Add VideoSendStream::Stats::prefered_media_bitrate_bps

webrtc/video/end_to_end_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 <algorithm>
@@ skipping 2729 matching lines @@
       send_stats_filled_["NumStreams"] |=
           stats.substreams.size() == expected_send_ssrcs_.size();
 
       send_stats_filled_["CpuOveruseMetrics"] |=
           stats.avg_encode_time_ms != 0 && stats.encode_usage_percent != 0;
 
       send_stats_filled_["EncoderImplementationName"] |=
           stats.encoder_implementation_name ==
           test::FakeEncoder::kImplementationName;
 
+      send_stats_filled_["EncoderPreferredBitrate"] |=
+          stats.preferred_media_bitrate_bps > 0;
+
       for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
                stats.substreams.begin();
            it != stats.substreams.end(); ++it) {
         if (expected_send_ssrcs_.find(it->first) == expected_send_ssrcs_.end())
           continue;  // Probably RTX.
 
         send_stats_filled_[CompoundKey("CapturedFrameRate", it->first)] |=
             stats.input_frame_rate != 0;
 
         const VideoSendStream::StreamStats& stream_stats = it->second;
@@ skipping 984 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc