Support Opus Ambisonics playback

media/filters/ffmpeg_audio_decoder.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/filters/ffmpeg_audio_decoder.h"
 
 #include <stdint.h>
 
 #include "base/callback_helpers.h"
 #include "base/single_thread_task_runner.h"
@@ skipping 78 matching lines @@
   // we need to take this into consideration.
   int buffer_size_in_bytes = av_samples_get_buffer_size(
       &frame->linesize[0], channels, frame->nb_samples, format,
       0 /* align, use ffmpeg default */);
   // Check for errors from av_samples_get_buffer_size().
   if (buffer_size_in_bytes < 0)
     return buffer_size_in_bytes;
   int frames_required = buffer_size_in_bytes / bytes_per_channel / channels;
   DCHECK_GE(frames_required, frame->nb_samples);
 
+  bool is_opus_discrete = false;

[DaleCurtis, 2017/03/29 22:46:55]

Isn't this information already in the AudioDecoderConfig?

[flim-chromium, 2017/03/30 19:35:10]

True, but this is a static function and the non-static config_ can't be accessed
here?

[DaleCurtis, 2017/03/30 20:22:21]

Ah right, this isn't like FFmpegVideoDecoder where we have a class level method.
Is it enough to just check OPUS && channels > 8?

[flim-chromium, 2017/03/31 00:12:14]

I believe so - it includes the additional check that the Opus packet was encoded
specifically with channel mapping 2. This mirrors the check in
media/ffmpeg/ffmpeg_common.cc.

I'd renamed to |is_opus_discrete| in both places because we'd want to extend
this in the future to include  mapping_family == 3 (and maybe 255 if that makes
sense)

[DaleCurtis, 2017/03/31 19:06:53]

I'm moving this to be a class method in
https://codereview.chromium.org/2788483003 so we can use the config_ here if
that lands before yours.

[flim-chromium, 2017/04/05 18:19:34]

thanks, this has been updated to use config_

+  if (s->codec_id == AV_CODEC_ID_OPUS && s->extradata_size >= 19) {
+    int mapping_family = s->extradata[18];
+    is_opus_discrete = mapping_family == 2;
+  }
   ChannelLayout channel_layout =
-      ChannelLayoutToChromeChannelLayout(s->channel_layout, s->channels);
+      is_opus_discrete && channels > 8
+          ? CHANNEL_LAYOUT_DISCRETE
+          : ChannelLayoutToChromeChannelLayout(s->channel_layout, s->channels);
 
   if (channel_layout == CHANNEL_LAYOUT_UNSUPPORTED) {
     DLOG(ERROR) << "Unsupported channel layout.";
     return AVERROR(EINVAL);
   }
 
   scoped_refptr<AudioBuffer> buffer = AudioBuffer::CreateBuffer(
       sample_format, channel_layout, channels, s->sample_rate, frames_required);
 
   // Initialize the data[] and extended_data[] fields to point into the memory
@@ skipping 186 matching lines @@
     scoped_refptr<AudioBuffer> output;
 
     bool config_changed = false;
     if (frame_decoded) {
       const int channels = DetermineChannels(av_frame_.get());
       ChannelLayout channel_layout = ChannelLayoutToChromeChannelLayout(
           codec_context_->channel_layout, codec_context_->channels);
 
       bool is_sample_rate_change =
           av_frame_->sample_rate != config_.samples_per_second();
-      bool is_config_stale =
-          is_sample_rate_change ||
-          channels != ChannelLayoutToChannelCount(config_.channel_layout()) ||
-          av_frame_->format != av_sample_format_;
+      bool is_config_stale = is_sample_rate_change ||
+                             channels != config_.channels() ||
+                             av_frame_->format != av_sample_format_;
 
       // Only consider channel layout changes for AAC.
       // TODO(tguilbert, dalecurtis): Due to http://crbug.com/600538 we need to
       // allow channel layout changes for the moment. See if ffmpeg is fixable.
       if (config_.codec() == kCodecAAC)
         is_config_stale |= channel_layout != config_.channel_layout();
 
       if (is_config_stale) {
         // Only allow midstream configuration changes for AAC. Sample format is
         // not expected to change between AAC profiles.
         if (config_.codec() == kCodecAAC &&
             av_frame_->format == av_sample_format_) {
           MEDIA_LOG(DEBUG, media_log_)
               << " Detected AAC midstream configuration change"
               << " PTS:" << buffer->timestamp().InMicroseconds()
               << " Sample Rate: " << av_frame_->sample_rate << " vs "
               << config_.samples_per_second()
               << ", ChannelLayout: " << channel_layout << " vs "
               << config_.channel_layout() << ", Channels: " << channels
-              << " vs "
-              << ChannelLayoutToChannelCount(config_.channel_layout());
+              << " vs " << config_.channels();
           config_.Initialize(config_.codec(), config_.sample_format(),
                              channel_layout, av_frame_->sample_rate,
                              config_.extra_data(), config_.encryption_scheme(),
                              config_.seek_preroll(), config_.codec_delay());
           config_changed = true;
           if (is_sample_rate_change)
             ResetTimestampState();
         } else {
           MEDIA_LOG(ERROR, media_log_)
               << "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.
           av_frame_unref(av_frame_.get());
           return false;
         }
       }
 
       // Get the AudioBuffer that the data was decoded into. Adjust the number
       // of frames, in case fewer than requested were actually decoded.
       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)
         output->TrimEnd(unread_frames);
       av_frame_unref(av_frame_.get());
     }
 
     // WARNING: |av_frame_| no longer has valid data at this point.
     const int decoded_frames = frame_decoded ? output->frame_count() : 0;
     if (IsEndOfStream(result, decoded_frames, buffer)) {
@@ skipping 44 matching lines @@
                 << codec_context_->codec_id;
     ReleaseFFmpegResources();
     state_ = kUninitialized;
     return false;
   }
 
   // Success!
   av_frame_.reset(av_frame_alloc());
   av_sample_format_ = codec_context_->sample_fmt;
 
-  if (codec_context_->channels !=
-      ChannelLayoutToChannelCount(config_.channel_layout())) {
-    DLOG(ERROR) << "Audio configuration specified "
-                << ChannelLayoutToChannelCount(config_.channel_layout())
+  if (codec_context_->channels != config_.channels()) {
+    DLOG(ERROR) << "Audio configuration specified " << config_.channels()
                 << " channels, but FFmpeg thinks the file contains "
                 << codec_context_->channels << " channels";
     ReleaseFFmpegResources();
     state_ = kUninitialized;
     return false;
   }
 
   ResetTimestampState();
   return true;
 }
 
 void FFmpegAudioDecoder::ResetTimestampState() {
   // Opus codec delay is handled by ffmpeg.
   const int codec_delay =
       config_.codec() == kCodecOpus ? 0 : config_.codec_delay();
   discard_helper_.reset(
       new AudioDiscardHelper(config_.samples_per_second(), codec_delay,
                              config_.codec() == kCodecVorbis));
   discard_helper_->Reset(codec_delay);
 }
 
 }  // namespace media