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Unified Diff: media/filters/audio_file_reader.cc

Issue 2655783004: Decode entire in-memory file for WebAudio (Closed)
Patch Set: Clean up implementation. Created 3 years, 11 months ago
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Index: media/filters/audio_file_reader.cc
diff --git a/media/filters/audio_file_reader.cc b/media/filters/audio_file_reader.cc
index 582ca9cdc7fbbde62cbefc4bea56a84563da4ebc..315164e0c4a5bd1c706881a69f637a53c7c2fd8d 100644
--- a/media/filters/audio_file_reader.cc
+++ b/media/filters/audio_file_reader.cc
@@ -33,14 +33,7 @@ AudioFileReader::~AudioFileReader() {
}
bool AudioFileReader::Open() {
- if (!OpenDemuxer())
- return false;
- if (!OpenDecoder())
- return false;
-
- // If the duration is unknown, fail out; this API can not work with streams of
- // unknown duration currently.
- return glue_->format_context()->duration != AV_NOPTS_VALUE;
+ return OpenDemuxer() && OpenDecoder();
}
bool AudioFileReader::OpenDemuxer() {
@@ -127,6 +120,10 @@ bool AudioFileReader::OpenDecoder() {
return true;
}
+bool AudioFileReader::HasKnownDuration() const {
+ return glue_->format_context()->duration != AV_NOPTS_VALUE;
+}
+
void AudioFileReader::Close() {
codec_context_.reset();
glue_.reset();
@@ -250,6 +247,124 @@ int AudioFileReader::Read(AudioBus* audio_bus) {
return current_frame;
}
+std::unique_ptr<AudioBus> AudioFileReader::StreamingRead() {
DaleCurtis 2017/01/25 22:42:25 Needs refactor, so we don't have two copies of the
+ DCHECK(glue_.get() && codec_context_)
+ << "AudioFileReader::StreamingRead() : reader is not opened!";
+ size_t bytes_per_sample = av_get_bytes_per_sample(codec_context_->sample_fmt);
+
+ // Holds decoded audio.
+ std::unique_ptr<AVFrame, ScopedPtrAVFreeFrame> av_frame(av_frame_alloc());
+
+ // Holds a vector of AudioBus' containing each decoded packet.
+ std::vector<std::unique_ptr<AudioBus>> decodedAudioPackets;
+
+ AVPacket packet;
+ int current_frame = 0;
+ size_t total_frames = 0;
+ bool continue_decoding = true;
+
+ while (continue_decoding && ReadPacket(&packet)) {
+ // Make a shallow copy of packet so we can slide packet.data as frames are
+ // decoded from the packet; otherwise av_packet_unref() will corrupt memory.
+ AVPacket packet_temp = packet;
+ do {
+ // Reset frame to default values.
+ av_frame_unref(av_frame.get());
+
+ int frame_decoded = 0;
+ int result = avcodec_decode_audio4(codec_context_.get(), av_frame.get(),
+ &frame_decoded, &packet_temp);
+
+ if (result < 0) {
+ DLOG(WARNING)
+ << "AudioFileReader::Read() : error in avcodec_decode_audio4() -"
+ << result;
+ break;
+ }
+
+ // Update packet size and data pointer in case we need to call the decoder
+ // with the remaining bytes from this packet.
+ packet_temp.size -= result;
+ packet_temp.data += result;
+
+ if (!frame_decoded)
+ continue;
+
+ // Determine the number of sample-frames we just decoded. Check overflow.
+ int frames_read = av_frame->nb_samples;
+ total_frames += frames_read;
+
+ if (frames_read < 0) {
+ continue_decoding = false;
+ break;
+ }
+
+#ifdef CHROMIUM_NO_AVFRAME_CHANNELS
+ int channels =
+ av_get_channel_layout_nb_channels(av_frame->channel_layout);
+#else
+ int channels = av_frame->channels;
+#endif
+ if (av_frame->sample_rate != sample_rate_ || channels != channels_ ||
+ av_frame->format != av_sample_format_) {
+ DLOG(ERROR) << "Unsupported midstream configuration change!"
+ << " Sample Rate: " << av_frame->sample_rate << " vs "
+ << sample_rate_ << ", Channels: " << channels << " vs "
+ << channels_ << ", Sample Format: " << av_frame->format
+ << " vs " << av_sample_format_;
+
+ // This is an unrecoverable error, so bail out.
+ continue_decoding = false;
+ break;
+ }
+
+ // Deinterleave each channel and convert to 32bit floating-point with
+ // nominal range -1.0 -> +1.0. If the output is already in float planar
+ // format, just copy it into the AudioBus.
+ std::unique_ptr<AudioBus> audio_bus =
+ AudioBus::Create(channels, frames_read);
+
+ if (codec_context_->sample_fmt == AV_SAMPLE_FMT_FLT) {
DaleCurtis 2017/01/25 22:42:25 I believe there's an AudioBus::FromInterleaved met
+ float* decoded_audio_data = reinterpret_cast<float*>(av_frame->data[0]);
+ int channels = audio_bus->channels();
+ for (int ch = 0; ch < channels; ++ch) {
+ float* bus_data = audio_bus->channel(ch) + current_frame;
+ for (int i = 0, offset = ch; i < frames_read;
+ ++i, offset += channels) {
+ bus_data[i] = decoded_audio_data[offset];
+ }
+ }
+ } else if (codec_context_->sample_fmt == AV_SAMPLE_FMT_FLTP) {
+ for (int ch = 0; ch < audio_bus->channels(); ++ch) {
+ memcpy(audio_bus->channel(ch) + current_frame,
+ av_frame->extended_data[ch], sizeof(float) * frames_read);
+ }
+ } else {
+ audio_bus->FromInterleavedPartial(av_frame->data[0], current_frame,
+ frames_read, bytes_per_sample);
+ }
+
+ decodedAudioPackets.push_back(std::move(audio_bus));
+ // current_frame += frames_read;
+ } while (packet_temp.size > 0);
+ av_packet_unref(&packet);
+ }
+
+ // Concatenate all the decoded chunks into one big bus.
+ std::unique_ptr<AudioBus> outputBus =
+ AudioBus::Create(channels_, total_frames);
+
+ int dest_start_frame = 0;
+ for (size_t k = 0; k < decodedAudioPackets.size(); ++k) {
+ int frame_count = decodedAudioPackets[k]->frames();
+ decodedAudioPackets[k]->CopyPartialFramesTo(
+ 0, frame_count, dest_start_frame, outputBus.get());
+ dest_start_frame += frame_count;
+ }
+
+ return outputBus;
+}
+
base::TimeDelta AudioFileReader::GetDuration() const {
const AVRational av_time_base = {1, AV_TIME_BASE};

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