213 lines
7.9 KiB
C++
213 lines
7.9 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) 2014-2016 John Preston, https://desktop.telegram.org
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*/
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#include "stdafx.h"
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#include "media/media_child_ffmpeg_loader.h"
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constexpr AVSampleFormat AudioToFormat = AV_SAMPLE_FMT_S16;
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constexpr int64_t AudioToChannelLayout = AV_CH_LAYOUT_STEREO;
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constexpr int32 AudioToChannels = 2;
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VideoSoundData::~VideoSoundData() {
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if (context) {
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avcodec_close(context);
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avcodec_free_context(&context);
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context = nullptr;
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}
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}
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ChildFFMpegLoader::ChildFFMpegLoader(uint64 videoPlayId, std_::unique_ptr<VideoSoundData> &&data) : AudioPlayerLoader(FileLocation(), QByteArray())
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, _videoPlayId(videoPlayId)
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, _parentData(std_::move(data)) {
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_frame = av_frame_alloc();
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}
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bool ChildFFMpegLoader::open(qint64 &position) {
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int res = 0;
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char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
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uint64_t layout = _parentData->context->channel_layout;
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_inputFormat = _parentData->context->sample_fmt;
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switch (layout) {
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case AV_CH_LAYOUT_MONO:
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switch (_inputFormat) {
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case AV_SAMPLE_FMT_U8:
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case AV_SAMPLE_FMT_U8P: _format = AL_FORMAT_MONO8; _sampleSize = 1; break;
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case AV_SAMPLE_FMT_S16:
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case AV_SAMPLE_FMT_S16P: _format = AL_FORMAT_MONO16; _sampleSize = sizeof(uint16); break;
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default:
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_sampleSize = -1; // convert needed
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break;
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}
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break;
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case AV_CH_LAYOUT_STEREO:
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switch (_inputFormat) {
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case AV_SAMPLE_FMT_U8: _format = AL_FORMAT_STEREO8; _sampleSize = 2; break;
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case AV_SAMPLE_FMT_S16: _format = AL_FORMAT_STEREO16; _sampleSize = 2 * sizeof(uint16); break;
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default:
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_sampleSize = -1; // convert needed
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break;
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}
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break;
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default:
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_sampleSize = -1; // convert needed
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break;
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}
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if (_parentData->frequency != 44100 && _parentData->frequency != 48000) {
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_sampleSize = -1; // convert needed
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}
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if (_sampleSize < 0) {
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_swrContext = swr_alloc();
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if (!_swrContext) {
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LOG(("Audio Error: Unable to swr_alloc for file '%1', data size '%2'").arg(file.name()).arg(data.size()));
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return false;
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}
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int64_t src_ch_layout = layout, dst_ch_layout = AudioToChannelLayout;
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_srcRate = _parentData->frequency;
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AVSampleFormat src_sample_fmt = _inputFormat, dst_sample_fmt = AudioToFormat;
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_dstRate = (_parentData->frequency != 44100 && _parentData->frequency != 48000) ? AudioVoiceMsgFrequency : _parentData->frequency;
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av_opt_set_int(_swrContext, "in_channel_layout", src_ch_layout, 0);
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av_opt_set_int(_swrContext, "in_sample_rate", _srcRate, 0);
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av_opt_set_sample_fmt(_swrContext, "in_sample_fmt", src_sample_fmt, 0);
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av_opt_set_int(_swrContext, "out_channel_layout", dst_ch_layout, 0);
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av_opt_set_int(_swrContext, "out_sample_rate", _dstRate, 0);
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av_opt_set_sample_fmt(_swrContext, "out_sample_fmt", dst_sample_fmt, 0);
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if ((res = swr_init(_swrContext)) < 0) {
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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)));
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return false;
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}
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_sampleSize = AudioToChannels * sizeof(short);
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_parentData->frequency = _dstRate;
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_parentData->length = av_rescale_rnd(_parentData->length, _dstRate, _srcRate, AV_ROUND_UP);
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position = av_rescale_rnd(position, _dstRate, _srcRate, AV_ROUND_DOWN);
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_format = AL_FORMAT_STEREO16;
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_maxResampleSamples = av_rescale_rnd(AVBlockSize / _sampleSize, _dstRate, _srcRate, AV_ROUND_UP);
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if ((res = av_samples_alloc_array_and_samples(&_dstSamplesData, 0, AudioToChannels, _maxResampleSamples, AudioToFormat, 0)) < 0) {
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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)));
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return false;
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}
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}
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return true;
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}
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AudioPlayerLoader::ReadResult ChildFFMpegLoader::readMore(QByteArray &result, int64 &samplesAdded) {
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int res;
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av_frame_unref(_frame);
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res = avcodec_receive_frame(_parentData->context, _frame);
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if (res >= 0) {
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return readFromReadyFrame(result, samplesAdded);
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}
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if (res == AVERROR_EOF) {
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return ReadResult::EndOfFile;
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} else if (res != AVERROR(EAGAIN)) {
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char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
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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)));
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return ReadResult::Error;
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}
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if (_queue.isEmpty()) {
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return _eofReached ? ReadResult::EndOfFile : ReadResult::Wait;
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}
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AVPacket packet;
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FFMpeg::packetFromDataWrap(packet, _queue.dequeue());
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_eofReached = FFMpeg::isNullPacket(packet);
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if (_eofReached) {
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avcodec_send_packet(_parentData->context, nullptr); // drain
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return ReadResult::Ok;
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}
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res = avcodec_send_packet(_parentData->context, &packet);
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if (res < 0) {
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FFMpeg::freePacket(&packet);
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char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
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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)));
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if (res == AVERROR_INVALIDDATA) {
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return ReadResult::NotYet; // try to skip bad packet
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}
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return ReadResult::Error;
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}
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FFMpeg::freePacket(&packet);
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return ReadResult::Ok;
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}
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AudioPlayerLoader::ReadResult ChildFFMpegLoader::readFromReadyFrame(QByteArray &result, int64 &samplesAdded) {
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int res = 0;
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if (_dstSamplesData) { // convert needed
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int64_t dstSamples = av_rescale_rnd(swr_get_delay(_swrContext, _srcRate) + _frame->nb_samples, _dstRate, _srcRate, AV_ROUND_UP);
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if (dstSamples > _maxResampleSamples) {
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_maxResampleSamples = dstSamples;
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av_free(_dstSamplesData[0]);
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if ((res = av_samples_alloc(_dstSamplesData, 0, AudioToChannels, _maxResampleSamples, AudioToFormat, 1)) < 0) {
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_dstSamplesData[0] = 0;
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char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
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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)));
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return ReadResult::Error;
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}
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}
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if ((res = swr_convert(_swrContext, _dstSamplesData, dstSamples, (const uint8_t**)_frame->extended_data, _frame->nb_samples)) < 0) {
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char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
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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)));
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return ReadResult::Error;
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}
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int32 resultLen = av_samples_get_buffer_size(0, AudioToChannels, res, AudioToFormat, 1);
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result.append((const char*)_dstSamplesData[0], resultLen);
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samplesAdded += resultLen / _sampleSize;
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} else {
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result.append((const char*)_frame->extended_data[0], _frame->nb_samples * _sampleSize);
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samplesAdded += _frame->nb_samples;
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}
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return ReadResult::Ok;
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}
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void ChildFFMpegLoader::enqueuePackets(QQueue<FFMpeg::AVPacketDataWrap> &packets) {
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_queue += std_::move(packets);
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packets.clear();
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}
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ChildFFMpegLoader::~ChildFFMpegLoader() {
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auto queue = createAndSwap(_queue);
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for (auto &packetData : queue) {
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AVPacket packet;
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FFMpeg::packetFromDataWrap(packet, packetData);
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FFMpeg::freePacket(&packet);
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}
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if (_swrContext) swr_free(&_swrContext);
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if (_dstSamplesData) {
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if (_dstSamplesData[0]) {
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av_freep(&_dstSamplesData[0]);
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}
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av_freep(&_dstSamplesData);
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}
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av_frame_free(&_frame);
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}
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