Remove dead code in IncomingVideoStream that's just causing contention.

webrtc/common_video/incoming_video_stream.cc

 /*
  *  Copyright (c) 2012 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/common_video/include/incoming_video_stream.h"
 
 #include <assert.h>
 
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common_video/video_render_frames.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 
 namespace webrtc {
 
 IncomingVideoStream::IncomingVideoStream(bool disable_prerenderer_smoothing)
     : disable_prerenderer_smoothing_(disable_prerenderer_smoothing),
       incoming_render_thread_(),
       deliver_buffer_event_(EventTimerWrapper::Create()),
       running_(false),
       external_callback_(nullptr),
-      render_buffers_(new VideoRenderFrames()),
-      incoming_rate_(0),
-      last_rate_calculation_time_ms_(0),
-      num_frames_since_last_calculation_(0) {}
+      render_buffers_(new VideoRenderFrames()) {}
 
 IncomingVideoStream::~IncomingVideoStream() {
   Stop();
 }
 
 void IncomingVideoStream::OnFrame(const VideoFrame& video_frame) {
   rtc::CritScope csS(&stream_critsect_);
 
   if (!running_) {
     return;
   }
 
-  // Rate statistics.
-  num_frames_since_last_calculation_++;
-  int64_t now_ms = rtc::TimeMillis();
-  if (now_ms >= last_rate_calculation_time_ms_ + kFrameRatePeriodMs) {
-    incoming_rate_ =
-        static_cast<uint32_t>(1000 * num_frames_since_last_calculation_ /
-                              (now_ms - last_rate_calculation_time_ms_));
-    num_frames_since_last_calculation_ = 0;
-    last_rate_calculation_time_ms_ = now_ms;
-  }
-
   // Hand over or insert frame.
   if (disable_prerenderer_smoothing_) {
     DeliverFrame(video_frame);
   } else {
     rtc::CritScope csB(&buffer_critsect_);
     if (render_buffers_->AddFrame(video_frame) == 1) {
       deliver_buffer_event_->Set();
     }
   }
 }
@@ skipping 60 matching lines @@
     }
   }
   if (thread) {
     thread->Stop();
     delete thread;
   }
   running_ = false;
   return 0;
 }
 
-int32_t IncomingVideoStream::Reset() {
-  rtc::CritScope cs_buffer(&buffer_critsect_);
-  render_buffers_->ReleaseAllFrames();
-  return 0;
-}
-
-uint32_t IncomingVideoStream::IncomingRate() const {
-  rtc::CritScope cs(&stream_critsect_);
-  return incoming_rate_;
-}
-
 bool IncomingVideoStream::IncomingVideoStreamThreadFun(void* obj) {
   return static_cast<IncomingVideoStream*>(obj)->IncomingVideoStreamProcess();
 }
 
 bool IncomingVideoStream::IncomingVideoStreamProcess() {
   if (kEventError != deliver_buffer_event_->Wait(kEventMaxWaitTimeMs)) {
     rtc::CritScope cs(&thread_critsect_);
     if (incoming_render_thread_ == NULL) {
       // Terminating
       return false;
@@ skipping 23 matching lines @@
 void IncomingVideoStream::DeliverFrame(const VideoFrame& video_frame) {
   rtc::CritScope cs(&thread_critsect_);
 
   // Send frame for rendering.
   if (external_callback_) {
     external_callback_->OnFrame(video_frame);
   }
 }
 
 }  // namespace webrtc