Measure encoding time on encode callbacks.

webrtc/video_send_stream.h

 /*
  *  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.
  */
 
@@ skipping 73 matching lines @@
     struct EncoderSettings {
       std::string ToString() const;
 
       std::string payload_name;
       int payload_type = -1;
 
       // TODO(sophiechang): Delete this field when no one is using internal
       // sources anymore.
       bool internal_source = false;
 
+      // Allow 100% encoder utilization. Used for HW encoders where CPU isn't
+      // expected to be the limiting factor, but a chip could be running at
+      // 30fps (for example) exactly.
+      bool full_overuse_time = false;
+
       // Uninitialized VideoEncoder instance to be used for encoding. Will be
       // initialized from inside the VideoSendStream.
       VideoEncoder* encoder = nullptr;
     } encoder_settings;
 
     static const size_t kDefaultMaxPacketSize = 1500 - 40;  // TCP over IPv4.
     struct Rtp {
       std::string ToString() const;
 
       std::vector<uint32_t> ssrcs;
@@ skipping 75 matching lines @@
   // in the config. Encoder settings are passed on to the encoder instance along
   // with the VideoStream settings.
   virtual bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0;
 
   virtual Stats GetStats() = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_SEND_STREAM_H_