Let ViEEncoder handle resolution changes.

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.
  */
 
 #ifndef WEBRTC_VIDEO_SEND_STREAM_H_
 #define WEBRTC_VIDEO_SEND_STREAM_H_
 
 #include <map>
 #include <string>
 #include <vector>
+#include <utility>
 
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/config.h"
 #include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/media/base/videosourceinterface.h"
 #include "webrtc/transport.h"
 
 namespace webrtc {
 
@@ skipping 19 matching lines @@
     RtcpStatistics rtcp_stats;
   };
 
   struct Stats {
     std::string ToString(int64_t time_ms) const;
     std::string encoder_implementation_name = "unknown";
     int input_frame_rate = 0;
     int encode_frame_rate = 0;
     int avg_encode_time_ms = 0;
     int encode_usage_percent = 0;
+    // Bitrate the encoder is currently configured to use due to bandwidth
+    // limitations.
     int target_media_bitrate_bps = 0;
+    // Bitrate the encoder is actually producing.
     int media_bitrate_bps = 0;
+    // Media bitrate this VideoSendStream is configured to prefer if there are
+    // no bandwidth limitations.
+    int preffered_media_bitrate_bps = 0;
     bool suspended = false;
     bool bw_limited_resolution = false;
     std::map<uint32_t, StreamStats> substreams;
   };
 
   struct Config {
    public:
     Config() = delete;
     Config(Config&&) = default;
     explicit Config(Transport* send_transport)
@@ skipping 124 matching lines @@
 
   virtual Stats GetStats() = 0;
 
  protected:
   virtual ~VideoSendStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_SEND_STREAM_H_