To M56: Use ffmpeg for opus decoding, no need to maintain our decoder.

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 311 matching lines @@
   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);
 
   int sample_rate = codec_context->sample_rate;
   switch (codec) {
-    case kCodecOpus:
-      // |codec_context->sample_fmt| is not set by FFmpeg because Opus decoding
-      // is not enabled in FFmpeg.  It doesn't matter what value is set here, so
-      // long as it's valid, the true sample format is selected inside the
-      // decoder.
-      sample_format = kSampleFormatF32;
-
-      // Always use 48kHz for OPUS.  Technically we should match to the highest
-      // supported hardware sample rate among [8, 12, 16, 24, 48] kHz, but we
-      // don't know the hardware sample rate at this point and those rates are
-      // rarely used for output.  See the "Input Sample Rate" section of the
-      // spec: http://tools.ietf.org/html/draft-terriberry-oggopus-01#page-11
-      sample_rate = 48000;
-      break;
-
     // 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.
       sample_format = kSampleFormatS16;
 #else
       NOTREACHED();
@@ skipping 23 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);
 
+#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.
-  switch (codec) {
-    case kCodecOpus:
-#if BUILDFLAG(ENABLE_AC3_EAC3_AUDIO_DEMUXING)
-    case kCodecAC3:
-    case kCodecEAC3:
-#endif
-      break;
-    default:
-      DCHECK_EQ(av_get_bytes_per_sample(codec_context->sample_fmt) * 8,
-                config->bits_per_channel());
-      break;
-  }
-
+  DCHECK_EQ(av_get_bytes_per_sample(codec_context->sample_fmt) * 8,
+            config->bits_per_channel());
   return true;
 }
 
 std::unique_ptr<AVCodecContext, ScopedPtrAVFreeContext>
 AVStreamToAVCodecContext(const AVStream* stream) {
   std::unique_ptr<AVCodecContext, ScopedPtrAVFreeContext> codec_context(
       avcodec_alloc_context3(nullptr));
   if (avcodec_parameters_to_context(codec_context.get(), stream->codecpar) <
       0) {
     return nullptr;
@@ skipping 398 matching lines @@
 TEST_COLORSPACE(RESERVED);
 TEST_COLORSPACE(FCC);
 TEST_COLORSPACE(BT470BG);
 TEST_COLORSPACE(SMPTE170M);
 TEST_COLORSPACE(SMPTE240M);
 TEST_COLORSPACE(YCOCG);
 TEST_COLORSPACE(BT2020_NCL);
 TEST_COLORSPACE(BT2020_CL);
 
 }  // namespace media