Making FakeNetworkPipe demux audio and video packets.

webrtc/video/video_quality_test.cc

Index: webrtc/video/video_quality_test.cc
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 3fa07db7fdbcc3f5a3ce60193b019daeb5229ddf..7ebc2eabfab83eecafae2fd5920fe063c9fd60f2 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -1712,7 +1712,7 @@ void VideoQualityTest::SetupAudio(int send_channel_id,
     audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps;
   }
   audio_send_config_.send_codec_spec.codec_inst =
-      CodecInst{120, "OPUS", 48000, 960, 2, 64000};
+      CodecInst{103, "OPUS", 48000, 960, 2, 64000};

[minyue-webrtc, 2017/04/04 06:47:59]

Karl, 120 fails, which is weird to me. Please advice.

BTW, 120 works with the origin version of the relanded CL.

[minyue-webrtc, 2017/04/04 09:35:44]

Continuing bisecting reflected another cause of failure, besides line 1727.

But 120 worked after rebasing. I am not sure what could cause it to work again,
and I am too lazy to find error in the relanding CL, which should have been
fixed anyway.

   audio_send_config_.send_codec_spec.enable_opus_dtx = params_.audio.dtx;
   audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_);
 
@@ -1724,6 +1724,7 @@ void VideoQualityTest::SetupAudio(int send_channel_id,
   audio_config.rtp.transport_cc = params_.call.send_side_bwe;
   audio_config.rtp.extensions = audio_send_config_.rtp.extensions;
   audio_config.decoder_factory = decoder_factory_;
+  audio_config.decoder_map = {{103, {"OPUS", 48000, 2}}};
   if (params_.video.enabled && params_.audio.sync_video)
     audio_config.sync_group = kSyncGroup;