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 1706 matching lines @@
   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 = {{120, {"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();
 
   // TODO(ivica): Remove bitrate_config and use the default Call::Config(), to
   // match the full stack tests.
   webrtc::RtcEventLogNullImpl event_log;
   Call::Config call_config(&event_log_);
   call_config.bitrate_config = params_.call.call_bitrate_config;
 
   ::VoiceEngineState voe;
   if (params_.audio.enabled) {
     CreateVoiceEngine(&voe, decoder_factory_);
     AudioState::Config audio_state_config;
     audio_state_config.voice_engine = voe.voice_engine;
     audio_state_config.audio_mixer = AudioMixerImpl::Create();
     call_config.audio_state = AudioState::Create(audio_state_config);
   }
 
   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;
   std::unique_ptr<test::VideoRenderer> local_preview;
   std::unique_ptr<test::VideoRenderer> loopback_video;
   if (params_.video.enabled) {
     // Create video renderers.
     local_preview.reset(test::VideoRenderer::Create(
         "Local Preview", params_.video.width, params_.video.height));
 
     size_t stream_id = params_.ss.selected_stream;
     std::string title = "Loopback Video";
     if (params_.ss.streams.size() > 1) {
       std::ostringstream s;
       s << stream_id;
       title += " - Stream #" + s.str();
     }
 
     loopback_video.reset(test::VideoRenderer::Create(
         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)
       video_receive_configs_[stream_id].sync_group = kSyncGroup;
 
     if (params_.screenshare.enabled)
       SetupScreenshareOrSVC();
 
     video_send_stream_ = call->CreateVideoSendStream(
         video_send_config_.Copy(), video_encoder_config_.Copy());
     if (params_.video.flexfec) {
       RTC_DCHECK(!flexfec_receive_configs_.empty());
       flexfec_receive_stream =
           call->CreateFlexfecReceiveStream(flexfec_receive_configs_[0]);
     }
     video_receive_stream = call->CreateVideoReceiveStream(
         video_receive_configs_[stream_id].Copy());
     CreateCapturer();
     video_send_stream_->SetSource(video_capturer_.get(),
                                   degradation_preference_);
   }
 
   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);
   StartEncodedFrameLogs(video_send_stream_);
 
   // Start sending and receiving video.
   if (params_.video.enabled) {
     if (flexfec_receive_stream)
       flexfec_receive_stream->Start();
     video_receive_stream->Start();
@@ skipping 31 matching lines @@
     video_send_stream_->Stop();
     video_receive_stream->Stop();
     if (flexfec_receive_stream) {
       flexfec_receive_stream->Stop();
       call->DestroyFlexfecReceiveStream(flexfec_receive_stream);
     }
     call->DestroyVideoReceiveStream(video_receive_stream);
     call->DestroyVideoSendStream(video_send_stream_);
   }
 
-  transport.StopSending();
+  audio_transport.StopSending();
+  video_transport.StopSending();
   if (params_.audio.enabled)
     DestroyVoiceEngine(&voe);
 }
 
 void VideoQualityTest::StartEncodedFrameLogs(VideoSendStream* stream) {
   if (!params_.video.encoded_frame_base_path.empty()) {
     std::ostringstream str;
     str << send_logs_++;
     std::string prefix =
         params_.video.encoded_frame_base_path + "." + str.str() + ".send.";
@@ skipping 10 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