Remove audio/video distinction for probe packets.

webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc

 /*
  *  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 81 matching lines @@
 
 double RateCounter::BitrateWindowS() const {
   return static_cast<double>(window_size_us_) / (1000 * 1000);
 }
 
 Packet::Packet()
     : flow_id_(0),
       creation_time_us_(-1),
       send_time_us_(-1),
       sender_timestamp_us_(-1),
-      payload_size_(0),
-      paced_(false) {
-}
+      payload_size_(0) {}
 
 Packet::Packet(int flow_id, int64_t send_time_us, size_t payload_size)
     : flow_id_(flow_id),
       creation_time_us_(send_time_us),
       send_time_us_(send_time_us),
       sender_timestamp_us_(send_time_us),
-      payload_size_(payload_size),
-      paced_(false) {
-}
+      payload_size_(payload_size) {}
 
 Packet::~Packet() {
 }
 
 bool Packet::operator<(const Packet& rhs) const {
   return send_time_us_ < rhs.send_time_us_;
 }
 
 void Packet::set_send_time_us(int64_t send_time_us) {
   assert(send_time_us >= 0);
@@ skipping 685 matching lines @@
 uint32_t PeriodicKeyFrameSource::NextPacketSize(uint32_t frame_size,
                                                 uint32_t remaining_payload) {
   uint32_t fragments =
       (frame_size + (kMaxPayloadSizeBytes - 1)) / kMaxPayloadSizeBytes;
   uint32_t avg_size = (frame_size + fragments - 1) / fragments;
   return std::min(avg_size, remaining_payload);
 }
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc