Improve bitrate probing for the audio-only case.

webrtc/modules/pacing/paced_sender.h

 /*
  *  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.
  */
 
@@ skipping 49 matching lines @@
   // encoding them). Bitrate sent may temporarily exceed target set by
   // UpdateBitrate() so that this limit will be upheld.
   static const int64_t kMaxQueueLengthMs;
   // Pacing-rate relative to our target send rate.
   // Multiplicative factor that is applied to the target bitrate to calculate
   // the number of bytes that can be transmitted per interval.
   // Increasing this factor will result in lower delays in cases of bitrate
   // overshoots from the encoder.
   static const float kDefaultPaceMultiplier;
 
-  static const size_t kMinProbePacketSize = 200;
-
   PacedSender(Clock* clock, PacketSender* packet_sender);
 
   virtual ~PacedSender();
 
   virtual void CreateProbeCluster(int bitrate_bps);
 
   // Temporarily pause all sending.
   void Pause();
 
   // Resume sending packets.
@@ skipping 94 matching lines @@
   uint32_t max_padding_bitrate_kbps_ GUARDED_BY(critsect_);
   uint32_t pacing_bitrate_kbps_ GUARDED_BY(critsect_);
 
   int64_t time_last_update_us_ GUARDED_BY(critsect_);
 
   std::unique_ptr<paced_sender::PacketQueue> packets_ GUARDED_BY(critsect_);
   uint64_t packet_counter_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_PACING_PACED_SENDER_H_