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

media/ffmpeg/ffmpeg_common.cc

Index: media/ffmpeg/ffmpeg_common.cc
diff --git a/media/ffmpeg/ffmpeg_common.cc b/media/ffmpeg/ffmpeg_common.cc
index 8c4a55c953111de0613d6692408f739b541428b5..2b6100bb913fb4c0efbf43ef3251e0499e7c6fd9 100644
--- a/media/ffmpeg/ffmpeg_common.cc
+++ b/media/ffmpeg/ffmpeg_common.cc
@@ -329,21 +329,6 @@ bool AVCodecContextToAudioDecoderConfig(
 
   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:
@@ -390,7 +375,6 @@ bool AVCodecContextToAudioDecoderConfig(
   // 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: