Support Opus Ambisonics playback

media/filters/ffmpeg_audio_decoder.cc

Index: media/filters/ffmpeg_audio_decoder.cc
diff --git a/media/filters/ffmpeg_audio_decoder.cc b/media/filters/ffmpeg_audio_decoder.cc
index e3560ff1b8bcf9d359f2a7c0ab06147b8e6b4e3f..7e7228991eb49820206487a2a65ac453637e6fbb 100644
--- a/media/filters/ffmpeg_audio_decoder.cc
+++ b/media/filters/ffmpeg_audio_decoder.cc
@@ -84,6 +84,18 @@ static int GetAudioBuffer(struct AVCodecContext* s, AVFrame* frame, int flags) {
     return AVERROR(EINVAL);
   }
 
+  // Use CHANNEL_LAYOUT_DISCRETE for Opus Ambisonics signals
+  int is_opus_ambisonics = 0;

[DaleCurtis, 2017/03/20 17:38:58]

bool.

[flim-chromium, 2017/03/22 06:20:51]

Done.

+  if (s->codec_id == AV_CODEC_ID_OPUS && s->extradata_size >= 19) {
+    int mapping_family = s->extradata[18];
+    is_opus_ambisonics = mapping_family == 2;
+  }
+  ChannelLayout channel_layout = is_opus_ambisonics

[DaleCurtis, 2017/03/20 17:38:58]

&& channels > 8.

[flim-chromium, 2017/03/22 06:20:51]

Ambisonics signals could also be < 8 channels. Or do you mean even if it's not
opus ambisonics signal?

[DaleCurtis, 2017/03/22 18:56:30]

I mean I don't think DISCRETE will be passed or work correctly all through out
the stack if < 8 channels; specifically OSX may have troubles. You'll need to do
some further research at all the places we convert from channel count to layout
and use DISCRETE today.

[flim-chromium, 2017/03/29 03:15:25]

Thanks for the explanation. I've added '&& channels > 8'.

+                                 ? CHANNEL_LAYOUT_DISCRETE
+                                 : ChannelLayoutToChromeChannelLayout(
+                                       s->channel_layout,
+                                       s->channels);
+
   // Determine how big the buffer should be and allocate it. FFmpeg may adjust
   // how big each channel data is in order to meet the alignment policy, so
   // we need to take this into consideration.
@@ -97,7 +109,7 @@ static int GetAudioBuffer(struct AVCodecContext* s, AVFrame* frame, int flags) {
   DCHECK_GE(frames_required, frame->nb_samples);
   scoped_refptr<AudioBuffer> buffer = AudioBuffer::CreateBuffer(
       sample_format,
-      ChannelLayoutToChromeChannelLayout(s->channel_layout, s->channels),
+      channel_layout,
       channels,
       s->sample_rate,
       frames_required);
@@ -301,7 +313,7 @@ bool FFmpegAudioDecoder::FFmpegDecode(
           av_frame_->sample_rate != config_.samples_per_second();
       bool is_config_stale =
           is_sample_rate_change ||
-          channels != ChannelLayoutToChannelCount(config_.channel_layout()) ||
+          channels != config_.channels() ||
           av_frame_->format != av_sample_format_;
 
       // Only consider channel layout changes for AAC.
@@ -323,7 +335,7 @@ bool FFmpegAudioDecoder::FFmpegDecode(
               << ", ChannelLayout: " << channel_layout << " vs "
               << config_.channel_layout() << ", Channels: " << channels
               << " vs "
-              << ChannelLayoutToChannelCount(config_.channel_layout());
+              << config_.channels();
           config_.Initialize(config_.codec(), config_.sample_format(),
                              channel_layout, av_frame_->sample_rate,
                              config_.extra_data(), config_.encryption_scheme(),
@@ -336,7 +348,7 @@ bool FFmpegAudioDecoder::FFmpegDecode(
               << "Unsupported midstream configuration change!"
               << " Sample Rate: " << av_frame_->sample_rate << " vs "
               << config_.samples_per_second() << ", Channels: " << channels
-              << " vs " << ChannelLayoutToChannelCount(config_.channel_layout())
+              << " vs " << config_.channels()
               << ", Sample Format: " << av_frame_->format << " vs "
               << av_sample_format_;
           // This is an unrecoverable error, so bail out.
@@ -350,8 +362,7 @@ bool FFmpegAudioDecoder::FFmpegDecode(
       output = reinterpret_cast<AudioBuffer*>(
           av_buffer_get_opaque(av_frame_->buf[0]));
 
-      DCHECK_EQ(ChannelLayoutToChannelCount(config_.channel_layout()),
-                output->channel_count());
+      DCHECK_EQ(config_.channels(), output->channel_count());
       const int unread_frames = output->frame_count() - av_frame_->nb_samples;
       DCHECK_GE(unread_frames, 0);
       if (unread_frames > 0)
@@ -416,10 +427,10 @@ bool FFmpegAudioDecoder::ConfigureDecoder() {
   av_frame_.reset(av_frame_alloc());
   av_sample_format_ = codec_context_->sample_fmt;
 
-  if (codec_context_->channels !=
-      ChannelLayoutToChannelCount(config_.channel_layout())) {
+
+  if (codec_context_->channels != config_.channels()) {
     DLOG(ERROR) << "Audio configuration specified "
-                << ChannelLayoutToChannelCount(config_.channel_layout())
+                << config_.channels()
                 << " channels, but FFmpeg thinks the file contains "
                 << codec_context_->channels << " channels";
     ReleaseFFmpegResources();