Fix deadlock in webrtc_perf_tests

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 921 matching lines @@
       copy.set_timestamp(copy.ntp_time_ms() * 90);
       analyzer_->AddCapturedFrameForComparison(copy);
       rtc::CritScope lock(&crit_);
       if (send_stream_input_)
         send_stream_input_->OnFrame(copy);
     }
 
     // Called when |send_stream_.SetSource()| is called.
     void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
                          const rtc::VideoSinkWants& wants) override {
-      rtc::CritScope lock(&crit_);
-      RTC_DCHECK(!send_stream_input_ || send_stream_input_ == sink);
-      send_stream_input_ = sink;
+      {
+        rtc::CritScope lock(&crit_);
+        RTC_DCHECK(!send_stream_input_ || send_stream_input_ == sink);
+        send_stream_input_ = sink;
+      }
       if (video_capturer_) {
         video_capturer_->AddOrUpdateSink(this, wants);
       }
     }
 
     // Called by |send_stream_| when |send_stream_.SetSource()| is called.
     void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {
       rtc::CritScope lock(&crit_);
       RTC_DCHECK(sink == send_stream_input_);
       send_stream_input_ = nullptr;
@@ skipping 1055 matching lines @@
   if (!params_.video.encoded_frame_base_path.empty()) {
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 }  // namespace webrtc