Now expect the correct number of streams in EndToEndTest.GetStats.

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 2744 matching lines @@
             stats.current_payload_type == kFakeVideoSendPayloadType;
       }
 
       return AllStatsFilled(receive_stats_filled_);
     }
 
     bool CheckSendStats() {
       RTC_DCHECK(send_stream_);
       VideoSendStream::Stats stats = send_stream_->GetStats();
 
+      size_t expected_num_streams = kNumSsrcs + expected_send_ssrcs_.size();
       send_stats_filled_["NumStreams"] |=
-          stats.substreams.size() == expected_send_ssrcs_.size();
+          stats.substreams.size() == expected_num_streams;
 
       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;
@@ skipping 1283 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc