tdesktop/Telegram/SourceFiles/media/media_child_ffmpeg_loader.cpp

215 lines
8.1 KiB
C++

/*
This file is part of Telegram Desktop,
the official desktop version of Telegram messaging app, see https://telegram.org
Telegram Desktop is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
It is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
In addition, as a special exception, the copyright holders give permission
to link the code of portions of this program with the OpenSSL library.
Full license: https://github.com/telegramdesktop/tdesktop/blob/master/LICENSE
Copyright (c) 2014-2016 John Preston, https://desktop.telegram.org
*/
#include "stdafx.h"
#include "media/media_child_ffmpeg_loader.h"
constexpr AVSampleFormat AudioToFormat = AV_SAMPLE_FMT_S16;
constexpr int64_t AudioToChannelLayout = AV_CH_LAYOUT_STEREO;
constexpr int32 AudioToChannels = 2;
VideoSoundData::~VideoSoundData() {
if (context) {
avcodec_close(context);
avcodec_free_context(&context);
context = nullptr;
}
}
ChildFFMpegLoader::ChildFFMpegLoader(uint64 videoPlayId, std_::unique_ptr<VideoSoundData> &&data) : AudioPlayerLoader(FileLocation(), QByteArray())
, _videoPlayId(videoPlayId)
, _parentData(std_::move(data)) {
_frame = av_frame_alloc();
}
bool ChildFFMpegLoader::open(qint64 &position) {
int res = 0;
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
uint64_t layout = _parentData->context->channel_layout;
_inputFormat = _parentData->context->sample_fmt;
switch (layout) {
case AV_CH_LAYOUT_MONO:
switch (_inputFormat) {
case AV_SAMPLE_FMT_U8:
case AV_SAMPLE_FMT_U8P: _format = AL_FORMAT_MONO8; _sampleSize = 1; break;
case AV_SAMPLE_FMT_S16:
case AV_SAMPLE_FMT_S16P: _format = AL_FORMAT_MONO16; _sampleSize = sizeof(uint16); break;
default:
_sampleSize = -1; // convert needed
break;
}
break;
case AV_CH_LAYOUT_STEREO:
switch (_inputFormat) {
case AV_SAMPLE_FMT_U8: _format = AL_FORMAT_STEREO8; _sampleSize = 2; break;
case AV_SAMPLE_FMT_S16: _format = AL_FORMAT_STEREO16; _sampleSize = 2 * sizeof(uint16); break;
default:
_sampleSize = -1; // convert needed
break;
}
break;
default:
_sampleSize = -1; // convert needed
break;
}
if (_parentData->frequency != 44100 && _parentData->frequency != 48000) {
_sampleSize = -1; // convert needed
}
if (_sampleSize < 0) {
_swrContext = swr_alloc();
if (!_swrContext) {
LOG(("Audio Error: Unable to swr_alloc for file '%1', data size '%2'").arg(file.name()).arg(data.size()));
return false;
}
int64_t src_ch_layout = layout, dst_ch_layout = AudioToChannelLayout;
_srcRate = _parentData->frequency;
AVSampleFormat src_sample_fmt = _inputFormat, dst_sample_fmt = AudioToFormat;
_dstRate = (_parentData->frequency != 44100 && _parentData->frequency != 48000) ? AudioVoiceMsgFrequency : _parentData->frequency;
av_opt_set_int(_swrContext, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(_swrContext, "in_sample_rate", _srcRate, 0);
av_opt_set_sample_fmt(_swrContext, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_int(_swrContext, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(_swrContext, "out_sample_rate", _dstRate, 0);
av_opt_set_sample_fmt(_swrContext, "out_sample_fmt", dst_sample_fmt, 0);
if ((res = swr_init(_swrContext)) < 0) {
LOG(("Audio Error: Unable to swr_init for file '%1', data size '%2', error %3, %4").arg(file.name()).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return false;
}
_sampleSize = AudioToChannels * sizeof(short);
_parentData->frequency = _dstRate;
_parentData->length = av_rescale_rnd(_parentData->length, _dstRate, _srcRate, AV_ROUND_UP);
position = av_rescale_rnd(position, _dstRate, _srcRate, AV_ROUND_DOWN);
_format = AL_FORMAT_STEREO16;
_maxResampleSamples = av_rescale_rnd(AVBlockSize / _sampleSize, _dstRate, _srcRate, AV_ROUND_UP);
if ((res = av_samples_alloc_array_and_samples(&_dstSamplesData, 0, AudioToChannels, _maxResampleSamples, AudioToFormat, 0)) < 0) {
LOG(("Audio Error: Unable to av_samples_alloc for file '%1', data size '%2', error %3, %4").arg(file.name()).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return false;
}
}
return true;
}
AudioPlayerLoader::ReadResult ChildFFMpegLoader::readMore(QByteArray &result, int64 &samplesAdded) {
int res;
av_frame_unref(_frame);
res = avcodec_receive_frame(_parentData->context, _frame);
if (res >= 0) {
return readFromReadyFrame(result, samplesAdded);
}
if (res == AVERROR_EOF) {
return ReadResult::EndOfFile;
} else if (res != AVERROR(EAGAIN)) {
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
LOG(("Audio Error: Unable to avcodec_receive_frame() file '%1', data size '%2', error %3, %4").arg(file.name()).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return ReadResult::Error;
}
if (_queue.isEmpty()) {
return _eofReached ? ReadResult::EndOfFile : ReadResult::Wait;
}
AVPacket packet;
FFMpeg::packetFromDataWrap(packet, _queue.dequeue());
_eofReached = FFMpeg::isNullPacket(packet);
if (_eofReached) {
avcodec_send_packet(_parentData->context, nullptr); // drain
return ReadResult::Ok;
}
res = avcodec_send_packet(_parentData->context, &packet);
if (res < 0) {
FFMpeg::freePacket(&packet);
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
LOG(("Audio Error: Unable to avcodec_send_packet() file '%1', data size '%2', error %3, %4").arg(file.name()).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
// There is a sample voice message where skipping such packet
// results in a crash (read_access to nullptr) in swr_convert().
//if (res == AVERROR_INVALIDDATA) {
// return ReadResult::NotYet; // try to skip bad packet
//}
return ReadResult::Error;
}
FFMpeg::freePacket(&packet);
return ReadResult::Ok;
}
AudioPlayerLoader::ReadResult ChildFFMpegLoader::readFromReadyFrame(QByteArray &result, int64 &samplesAdded) {
int res = 0;
if (_dstSamplesData) { // convert needed
int64_t dstSamples = av_rescale_rnd(swr_get_delay(_swrContext, _srcRate) + _frame->nb_samples, _dstRate, _srcRate, AV_ROUND_UP);
if (dstSamples > _maxResampleSamples) {
_maxResampleSamples = dstSamples;
av_free(_dstSamplesData[0]);
if ((res = av_samples_alloc(_dstSamplesData, 0, AudioToChannels, _maxResampleSamples, AudioToFormat, 1)) < 0) {
_dstSamplesData[0] = 0;
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
LOG(("Audio Error: Unable to av_samples_alloc for file '%1', data size '%2', error %3, %4").arg(file.name()).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return ReadResult::Error;
}
}
if ((res = swr_convert(_swrContext, _dstSamplesData, dstSamples, (const uint8_t**)_frame->extended_data, _frame->nb_samples)) < 0) {
char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
LOG(("Audio Error: Unable to swr_convert for file '%1', data size '%2', error %3, %4").arg(file.name()).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
return ReadResult::Error;
}
int32 resultLen = av_samples_get_buffer_size(0, AudioToChannels, res, AudioToFormat, 1);
result.append((const char*)_dstSamplesData[0], resultLen);
samplesAdded += resultLen / _sampleSize;
} else {
result.append((const char*)_frame->extended_data[0], _frame->nb_samples * _sampleSize);
samplesAdded += _frame->nb_samples;
}
return ReadResult::Ok;
}
void ChildFFMpegLoader::enqueuePackets(QQueue<FFMpeg::AVPacketDataWrap> &packets) {
_queue += std_::move(packets);
packets.clear();
}
ChildFFMpegLoader::~ChildFFMpegLoader() {
auto queue = createAndSwap(_queue);
for (auto &packetData : queue) {
AVPacket packet;
FFMpeg::packetFromDataWrap(packet, packetData);
FFMpeg::freePacket(&packet);
}
if (_swrContext) swr_free(&_swrContext);
if (_dstSamplesData) {
if (_dstSamplesData[0]) {
av_freep(&_dstSamplesData[0]);
}
av_freep(&_dstSamplesData);
}
av_frame_free(&_frame);
}