Let ViEEncoder handle resolution changes.

webrtc/call/bitrate_estimator_tests.cc

 /*
  *  Copyright (c) 2013 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 <functional>
@@ skipping 108 matching lines @@
     receive_transport_.reset(new test::DirectTransport(receiver_call_.get()));
     receive_transport_->SetReceiver(sender_call_->Receiver());
 
     video_send_config_ = VideoSendStream::Config(send_transport_.get());
     video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
     // Encoders will be set separately per stream.
     video_send_config_.encoder_settings.encoder = nullptr;
     video_send_config_.encoder_settings.payload_name = "FAKE";
     video_send_config_.encoder_settings.payload_type =
         kFakeVideoSendPayloadType;
-    video_encoder_config_.streams = test::CreateVideoStreams(1);
+    test::FillEncoderConfiguration(1, &video_encoder_config_);
 
     receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
     // receive_config_.decoders will be set by every stream separately.
     receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
     receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
     receive_config_.rtp.remb = true;
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
@@ skipping 27 matching lines @@
           audio_receive_stream_(nullptr),
           video_receive_stream_(nullptr),
           frame_generator_capturer_(),
           fake_encoder_(Clock::GetRealTimeClock()),
           fake_decoder_() {
       test_->video_send_config_.rtp.ssrcs[0]++;
       test_->video_send_config_.encoder_settings.encoder = &fake_encoder_;
       send_stream_ = test_->sender_call_->CreateVideoSendStream(
           test_->video_send_config_.Copy(),
           test_->video_encoder_config_.Copy());
-      RTC_DCHECK_EQ(1u, test_->video_encoder_config_.streams.size());
+      RTC_DCHECK_EQ(1u, test_->video_encoder_config_.number_of_streams);
       frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
-          test_->video_encoder_config_.streams[0].width,
-          test_->video_encoder_config_.streams[0].height, 30,
+          kDefaultWidth, kDefaultHeight, kDefaultFramerate,
           Clock::GetRealTimeClock()));
       send_stream_->SetSource(frame_generator_capturer_.get());
       send_stream_->Start();
       frame_generator_capturer_->Start();
 
       if (receive_audio) {
         AudioReceiveStream::Config receive_config;
         receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0];
         // Bogus non-default id to prevent hitting a RTC_DCHECK when creating
         // the AudioReceiveStream. Every receive stream has to correspond to
@@ skipping 133 matching lines @@
       RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc