_cast<Private*>(opaque);
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int32 newPos = -1;
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switch (whence) {
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case SEEK_SET: newPos = offset; break;
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case SEEK_CUR: newPos = l->dataPos + offset; break;
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case SEEK_END: newPos = l->data.size() + offset; break;
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case AVSEEK_SIZE: {
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// Special whence for determining filesize without any seek.
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return l->data.size();
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} break;
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}
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if (newPos < 0) {
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return -1;
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}
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l->dataPos = newPos;
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return l->dataPos;
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}
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};
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Instance::Inner::Inner(QThread *thread) : d(new Private()) {
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moveToThread(thread);
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_timer.moveToThread(thread);
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connect(&_timer, SIGNAL(timeout()), this, SLOT(onTimeout()));
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}
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Instance::Inner::~Inner() {
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onStop(false);
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delete d;
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}
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void Instance::Inner::onInit() {
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}
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void Instance::Inner::onStart() {
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// Start OpenAL Capture
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const ALCchar *dName = alcGetString(0, ALC_CAPTURE_DEFAULT_DEVICE_SPECIFIER);
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DEBUG_LOG(("Audio Info: Capture device name '%1'").arg(dName));
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d->device = alcCaptureOpenDevice(dName, kCaptureFrequency, AL_FORMAT_MONO16, kCaptureFrequency / 5);
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if (!d->device) {
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LOG(("Audio Error: capture device not present!"));
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emit error();
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return;
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}
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alcCaptureStart(d->device);
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if (ErrorHappened(d->device)) {
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alcCaptureCloseDevice(d->device);
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d->device = nullptr;
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emit error();
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return;
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}
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// Create encoding context
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d->ioBuffer = (uchar*)av_malloc(AVBlockSize);
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d->ioContext = avio_alloc_context(d->ioBuffer, AVBlockSize, 1, static_cast<void*>(d), &Private::_read_data, &Private::_write_data, &Private::_seek_data);
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int res = 0;
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char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
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AVOutputFormat *fmt = 0;
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while ((fmt = av_oformat_next(fmt))) {
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if (fmt->name == qstr("opus")) {
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break;
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}
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}
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if (!fmt) {
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LOG(("Audio Error: Unable to find opus AVOutputFormat for capture"));
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onStop(false);
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emit error();
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return;
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}
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if ((res = avformat_alloc_output_context2(&d->fmtContext, fmt, 0, 0)) < 0) {
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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)));
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onStop(false);
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emit error();
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return;
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}
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d->fmtContext->pb = d->ioContext;
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d->fmtContext->flags |= AVFMT_FLAG_CUSTOM_IO;
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d->opened = true;
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// Add audio stream
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d->codec = avcodec_find_encoder(fmt->audio_codec);
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if (!d->codec) {
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LOG(("Audio Error: Unable to avcodec_find_encoder for capture"));
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onStop(false);
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emit error();
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return;
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}
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d->stream = avformat_new_stream(d->fmtContext, d->codec);
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if (!d->stream) {
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LOG(("Audio Error: Unable to avformat_new_stream for capture"));
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onStop(false);
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emit error();
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return;
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}
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d->stream->id = d->fmtContext->nb_streams - 1;
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d->codecContext = avcodec_alloc_context3(d->codec);
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if (!d->codecContext) {
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LOG(("Audio Error: Unable to avcodec_alloc_context3 for capture"));
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onStop(false);
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emit error();
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return;
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}
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av_opt_set_int(d->codecContext, "refcounted_frames", 1, 0);
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d->codecContext->sample_fmt = AV_SAMPLE_FMT_FLTP;
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d->codecContext->bit_rate = 64000;
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d->codecContext->channel_layout = AV_CH_LAYOUT_MONO;
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d->codecContext->sample_rate = kCaptureFrequency;
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d->codecContext->channels = 1;
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if (d->fmtContext->oformat->flags & AVFMT_GLOBALHEADER) {
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d->codecContext->flags |= CODEC_FLAG_GLOBAL_HEADER;
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}
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// Open audio stream
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if ((res = avcodec_open2(d->codecContext, d->codec, nullptr)) < 0) {
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LOG(("Audio Error: Unable to avcodec_open2 for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
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onStop(false);
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emit error();
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return;
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}
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// Alloc source samples
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d->srcSamples = (d->codecContext->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE) ? 10000 : d->codecContext->frame_size;
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//if ((res = av_samples_alloc_array_and_samples(&d->srcSamplesData, 0, d->codecContext->channels, d->srcSamples, d->codecContext->sample_fmt, 0)) < 0) {
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// 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)));
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// onStop(false);
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// emit error();
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// return;
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//}
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// Using _captured directly
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// Prepare resampling
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d->swrContext = swr_alloc();
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if (!d->swrContext) {
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fprintf(stderr, "Could not allocate resampler context\n");
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exit(1);
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}
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av_opt_set_int(d->swrContext, "in_channel_count", d->codecContext->channels, 0);
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av_opt_set_int(d->swrContext, "in_sample_rate", d->codecContext->sample_rate, 0);
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av_opt_set_sample_fmt(d->swrContext, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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av_opt_set_int(d->swrContext, "out_channel_count", d->codecContext->channels, 0);
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av_opt_set_int(d->swrContext, "out_sample_rate", d->codecContext->sample_rate, 0);
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av_opt_set_sample_fmt(d->swrContext, "out_sample_fmt", d->codecContext->sample_fmt, 0);
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if ((res = swr_init(d->swrContext)) < 0) {
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LOG(("Audio Error: Unable to swr_init for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
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onStop(false);
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emit error();
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return;
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}
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d->maxDstSamples = d->srcSamples;
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if ((res = av_samples_alloc_array_and_samples(&d->dstSamplesData, 0, d->codecContext->channels, d->maxDstSamples, d->codecContext->sample_fmt, 0)) < 0) {
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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)));
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onStop(false);
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emit error();
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return;
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}
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d->dstSamplesSize = av_samples_get_buffer_size(0, d->codecContext->channels, d->maxDstSamples, d->codecContext->sample_fmt, 0);
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if ((res = avcodec_parameters_from_context(d->stream->codecpar, d->codecContext)) < 0) {
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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)));
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onStop(false);
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emit error();
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return;
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}
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// Write file header
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if ((res = avformat_write_header(d->fmtContext, 0)) < 0) {
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LOG(("Audio Error: Unable to avformat_write_header for capture, error %1, %2").arg(res).arg(av_make_error_string(err, sizeof(err), res)));
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onStop(false);
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emit error();
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return;
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}
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_timer.start(50);
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_captured.clear();
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_captured.reserve(AudioVoiceMsgBufferSize);
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DEBUG_LOG(("Audio Capture: started!"));
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}
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void Instance::Inner::onStop(bool needResult) {
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if (!_timer.isActive()) return; // in onStop() already
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_timer.stop();
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if (d->device) {
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alcCaptureStop(d->device);
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onTimeout(); // get last data
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}
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// Write what is left
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if (!_captured.isEmpty()) {
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auto fadeSamples = kCaptureFadeInDuration * kCaptureFrequency / 1000;
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auto capturedSamples = static_cast<int>(_captured.size() / sizeof(short));
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if ((_captured.size() % sizeof(short)) || (d->fullSamples + capturedSamples < kCaptureFrequency) || (capturedSamples < fadeSamples)) {
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d->fullSamples = 0;
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d->dataPos = 0;
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d->data.clear();
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d->waveformMod = 0;
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d->waveformPeak = 0;
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d->waveform.clear();
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} else {
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float64 coef = 1. / fadeSamples, fadedFrom = 0;
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for (short *ptr = ((short*)_captured.data()) + capturedSamples, *end = ptr - fadeSamples; ptr != end; ++fadedFrom) {
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--ptr;
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*ptr = qRound(fadedFrom * coef * *ptr);
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}
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if (capturedSamples % d->srcSamples) {
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int32 s = _captured.size();
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_captured.resize(s + (d->srcSamples - (capturedSamples % d->srcSamples)) * sizeof(short));
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memset(_captured.data() + s, 0, _captured.size() - s);
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}
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int32 framesize = d->srcSamples * d->codecContext->channels * sizeof(short), encoded = 0;
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while (_captured.size() >= encoded + framesize) {
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processFrame(encoded, framesize);
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encoded += framesize;
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}
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writeFrame(nullptr); // drain the codec
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if (encoded != _captured.size()) {
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d->fullSamples = 0;
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d->dataPos = 0;
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d->data.clear();
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d->waveformMod = 0;
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d->waveformPeak = 0;
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d->waveform.clear();
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}
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}
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}
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DEBUG_LOG(("Audio Capture: stopping (need result: %1), size: %2, samples: %3").arg(Logs::b(needResult)).arg(d->data.size()).arg(d->fullSamples));
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_captured = QByteArray();
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// Finish stream
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if (d->device) {
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av_write_trailer(d->fmtContext);
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}
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QByteArray result = d->fullSamples ? d->data : QByteArray();
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VoiceWaveform waveform;
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qint32 samples = d->fullSamples;
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if (samples && !d->waveform.isEmpty()) {
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int64 count = d->waveform.size(), sum = 0;
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if (count >= Player::kWaveformSamplesCount) {
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QVector<uint16> peaks;
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peaks.reserve(Player::kWaveformSamplesCount);
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uint16 peak = 0;
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for (int32 i = 0; i < count; ++i) {
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uint16 sample = uint16(d->waveform.at(i)) * 256;
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if (peak < sample) {
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peak = sample;
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}
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sum += Player::kWaveformSamplesCount;
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if (sum >= count) {
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sum -= count;
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peaks.push_back(peak);
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peak = 0;
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}
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}
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auto sum = std::accumulate(peaks.cbegin(), peaks.cend(), 0LL);
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peak = qMax(int32(sum * 1.8 / peaks.size()), 2500);
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waveform.resize(peaks.size());
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for (int32 i = 0, l = peaks.size(); i != l; ++i) {
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waveform[i] = char(qMin(31U, uint32(qMin(peaks.at(i), peak)) * 31 / peak));
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}
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}
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}
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if (d->device) {
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alcCaptureStop(d->device);
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alcCaptureCloseDevice(d->device);
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d->device = nullptr;
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if (d->codecContext) {
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avcodec_free_context(&d->codecContext);
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d->codecContext = nullptr;
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}
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if (d->srcSamplesData) {
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if (d->srcSamplesData[0]) {
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av_freep(&d->srcSamplesData[0]);
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}
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av_freep(&d->srcSamplesData);
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}
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if (d->dstSamplesData) {
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if (d->dstSamplesData[0]) {
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av_freep(&d->dstSamplesData[0]);
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}
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av_freep(&d->dstSamplesData);
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}
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d->fullSamples = 0;
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if (d->swrContext) {
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swr_free(&d->swrContext);
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d->swrContext = nullptr;
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}
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if (d->opened) {
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avformat_close_input(&d->fmtContext);
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d->opened = false;
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}
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if (d->ioContext) {
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av_freep(&d->ioContext->buffer);
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av_freep(&d->ioContext);
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d->ioBuffer = nullptr;
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} else if (d->ioBuffer) {
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av_freep(&d->ioBuffer);
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}
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if (d->fmtContext) {
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avformat_free_context(d->fmtContext);
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d->fmtContext = nullptr;
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}
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d->fmt = nullptr;
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d->stream = nullptr;
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d->codec = nullptr;
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d->lastUpdate = 0;
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d->levelMax = 0;
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d->dataPos = 0;
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d->data.clear();
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d->waveformMod = 0;
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d->waveformPeak = 0;
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d->waveform.clear();
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}
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if (needResult) emit done(result, waveform, samples);
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}
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void Instance::Inner::onTimeout() {
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if (!d->device) {
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_timer.stop();
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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
|