Add audio streams to CallTest and a first A/V call test.

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 <stdio.h>
@@ skipping 763 matching lines @@
     params->ss.spatial_layers.push_back(layer);
   }
 }
 
 void VideoQualityTest::SetupCommon(Transport* send_transport,
                                    Transport* recv_transport) {
   if (params_.logs)
     trace_to_stderr_.reset(new test::TraceToStderr);
 
   size_t num_streams = params_.ss.streams.size();
-  CreateSendConfig(num_streams, send_transport);
+  CreateSendConfig(num_streams, 0, send_transport);
 
   int payload_type;
   if (params_.common.codec == "VP8") {
     encoder_.reset(VideoEncoder::Create(VideoEncoder::kVp8));
     payload_type = kPayloadTypeVP8;
   } else if (params_.common.codec == "VP9") {
     encoder_.reset(VideoEncoder::Create(VideoEncoder::kVp9));
     payload_type = kPayloadTypeVP9;
   } else {
     RTC_NOTREACHED() << "Codec not supported!";
@@ skipping 15 matching lines @@
   } else {
     video_send_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kAbsSendTime, test::kAbsSendTimeExtensionId));
   }
 
   video_encoder_config_.min_transmit_bitrate_bps =
       params_.common.min_transmit_bps;
   video_encoder_config_.streams = params_.ss.streams;
   video_encoder_config_.spatial_layers = params_.ss.spatial_layers;
 
-  CreateMatchingReceiveConfigs(recv_transport);
+  CreateMatchingReceiveConfigs(false, recv_transport);
 
   for (size_t i = 0; i < num_streams; ++i) {
     video_receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
     video_receive_configs_[i].rtp.rtx[kSendRtxPayloadType].ssrc =
         kSendRtxSsrcs[i];
     video_receive_configs_[i].rtp.rtx[kSendRtxPayloadType].payload_type =
         kSendRtxPayloadType;
     video_receive_configs_[i].rtp.transport_cc = params_.common.send_side_bwe;
   }
 }
@@ skipping 126 matching lines @@
             "Warning: Calculating PSNR and SSIM for downsized resolution "
             "not implemented yet! Skipping PSNR and SSIM calculations!");
   }
 
   VideoAnalyzer analyzer(
       &send_transport, params_.analyzer.test_label,
       disable_quality_check ? -1.1 : params_.analyzer.avg_psnr_threshold,
       disable_quality_check ? -1.1 : params_.analyzer.avg_ssim_threshold,
       params_.analyzer.test_durations_secs * params_.common.fps,
       graph_data_output_file, graph_title,
-      kSendSsrcs[params_.ss.selected_stream]);
+      kVideoSendSsrcs[params_.ss.selected_stream]);
 
   analyzer.SetReceiver(receiver_call_->Receiver());
   send_transport.SetReceiver(&analyzer);
   recv_transport.SetReceiver(sender_call_->Receiver());
 
   SetupCommon(&analyzer, &recv_transport);
   video_send_config_.encoding_time_observer = &analyzer;
   video_receive_configs_[params_.ss.selected_stream].renderer = &analyzer;
   for (auto& config : video_receive_configs_)
     config.pre_decode_callback = &analyzer;
 
   if (params_.screenshare.enabled)
     SetupScreenshare();
 
-  CreateStreams();
+  CreateVideoStreams();
   analyzer.input_ = video_send_stream_->Input();
   analyzer.send_stream_ = video_send_stream_;
 
   CreateCapturer(&analyzer);
 
   video_send_stream_->Start();
   for (VideoReceiveStream* receive_stream : video_receive_streams_)
     receive_stream->Start();
   capturer_->Start();
 
@@ skipping 71 matching lines @@
   video_send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(video_send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc