Let ViEEncoder handle resolution changes.

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 10 matching lines @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
+#include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/layer_filtering_transport.h"
 #include "webrtc/test/run_loop.h"
 #include "webrtc/test/statistics.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/vcm_capturer.h"
 #include "webrtc/test/video_renderer.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 
 namespace {
@@ skipping 964 matching lines @@
     video_send_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTransportSequenceNumberUri,
                      test::kTransportSequenceNumberExtensionId));
   } else {
     video_send_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
   }
 
   video_encoder_config_.min_transmit_bitrate_bps =
       params_.common.min_transmit_bps;
-  video_encoder_config_.streams = params_.ss.streams;
+  test::FillEncoderConfiguration(params_.ss.streams.size(),
+                                 &video_encoder_config_);
   video_encoder_config_.spatial_layers = params_.ss.spatial_layers;
 
   CreateMatchingReceiveConfigs(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[payload_type].ssrc = kSendRtxSsrcs[i];
     video_receive_configs_[i].rtp.rtx[payload_type].payload_type =
         kSendRtxPayloadType;
     video_receive_configs_[i].rtp.transport_cc = params_.common.send_side_bwe;
@@ skipping 278 matching lines @@
     audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
     audio_config.rtcp_send_transport = &transport;
     audio_config.voe_channel_id = voe.receive_channel_id;
     audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
     audio_config.rtp.transport_cc = params_.common.send_side_bwe;
     audio_config.rtp.extensions = audio_send_config_.rtp.extensions;
     audio_config.decoder_factory = decoder_factory_;
     if (params_.audio_video_sync)
       audio_config.sync_group = kSyncGroup;
 
-    audio_receive_stream =call->CreateAudioReceiveStream(audio_config);
+    audio_receive_stream = call->CreateAudioReceiveStream(audio_config);
 
     const CodecInst kOpusInst = {120, "OPUS", 48000, 960, 2, 64000};
     EXPECT_EQ(0, voe.codec->SetSendCodec(voe.send_channel_id, kOpusInst));
   }
 
   // Start sending and receiving video.
   video_receive_stream->Start();
   video_send_stream_->Start();
   capturer_->Start();
 
@@ skipping 33 matching lines @@
      call->DestroyAudioSendStream(audio_send_stream_);
      call->DestroyAudioReceiveStream(audio_receive_stream);
   }
 
   transport.StopSending();
   if (params_.audio)
     DestroyVoiceEngine(&voe);
 }
 
 }  // namespace webrtc