mirror of https://github.com/procxx/kepka.git
ffmpeg support for audio added
This commit is contained in:
John Preston
parent
e3ab8821b9
commit
981162e6b9
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@ -29,6 +29,15 @@ Copyright (c) 2014 John Preston, https://desktop.telegram.org
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#include <neaacdec.h>
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#include <mp4ff.h>
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extern "C" {
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#include <libavcodec/avcodec.h>
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#include <libavformat/avformat.h>
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#include <libavutil/opt.h>
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#include <libswresample/swresample.h>
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}
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namespace {
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ALCdevice *audioDevice = 0;
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ALCcontext *audioContext = 0;
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@ -165,6 +174,9 @@ void audioInit() {
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LOG(("Could not init MPG123, result: %1").arg(mpg123res));
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}
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av_register_all();
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avcodec_register_all();
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LOG(("Audio init time: %1").arg(getms() - ms));
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}
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@ -1084,6 +1096,288 @@ private:
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}
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};
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static const uint32 AVBlockSize = 4096; // 4Kb
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static const AVSampleFormat _toFormat = AV_SAMPLE_FMT_S16;
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static const int64_t _toChannelLayout = AV_CH_LAYOUT_STEREO;
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static const int32 _toChannels = 2;
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class FFMpegLoader : public VoiceMessagesLoader {
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public:
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FFMpegLoader(const QString &fname, const QByteArray &data) : VoiceMessagesLoader(fname, data),
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freq(AudioVoiceMsgFrequency), fmt(AL_FORMAT_STEREO16),
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sampleSize(2 * sizeof(short)), srcRate(AudioVoiceMsgFrequency), dstRate(AudioVoiceMsgFrequency),
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maxResampleSamples(1024), resampleData(0), len(0),
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ioBuffer(0), ioContext(0), fmtContext(0), codec(0), codecContext(0), streamId(0), frame(0), swrContext(0),
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_opened(false) {
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frame = av_frame_alloc();
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}
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bool open() {
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if (!VoiceMessagesLoader::openFile()) {
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return false;
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}
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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, static_cast<void*>(this), &FFMpegLoader::_read_file, 0, &FFMpegLoader::_seek_file);
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} else {
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ioContext = avio_alloc_context(ioBuffer, AVBlockSize, 0, static_cast<void*>(this), &FFMpegLoader::_read_data, 0, &FFMpegLoader::_seek_data);
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}
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fmtContext = avformat_alloc_context();
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fmtContext->pb = ioContext;
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if (!fmtContext) {
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LOG(("Audio Error: Unable to avformat_alloc_context for file '%1', data size '%2'").arg(fname).arg(data.size()));
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return false;
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}
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int res = 0;
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if ((res = avformat_open_input(&fmtContext, 0, 0, 0)) < 0) {
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LOG(("Audio Error: Unable to avformat_open_input for file '%1', data size '%2', error %3, %4").arg(fname).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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LOG(("Audio Error: Unable to avformat_find_stream_info for file '%1', data size '%2', error %3, %4").arg(fname).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(fname).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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// Get a pointer to the codec context for the audio stream
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codecContext = fmtContext->streams[streamId]->codec;
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av_opt_set_int(codecContext, "refcounted_frames", 1, 0);
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if ((res = avcodec_open2(codecContext, codec, 0)) < 0) {
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LOG(("Audio Error: Unable to avcodec_open2 for file '%1', data size '%2', error %3, %4").arg(fname).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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freq = fmtContext->streams[streamId]->codec->sample_rate;
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len = (fmtContext->streams[streamId]->duration * freq) / fmtContext->streams[streamId]->time_base.den;
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uint64_t layout = fmtContext->streams[streamId]->codec->channel_layout;
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inputFormat = fmtContext->streams[streamId]->codec->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 = 2; 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 = sizeof(short); break;
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case AV_SAMPLE_FMT_S16: fmt = AL_FORMAT_STEREO16; sampleSize = 2 * sizeof(short); 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(fname).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 = _toChannelLayout;
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srcRate = freq;
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AVSampleFormat src_sample_fmt = inputFormat, dst_sample_fmt = _toFormat;
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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(fname).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 = _toChannels * 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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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(&resampleData, 0, _toChannels, maxResampleSamples, _toFormat, 0)) < 0) {
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LOG(("Audio Error: Unable to av_samples_alloc for file '%1', data size '%2', error %3, %4").arg(fname).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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int64 duration() {
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return len;
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}
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int32 frequency() {
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return freq;
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}
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int32 format() {
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return fmt;
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}
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void started() {
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}
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bool 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(fname).arg(data.size()).arg(res).arg(av_make_error_string(err, sizeof(err), res)));
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}
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return false;
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}
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if (avpkt.stream_index == streamId) {
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avcodec_get_frame_defaults(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(fname).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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if (got_frame) {
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if (resampleData) { // 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(resampleData[0]);
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if ((res = av_samples_alloc(resampleData, 0, _toChannels, maxResampleSamples, _toFormat, 1)) < 0) {
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resampleData[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(fname).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 ((res = swr_convert(swrContext, resampleData, 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(fname).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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int32 resultLen = av_samples_get_buffer_size(0, _toChannels, res, _toFormat, 1);
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result.append((const char*)resampleData[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_free_packet(&avpkt);
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return true;
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}
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~FFMpegLoader() {
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if (ioContext) av_free(ioContext);
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if (codecContext) avcodec_close(codecContext);
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if (swrContext) swr_free(&swrContext);
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if (resampleData) {
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if (resampleData[0]) {
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av_freep(&resampleData[0]);
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}
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av_freep(&resampleData);
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}
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if (_opened) {
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avformat_close_input(&fmtContext);
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} else {
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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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av_frame_free(&frame);
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}
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private:
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int32 freq, fmt, channels;
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int32 sampleSize, srcRate, dstRate, maxResampleSamples;
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uint8_t **resampleData;
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int64 len;
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uchar *ioBuffer;
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AVIOContext *ioContext;
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AVFormatContext *fmtContext;
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AVCodec *codec;
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AVCodecContext *codecContext;
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AVPacket avpkt;
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int32 streamId;
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AVSampleFormat inputFormat;
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AVFrame *frame;
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SwrContext *swrContext;
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bool _opened;
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static int _read_data(void *opaque, uint8_t *buf, int buf_size) {
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FFMpegLoader *l = reinterpret_cast<FFMpegLoader*>(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 int64_t _seek_data(void *opaque, int64_t offset, int whence) {
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FFMpegLoader *l = reinterpret_cast<FFMpegLoader*>(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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static int _read_file(void *opaque, uint8_t *buf, int buf_size) {
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FFMpegLoader *l = reinterpret_cast<FFMpegLoader*>(opaque);
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return ssize_t(l->f.read((char*)(buf), buf_size));
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}
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static int64_t _seek_file(void *opaque, int64_t offset, int whence) {
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FFMpegLoader *l = reinterpret_cast<FFMpegLoader*>(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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};
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VoiceMessagesLoaders::VoiceMessagesLoaders(QThread *thread) {
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moveToThread(thread);
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}
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@ -1153,7 +1447,9 @@ void VoiceMessagesLoaders::onLoad(AudioData *audio) {
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uint32 mpegHead = (uint32(uchar(header.at(0))) << 24) | (uint32(uchar(header.at(1))) << 16) | (uint32(uchar(header.at(2))) << 8) | uint32(uchar(header.at(3)));
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bool validMpegHead = ((mpegHead & HDR_SYNC) == HDR_SYNC) && !!(HDR_LAYER_VAL(mpegHead)) && (HDR_BITRATE_VAL(mpegHead) != 0x0F) && (HDR_SAMPLERATE_VAL(mpegHead) != 0x03);
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if (header.at(0) == 'O' && header.at(1) == 'g' && header.at(2) == 'g' && header.at(3) == 'S') {
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if (true) {
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j = _loaders.insert(audio, new FFMpegLoader(m.fname, m.data));
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} else if (header.at(0) == 'O' && header.at(1) == 'g' && header.at(2) == 'g' && header.at(3) == 'S') {
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j = _loaders.insert(audio, new OggOpusLoader(m.fname, m.data));
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} else if (header.at(4) == 'f' && header.at(5) == 't' && header.at(6) == 'y' && header.at(7) == 'p') {
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j = _loaders.insert(audio, new FAADMp4Loader(m.fname, m.data));
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@ -1816,8 +1816,8 @@ bool EmojiPan::animStep(float64 ms) {
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res1 = false;
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hideFinish();
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} else {
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if (_toCache.isNull()) showAll();
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_cache = QPixmap();
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if (_toCache.isNull()) showAll();
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}
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} else {
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a_opacity.update(dt, anim::linear);
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