Making FakeNetworkPipe demux audio and video packets.

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include "webrtc/video/video_quality_test.h"
@@ skipping 1694 matching lines @@
   // adaptation.
   audio_send_config_.rtp.extensions.clear();
   if (params_.call.send_side_bwe) {
     audio_send_config_.rtp.extensions.push_back(
         webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri,
                              test::kTransportSequenceNumberExtensionId));
     audio_send_config_.min_bitrate_bps = kOpusMinBitrateBps;
     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_);
 
   AudioReceiveStream::Config audio_config;
   audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
   audio_config.rtcp_send_transport = transport;
   audio_config.voe_channel_id = receive_channel_id;
   audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
   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;
 
   *audio_receive_stream = call->CreateAudioReceiveStream(audio_config);
 }
 
 void VideoQualityTest::RunWithRenderers(const Params& params) {
   params_ = params;
   CheckParams();
 
@@ skipping 147 matching lines @@
   if (!params_.video.encoded_frame_base_path.empty()) {
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 }  // namespace webrtc