mirror of
https://github.com/telegramdesktop/tdesktop
synced 2025-04-01 00:08:02 +00:00
9fe714189d
Video sync and frame duration count improved. Seek in not 44100 and not 48000 hz audio streams fixed.
304 lines
11 KiB
C++
304 lines
11 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_audio_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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bool AbstractFFMpegLoader::open(qint64 &position) {
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if (!AudioPlayerLoader::openFile()) {
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return false;
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}
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int res = 0;
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char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
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ioBuffer = (uchar*)av_malloc(AVBlockSize);
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if (data.isEmpty()) {
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ioContext = avio_alloc_context(ioBuffer, AVBlockSize, 0, reinterpret_cast<void*>(this), &AbstractFFMpegLoader::_read_file, 0, &AbstractFFMpegLoader::_seek_file);
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} else {
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ioContext = avio_alloc_context(ioBuffer, AVBlockSize, 0, reinterpret_cast<void*>(this), &AbstractFFMpegLoader::_read_data, 0, &AbstractFFMpegLoader::_seek_data);
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}
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fmtContext = avformat_alloc_context();
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if (!fmtContext) {
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DEBUG_LOG(("Audio Read Error: Unable to avformat_alloc_context 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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fmtContext->pb = ioContext;
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if ((res = avformat_open_input(&fmtContext, 0, 0, 0)) < 0) {
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ioBuffer = 0;
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DEBUG_LOG(("Audio Read Error: Unable to avformat_open_input 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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_opened = true;
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if ((res = avformat_find_stream_info(fmtContext, 0)) < 0) {
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DEBUG_LOG(("Audio Read Error: Unable to avformat_find_stream_info 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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streamId = av_find_best_stream(fmtContext, AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0);
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if (streamId < 0) {
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LOG(("Audio Error: Unable to av_find_best_stream for file '%1', data size '%2', error %3, %4").arg(file.name()).arg(data.size()).arg(streamId).arg(av_make_error_string(err, sizeof(err), streamId)));
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return false;
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}
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freq = fmtContext->streams[streamId]->codec->sample_rate;
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if (fmtContext->streams[streamId]->duration == AV_NOPTS_VALUE) {
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len = (fmtContext->duration * freq) / AV_TIME_BASE;
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} else {
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len = (fmtContext->streams[streamId]->duration * freq * fmtContext->streams[streamId]->time_base.num) / fmtContext->streams[streamId]->time_base.den;
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}
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return true;
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}
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AbstractFFMpegLoader::~AbstractFFMpegLoader() {
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if (_opened) {
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avformat_close_input(&fmtContext);
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}
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if (ioContext) {
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av_free(ioContext->buffer);
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av_free(ioContext);
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} else if (ioBuffer) {
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av_free(ioBuffer);
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}
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if (fmtContext) avformat_free_context(fmtContext);
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}
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int AbstractFFMpegLoader::_read_data(void *opaque, uint8_t *buf, int buf_size) {
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AbstractFFMpegLoader *l = reinterpret_cast<AbstractFFMpegLoader*>(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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int64_t AbstractFFMpegLoader::_seek_data(void *opaque, int64_t offset, int whence) {
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AbstractFFMpegLoader *l = reinterpret_cast<AbstractFFMpegLoader*>(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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}
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if (newPos < 0 || newPos > l->data.size()) {
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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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int AbstractFFMpegLoader::_read_file(void *opaque, uint8_t *buf, int buf_size) {
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AbstractFFMpegLoader *l = reinterpret_cast<AbstractFFMpegLoader*>(opaque);
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return int(l->f.read((char*)(buf), buf_size));
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}
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int64_t AbstractFFMpegLoader::_seek_file(void *opaque, int64_t offset, int whence) {
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AbstractFFMpegLoader *l = reinterpret_cast<AbstractFFMpegLoader*>(opaque);
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switch (whence) {
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case SEEK_SET: return l->f.seek(offset) ? l->f.pos() : -1;
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case SEEK_CUR: return l->f.seek(l->f.pos() + offset) ? l->f.pos() : -1;
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case SEEK_END: return l->f.seek(l->f.size() + offset) ? l->f.pos() : -1;
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}
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return -1;
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}
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FFMpegLoader::FFMpegLoader(const FileLocation &file, const QByteArray &data) : AbstractFFMpegLoader(file, data) {
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frame = av_frame_alloc();
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}
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bool FFMpegLoader::open(qint64 &position) {
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if (!AbstractFFMpegLoader::open(position)) {
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return false;
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}
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int res = 0;
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char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
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// Get a pointer to the codec context for the audio stream
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av_opt_set_int(fmtContext->streams[streamId]->codec, "refcounted_frames", 1, 0);
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if ((res = avcodec_open2(fmtContext->streams[streamId]->codec, codec, 0)) < 0) {
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LOG(("Audio Error: Unable to avcodec_open2 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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codecContext = fmtContext->streams[streamId]->codec;
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uint64_t layout = codecContext->channel_layout;
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inputFormat = codecContext->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: fmt = AL_FORMAT_MONO8; sampleSize = 1; break;
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case AV_SAMPLE_FMT_S16:
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case AV_SAMPLE_FMT_S16P: fmt = 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: fmt = AL_FORMAT_STEREO8; sampleSize = 2; break;
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case AV_SAMPLE_FMT_S16: fmt = 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 (freq != 44100 && freq != 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 = freq;
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AVSampleFormat src_sample_fmt = inputFormat, dst_sample_fmt = AudioToFormat;
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dstRate = (freq != 44100 && freq != 48000) ? AudioVoiceMsgFrequency : freq;
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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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freq = dstRate;
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len = av_rescale_rnd(len, dstRate, srcRate, AV_ROUND_UP);
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position = av_rescale_rnd(position, dstRate, srcRate, AV_ROUND_DOWN);
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fmt = 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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if (position) {
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int64 ts = (position * fmtContext->streams[streamId]->time_base.den) / (freq * fmtContext->streams[streamId]->time_base.num);
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if (av_seek_frame(fmtContext, streamId, ts, AVSEEK_FLAG_ANY) < 0) {
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if (av_seek_frame(fmtContext, streamId, ts, 0) < 0) {
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}
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}
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}
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return true;
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}
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AudioPlayerLoader::ReadResult FFMpegLoader::readMore(QByteArray &result, int64 &samplesAdded) {
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int res;
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if ((res = av_read_frame(fmtContext, &avpkt)) < 0) {
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if (res != AVERROR_EOF) {
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char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
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LOG(("Audio Error: Unable to av_read_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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return ReadResult::EndOfFile;
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}
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if (avpkt.stream_index == streamId) {
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av_frame_unref(frame);
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int got_frame = 0;
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if ((res = avcodec_decode_audio4(codecContext, frame, &got_frame, &avpkt)) < 0) {
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char err[AV_ERROR_MAX_STRING_SIZE] = { 0 };
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LOG(("Audio Error: Unable to avcodec_decode_audio4() 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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av_packet_unref(&avpkt);
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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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if (got_frame) {
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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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av_packet_unref(&avpkt);
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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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av_packet_unref(&avpkt);
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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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}
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}
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av_packet_unref(&avpkt);
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return ReadResult::Ok;
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}
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FFMpegLoader::~FFMpegLoader() {
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if (codecContext) avcodec_close(codecContext);
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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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