Enable probing with repeated payload packets by default.

webrtc/modules/rtp_rtcp/source/rtp_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 279 matching lines @@
                            uint8_t payload_type_red,
                            uint8_t payload_type_fec);
 
   void GenericFECStatus(bool* enable,
                         uint8_t* payload_type_red,
                         uint8_t* payload_type_fec) const;
 
   int32_t SetFecParameters(const FecProtectionParams *delta_params,
                            const FecProtectionParams *key_params);
 
-  size_t SendPadData(uint32_t timestamp,
-                     int64_t capture_time_ms,
-                     size_t bytes);
+  size_t SendPadData(size_t bytes,
+                     bool timestamp_provided,
+                     uint32_t timestamp,
+                     int64_t capture_time_ms);
 
   // Called on update of RTP statistics.
   void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
   StreamDataCountersCallback* GetRtpStatisticsCallback() const;
 
   uint32_t BitrateSent() const;
 
   void SetRtpState(const RtpState& rtp_state);
   RtpState GetRtpState() const;
   void SetRtxRtpState(const RtpState& rtp_state);
@@ skipping 21 matching lines @@
 
   bool PrepareAndSendPacket(uint8_t* buffer,
                             size_t length,
                             int64_t capture_time_ms,
                             bool send_over_rtx,
                             bool is_retransmit);
 
   // Return the number of bytes sent.  Note that both of these functions may
   // return a larger value that their argument.
   size_t TrySendRedundantPayloads(size_t bytes);
-  size_t TrySendPadData(size_t bytes);
 
-  size_t BuildPaddingPacket(uint8_t* packet, size_t header_length);
+  void BuildPaddingPacket(uint8_t* packet,
+                          size_t header_length,
+                          size_t padding_length);
 
   void BuildRtxPacket(uint8_t* buffer, size_t* length,
                       uint8_t* buffer_rtx);
 
   bool SendPacketToNetwork(const uint8_t *packet, size_t size);
 
   void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
 
   // Find the byte position of the RTP extension as indicated by |type| in
   // |rtp_packet|. Return false if such extension doesn't exist.
@@ skipping 104 matching lines @@
   // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember
   // that by the time the function returns there is no guarantee
   // that the target bitrate is still valid.
   rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
   uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_