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..c5a267bd385da03bda2824dd0179a39ddc408f0c 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -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 = {{120, {"opus", 48000, 2}}};
   if (params_.video.enabled && params_.audio.sync_video)
     audio_config.sync_group = kSyncGroup;
 
@@ -1751,15 +1752,20 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
 
   std::unique_ptr<Call> call(Call::Create(call_config));
 
-  // TODO(minyue): consider if this is a good transport even for audio only
-  // calls.
-  test::LayerFilteringTransport transport(
+  test::LayerFilteringTransport video_transport(
       params.pipe, call.get(), kPayloadTypeVP8, kPayloadTypeVP9,
       params.video.selected_tl, params_.ss.selected_sl);
+  MediaTypePacketReceiver video_receiver(call->Receiver(), MediaType::VIDEO);
+
   // TODO(ivica): Use two calls to be able to merge with RunWithAnalyzer or at
   // least share as much code as possible. That way this test would also match
   // the full stack tests better.
-  transport.SetReceiver(call->Receiver());
+  video_transport.SetReceiver(&video_receiver);
+
+  test::DirectTransport audio_transport(params.pipe, call.get(),
+                                        MediaType::AUDIO);
+  MediaTypePacketReceiver audio_receiver(call->Receiver(), MediaType::AUDIO);

[nisse-webrtc, 2017/04/04 09:57:27]

If it's ok to use two separate transports in this test, I don't think you need
MediaTypePacketReceiver. Instead, make sure each transport is configured to
generate packets with the correct media type.

To use the same transport for both audio and video, you'd need a workaround
similar to CallTest::PayloadDemuxer.

[stefan-webrtc, 2017/04/05 08:52:17]

Let's not introduce an additional transport. We want to be able to test audio
and video over the same bottleneck.

+  audio_transport.SetReceiver(&audio_receiver);
 
   VideoReceiveStream* video_receive_stream = nullptr;
   FlexfecReceiveStream* flexfec_receive_stream = nullptr;
@@ -1782,7 +1788,7 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
         title.c_str(), params_.ss.streams[stream_id].width,
         params_.ss.streams[stream_id].height));
 
-    SetupVideo(&transport, &transport);
+    SetupVideo(&video_transport, &video_transport);
     video_send_config_.pre_encode_callback = local_preview.get();
     video_receive_configs_[stream_id].renderer = loopback_video.get();
     if (params_.audio.enabled && params_.audio.sync_video)
@@ -1808,7 +1814,7 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
   AudioReceiveStream* audio_receive_stream = nullptr;
   if (params_.audio.enabled) {
     SetupAudio(voe.send_channel_id, voe.receive_channel_id, call.get(),
-               &transport, &audio_receive_stream);
+               &audio_transport, &audio_receive_stream);
   }
 
   StartEncodedFrameLogs(video_receive_stream);
@@ -1860,7 +1866,8 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
     call->DestroyVideoSendStream(video_send_stream_);
   }
 
-  transport.StopSending();
+  audio_transport.StopSending();
+  video_transport.StopSending();
   if (params_.audio.enabled)
     DestroyVoiceEngine(&voe);
 }