Support Opus Ambisonics playback

media/ffmpeg/ffmpeg_common.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/ffmpeg/ffmpeg_common.h"
 
 #include "base/logging.h"
 #include "base/sha1.h"
 #include "base/strings/string_number_conversions.h"
 #include "base/strings/string_split.h"
@@ skipping 306 matching lines @@
     const AVCodecContext* codec_context,
     const EncryptionScheme& encryption_scheme,
     AudioDecoderConfig* config) {
   DCHECK_EQ(codec_context->codec_type, AVMEDIA_TYPE_AUDIO);
 
   AudioCodec codec = CodecIDToAudioCodec(codec_context->codec_id);
 
   SampleFormat sample_format = AVSampleFormatToSampleFormat(
       codec_context->sample_fmt, codec_context->codec_id);
 
-  ChannelLayout channel_layout = ChannelLayoutToChromeChannelLayout(
-      codec_context->channel_layout, codec_context->channels);
+  bool is_opus_discrete = false;
+  if (codec == kCodecOpus && codec_context->extradata_size >= 19) {
+    int mapping_family = codec_context->extradata[18];
+    is_opus_discrete = mapping_family == 2;
+  }
+  ChannelLayout channel_layout =
+      is_opus_discrete && codec_context->channels > 8

[DaleCurtis, 2017/05/24 00:49:58]

Did you ever figure out if this > 8 check could be dropped? Or does it not
matter for your use case?

[flim-chromium, 2017/05/25 00:04:25]

I'm not familiar enough with the potential problems (e.g. in OSX that you
pointed out before) to drop this. 

Anyway, I think it's actually useful to keep this such that for <= 8ch, the
default up/downmixing is applied if a webaudio renderer isn't attached. It
wouldn't sound spatially correct but that's to be expected, and at least we'd
get to preview the audio content. Potentially a future CL could look at some
saner up/downmixing matrices, and also for >8 ch ambisonics. wdyt?

[DaleCurtis, 2017/05/25 00:29:16]

Leaving sgtm for now, though I think WebAudio doesn't want channel mixing, so if
we could always skip it when WebAudio is connected that is fine.

+          ? CHANNEL_LAYOUT_DISCRETE
+          : ChannelLayoutToChromeChannelLayout(codec_context->channel_layout,
+                                               codec_context->channels);
 
   int sample_rate = codec_context->sample_rate;
   switch (codec) {
     // For AC3/EAC3 we enable only demuxing, but not decoding, so FFmpeg does
     // not fill |sample_fmt|.
     case kCodecAC3:
     case kCodecEAC3:
 #if BUILDFLAG(ENABLE_AC3_EAC3_AUDIO_DEMUXING)
       // The spec for AC3/EAC3 audio is ETSI TS 102 366. According to sections
       // F.3.1 and F.5.1 in that spec the sample_format for AC3/EAC3 must be 16.
@@ skipping 25 matching lines @@
 
   std::vector<uint8_t> extra_data;
   if (codec_context->extradata_size > 0) {
     extra_data.assign(codec_context->extradata,
                       codec_context->extradata + codec_context->extradata_size);
   }
 
   config->Initialize(codec, sample_format, channel_layout, sample_rate,
                      extra_data, encryption_scheme, seek_preroll,
                      codec_context->delay);
+  config->SetChannelsForDiscrete(codec_context->channels);

[DaleCurtis, 2017/05/24 00:49:58]

Probably you don't want to always do this. channels from ffmpeg is not reliable.
Only with layout == discrete should this be called I think.

[flim-chromium, 2017/05/25 00:04:26]

Right, I ran into this problem with the ffmpeg tests, hence the change in
SetChannelsForDiscrete. Your way makes sense, I've changed it back now.

 
 #if BUILDFLAG(ENABLE_AC3_EAC3_AUDIO_DEMUXING)
   // These are bitstream formats unknown to ffmpeg, so they don't have
   // a known sample format size.
   if (codec == kCodecAC3 || codec == kCodecEAC3)
     return true;
 #endif
 
   // Verify that AudioConfig.bits_per_channel was calculated correctly for
   // codecs that have |sample_fmt| set by FFmpeg.
@@ skipping 28 matching lines @@
 void AudioDecoderConfigToAVCodecContext(const AudioDecoderConfig& config,
                                         AVCodecContext* codec_context) {
   codec_context->codec_type = AVMEDIA_TYPE_AUDIO;
   codec_context->codec_id = AudioCodecToCodecID(config.codec(),
                                                 config.sample_format());
   codec_context->sample_fmt = SampleFormatToAVSampleFormat(
       config.sample_format());
 
   // TODO(scherkus): should we set |channel_layout|? I'm not sure if FFmpeg uses
   // said information to decode.
-  codec_context->channels =
-      ChannelLayoutToChannelCount(config.channel_layout());
+  codec_context->channels = config.channels();
   codec_context->sample_rate = config.samples_per_second();
 
   if (config.extra_data().empty()) {
     codec_context->extradata = nullptr;
     codec_context->extradata_size = 0;
   } else {
     codec_context->extradata_size = config.extra_data().size();
     codec_context->extradata = reinterpret_cast<uint8_t*>(
         av_malloc(config.extra_data().size() + FF_INPUT_BUFFER_PADDING_SIZE));
     memcpy(codec_context->extradata, &config.extra_data()[0],
@@ skipping 317 matching lines @@
 }
 
 int32_t HashCodecName(const char* codec_name) {
   // Use the first 32-bits from the SHA1 hash as the identifier.
   int32_t hash;
   memcpy(&hash, base::SHA1HashString(codec_name).substr(0, 4).c_str(), 4);
   return hash;
 }
 
 }  // namespace media