began work on exr proxy task
This commit is contained in:
itsmattkc
+156
-154
@@ -623,24 +623,170 @@ void FFmpegDecoder::Error(const QString &s)
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ClearResources();
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}
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void FFmpegDecoder::Index(const QAtomicInt* cancelled)
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void FFmpegDecoder::ProxyVideo(const QAtomicInt *cancelled, int divider)
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{
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// Iterate through each video frame transcode each frame to compressed EXR
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QMutexLocker locker(stream()->index_process_lock());
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if (stream()->type() == Stream::kAudio) {
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QByteArray fn_bytes = stream()->footage()->filename().toUtf8();
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if (QFileInfo::exists(GetIndexFilename())) {
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WaveInput input(GetIndexFilename());
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if (input.open()) {
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std::static_pointer_cast<AudioStream>(stream())->set_index_done(true);
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std::static_pointer_cast<AudioStream>(stream())->set_index_length(input.params().bytes_to_time(input.data_length()));
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FFmpegDecoderInstance index_instance(fn_bytes.constData(), stream()->index());
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input.close();
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}
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void FFmpegDecoder::ProxyAudio(const QAtomicInt *cancelled)
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{
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// Iterate through each audio frame and extract the PCM data
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QMutexLocker locker(stream()->index_process_lock());
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if (QFileInfo::exists(GetIndexFilename())) {
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WaveInput input(GetIndexFilename());
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if (input.open()) {
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std::static_pointer_cast<AudioStream>(stream())->set_index_done(true);
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std::static_pointer_cast<AudioStream>(stream())->set_index_length(input.params().bytes_to_time(input.data_length()));
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input.close();
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}
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} else {
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QByteArray fn_bytes = stream()->footage()->filename().toUtf8();
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FFmpegDecoderInstance index_instance(fn_bytes.constData(), stream()->index());
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uint64_t channel_layout = index_instance.stream()->codecpar->channel_layout;
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if (!channel_layout) {
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if (!index_instance.stream()->codecpar->channels) {
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// No channel data - we can't do anything with this
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return;
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}
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} else {
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UnconditionalAudioIndex(cancelled);
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channel_layout = static_cast<uint64_t>(av_get_default_channel_layout(index_instance.stream()->codecpar->channels));
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}
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AudioStreamPtr audio_stream = std::static_pointer_cast<AudioStream>(stream());
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// This should be unnecessary, but just in case...
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audio_stream->clear_index();
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SwrContext* resampler = nullptr;
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AVSampleFormat src_sample_fmt = static_cast<AVSampleFormat>(index_instance.stream()->codecpar->format);
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AVSampleFormat dst_sample_fmt;
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// We don't use planar types internally, so if this is a planar format convert it now
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if (av_sample_fmt_is_planar(src_sample_fmt)) {
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dst_sample_fmt = av_get_packed_sample_fmt(src_sample_fmt);
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resampler = swr_alloc_set_opts(nullptr,
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static_cast<int64_t>(index_instance.stream()->codecpar->channel_layout),
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dst_sample_fmt,
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index_instance.stream()->codecpar->sample_rate,
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static_cast<int64_t>(index_instance.stream()->codecpar->channel_layout),
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src_sample_fmt,
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index_instance.stream()->codecpar->sample_rate,
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0,
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nullptr);
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swr_init(resampler);
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} else {
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dst_sample_fmt = src_sample_fmt;
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}
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AudioRenderingParams wave_params(index_instance.stream()->codecpar->sample_rate,
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channel_layout,
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FFmpegCommon::GetNativeSampleFormat(dst_sample_fmt));
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WaveOutput wave_out(GetIndexFilename(), wave_params);
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AVPacket* pkt = av_packet_alloc();
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AVFrame* frame = av_frame_alloc();
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int ret;
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if (wave_out.open()) {
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bool success = false;
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while (true) {
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// Check if we have a `cancelled` ptr and its value
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if (cancelled && *cancelled) {
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break;
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}
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ret = index_instance.GetFrame(pkt, frame);
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if (ret < 0) {
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if (ret == AVERROR_EOF) {
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success = true;
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} else {
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char err_str[50];
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av_strerror(ret, err_str, 50);
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qWarning() << "Failed to index:" << ret << err_str;
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}
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break;
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} else {
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char* data;
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int nb_samples;
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if (resampler) {
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nb_samples = swr_get_out_samples(resampler, frame->nb_samples);
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data = new char[wave_params.samples_to_bytes(nb_samples)];
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// We must need to resample this (mainly just convert from planar to packed if necessary)
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nb_samples = swr_convert(resampler,
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reinterpret_cast<uint8_t**>(&data),
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nb_samples,
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const_cast<const uint8_t**>(frame->data),
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frame->nb_samples);
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if (nb_samples < 0) {
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char err_str[50];
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av_strerror(nb_samples, err_str, 50);
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qWarning() << "libswresample failed with error:" << nb_samples << err_str;
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break;
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}
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} else {
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// No resampling required, we can write directly from the frame buffer
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data = reinterpret_cast<char*>(frame->data[0]);
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nb_samples = frame->nb_samples;
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}
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// Write packed WAV data to the disk cache
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wave_out.write(data, wave_params.samples_to_bytes(nb_samples));
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audio_stream->set_index_length(wave_params.bytes_to_time(wave_out.data_length()));
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// If we allocated an output for the resampler, delete it here
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if (data != reinterpret_cast<char*>(frame->data[0])) {
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delete [] data;
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}
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SignalIndexProgress(frame->pts);
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}
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}
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wave_out.close();
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if (success) {
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audio_stream->set_index_done(true);
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} else {
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// Audio index didn't complete, delete it
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QFile(GetIndexFilename()).remove();
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audio_stream->clear_index();
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}
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} else {
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qWarning() << "Failed to open WAVE output for indexing";
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}
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if (resampler != nullptr) {
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swr_free(&resampler);
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}
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av_frame_free(&frame);
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av_packet_free(&pkt);
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}
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}
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@@ -655,150 +801,6 @@ int FFmpegDecoder::GetScaledDimension(int dim, int divider)
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return dim / divider;
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}
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void FFmpegDecoder::UnconditionalAudioIndex(const QAtomicInt* cancelled)
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{
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// Iterate through each audio frame and extract the PCM data
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QByteArray fn_bytes = stream()->footage()->filename().toUtf8();
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FFmpegDecoderInstance index_instance(fn_bytes.constData(), stream()->index());
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uint64_t channel_layout = index_instance.stream()->codecpar->channel_layout;
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if (!channel_layout) {
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if (!index_instance.stream()->codecpar->channels) {
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// No channel data - we can't do anything with this
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return;
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}
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channel_layout = static_cast<uint64_t>(av_get_default_channel_layout(index_instance.stream()->codecpar->channels));
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}
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AudioStreamPtr audio_stream = std::static_pointer_cast<AudioStream>(stream());
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// This should be unnecessary, but just in case...
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audio_stream->clear_index();
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SwrContext* resampler = nullptr;
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AVSampleFormat src_sample_fmt = static_cast<AVSampleFormat>(index_instance.stream()->codecpar->format);
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AVSampleFormat dst_sample_fmt;
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// We don't use planar types internally, so if this is a planar format convert it now
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if (av_sample_fmt_is_planar(src_sample_fmt)) {
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dst_sample_fmt = av_get_packed_sample_fmt(src_sample_fmt);
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resampler = swr_alloc_set_opts(nullptr,
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static_cast<int64_t>(index_instance.stream()->codecpar->channel_layout),
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dst_sample_fmt,
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index_instance.stream()->codecpar->sample_rate,
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static_cast<int64_t>(index_instance.stream()->codecpar->channel_layout),
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src_sample_fmt,
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index_instance.stream()->codecpar->sample_rate,
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0,
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nullptr);
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swr_init(resampler);
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} else {
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dst_sample_fmt = src_sample_fmt;
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}
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AudioRenderingParams wave_params(index_instance.stream()->codecpar->sample_rate,
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channel_layout,
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FFmpegCommon::GetNativeSampleFormat(dst_sample_fmt));
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WaveOutput wave_out(GetIndexFilename(), wave_params);
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AVPacket* pkt = av_packet_alloc();
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AVFrame* frame = av_frame_alloc();
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int ret;
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if (wave_out.open()) {
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bool success = false;
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while (true) {
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// Check if we have a `cancelled` ptr and its value
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if (cancelled && *cancelled) {
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break;
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}
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ret = index_instance.GetFrame(pkt, frame);
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if (ret < 0) {
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if (ret == AVERROR_EOF) {
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success = true;
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} else {
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char err_str[50];
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av_strerror(ret, err_str, 50);
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qWarning() << "Failed to index:" << ret << err_str;
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}
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break;
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} else {
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char* data;
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int nb_samples;
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if (resampler) {
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nb_samples = swr_get_out_samples(resampler, frame->nb_samples);
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data = new char[wave_params.samples_to_bytes(nb_samples)];
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// We must need to resample this (mainly just convert from planar to packed if necessary)
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nb_samples = swr_convert(resampler,
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reinterpret_cast<uint8_t**>(&data),
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nb_samples,
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const_cast<const uint8_t**>(frame->data),
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frame->nb_samples);
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if (nb_samples < 0) {
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char err_str[50];
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av_strerror(nb_samples, err_str, 50);
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qWarning() << "libswresample failed with error:" << nb_samples << err_str;
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break;
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}
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} else {
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// No resampling required, we can write directly from the frame buffer
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data = reinterpret_cast<char*>(frame->data[0]);
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nb_samples = frame->nb_samples;
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}
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// Write packed WAV data to the disk cache
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wave_out.write(data, wave_params.samples_to_bytes(nb_samples));
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audio_stream->set_index_length(wave_params.bytes_to_time(wave_out.data_length()));
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// If we allocated an output for the resampler, delete it here
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if (data != reinterpret_cast<char*>(frame->data[0])) {
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delete [] data;
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}
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SignalIndexProgress(frame->pts);
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}
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}
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wave_out.close();
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if (success) {
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audio_stream->set_index_done(true);
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} else {
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// Audio index didn't complete, delete it
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QFile(GetIndexFilename()).remove();
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audio_stream->clear_index();
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}
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} else {
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qWarning() << "Failed to open WAVE output for indexing";
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}
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if (resampler != nullptr) {
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swr_free(&resampler);
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}
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av_frame_free(&frame);
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av_packet_free(&pkt);
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}
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int FFmpegDecoderInstance::GetFrame(AVPacket *pkt, AVFrame *frame)
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{
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bool eof = false;
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