Style cleanups in RtpSender.

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 29 matching lines @@
 class RtcEventLog;
 
 class RTPSenderInterface {
  public:
   RTPSenderInterface() {}
   virtual ~RTPSenderInterface() {}
 
   virtual uint32_t SSRC() const = 0;
   virtual uint32_t Timestamp() const = 0;
 
+  // Deprecated version of BuildRtpHeader(). |timestamp_provided| and
+  // |inc_sequence_number| are ignored.
+  // TODO(sergeyu): Remove this method.
   virtual int32_t BuildRTPheader(uint8_t* data_buffer,
                                  int8_t payload_type,
                                  bool marker_bit,
                                  uint32_t capture_timestamp,
                                  int64_t capture_time_ms,
                                  bool timestamp_provided = true,
                                  bool inc_sequence_number = true) = 0;
 
+  virtual int32_t BuildRtpHeader(uint8_t* data_buffer,
+                                 int8_t payload_type,
+                                 bool marker_bit,
+                                 uint32_t capture_timestamp,
+                                 int64_t capture_time_ms) = 0;
+
   // This returns the expected header length taking into consideration
   // the optional RTP header extensions that may not be currently active.
   virtual size_t RtpHeaderLength() const = 0;
   // Returns the next sequence number to use for a packet and allocates
   // 'packets_to_send' number of sequence numbers. It's important all allocated
   // sequence numbers are used in sequence to avoid perceived packet loss.
   virtual uint16_t AllocateSequenceNumber(uint16_t packets_to_send) = 0;
   virtual uint16_t SequenceNumber() const = 0;
   virtual size_t MaxPayloadLength() const = 0;
   virtual size_t MaxDataPayloadLength() const = 0;
@@ skipping 77 matching lines @@
 
   void SetMaxPayloadLength(size_t max_payload_length);
 
   int32_t SendOutgoingData(FrameType frame_type,
                            int8_t payload_type,
                            uint32_t timestamp,
                            int64_t capture_time_ms,
                            const uint8_t* payload_data,
                            size_t payload_size,
                            const RTPFragmentationHeader* fragmentation,
-                           const RTPVideoHeader* rtp_hdr = NULL);
+                           const RTPVideoHeader* rtp_header);
 
   // RTP header extension
   int32_t SetTransmissionTimeOffset(int32_t transmission_time_offset);
   int32_t SetAbsoluteSendTime(uint32_t absolute_send_time);
   void SetVideoRotation(VideoRotation rotation);
   int32_t SetTransportSequenceNumber(uint16_t sequence_number);
 
   int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
   bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override;
   int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
 
   size_t RtpHeaderExtensionLength() const;
 
-  uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer, bool marker_bit) const
+  uint16_t BuildRtpHeaderExtension(uint8_t* data_buffer, bool marker_bit) const
       EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
 
   uint8_t BuildTransmissionTimeOffsetExtension(uint8_t* data_buffer) const
       EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
   uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const
       EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
   uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const
       EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
   uint8_t BuildVideoRotationExtension(uint8_t* data_buffer) const
       EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
@@ skipping 64 matching lines @@
   void SetRtxSsrc(uint32_t ssrc);
 
   void SetRtxPayloadType(int payload_type, int associated_payload_type);
 
   // Functions wrapping RTPSenderInterface.
   int32_t BuildRTPheader(uint8_t* data_buffer,
                          int8_t payload_type,
                          bool marker_bit,
                          uint32_t capture_timestamp,
                          int64_t capture_time_ms,
-                         const bool timestamp_provided = true,
-                         const bool inc_sequence_number = true) override;
+                         bool timestamp_provided = true,
+                         bool inc_sequence_number = true) override;
+  int32_t BuildRtpHeader(uint8_t* data_buffer,
+                         int8_t payload_type,
+                         bool marker_bit,
+                         uint32_t capture_timestamp,
+                         int64_t capture_time_ms) override;
 
   size_t RtpHeaderLength() const override;
   uint16_t AllocateSequenceNumber(uint16_t packets_to_send) override;
   size_t MaxPayloadLength() const override;
 
   // Current timestamp.
   uint32_t Timestamp() const override;
   uint32_t SSRC() const override;
 
   int32_t SendToNetwork(uint8_t* data_buffer,
@@ skipping 35 matching lines @@
                         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(size_t bytes,
                      bool timestamp_provided,
                      uint32_t timestamp,
                      int64_t capture_time_ms);
-
   size_t SendPadData(size_t bytes,
                      bool timestamp_provided,
                      uint32_t timestamp,
                      int64_t capture_time_ms,
                      int probe_cluster_id);
 
   // Called on update of RTP statistics.
   void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
   StreamDataCountersCallback* GetRtpStatisticsCallback() const;
 
@@ skipping 162 matching lines @@
   std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
 
   RateLimiter* const retransmission_rate_limiter_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_