Style cleanups in RtpSender.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  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 1222 matching lines @@
       ByteWriter<uint32_t>::WriteBigEndian(ptr, csrcs[i]);
       ptr += 4;
     }
     header[0] = (header[0] & 0xf0) | csrcs.size();
 
     // Update length of header.
     rtp_header_length += sizeof(uint32_t) * csrcs.size();
   }
 
   uint16_t len =
-      BuildRTPHeaderExtension(header + rtp_header_length, marker_bit);
+      BuildRtpHeaderExtension(header + rtp_header_length, marker_bit);
   if (len > 0) {
     header[0] |= 0x10;  // Set extension bit.
     rtp_header_length += len;
   }
   return rtp_header_length;
 }
 
 int32_t RTPSender::BuildRTPheader(uint8_t* data_buffer,
                                   int8_t payload_type,
                                   bool marker_bit,
                                   uint32_t capture_timestamp,
                                   int64_t capture_time_ms,
                                   bool timestamp_provided,
                                   bool inc_sequence_number) {
+  if (!timestamp_provided) {
+    rtc::CritScope lock(&send_critsect_);

[danilchap, 2016/06/15 20:04:21]

this is not a thread-safe way to keep old behavior.
But feel free to ignore timestamp_provided parameter.
(rtp_sender is not officially public interface, and the use I have in mind
doesn't really care what timestamp will be written)

[Sergey Ulanov, 2016/06/15 20:53:24]

Done.

+    // Calculate |capture_timestamp| so that BuildRtpHeader() effectively will
+    // set timestamp_ = timestamp_ + 1.
+    capture_timestamp = timestamp_ - start_timestamp_ + 1;
+  }
+  return BuildRtpHeader(data_buffer, payload_type, marker_bit,
+                        capture_timestamp, capture_time_ms);
+}
+
+int32_t RTPSender::BuildRtpHeader(uint8_t* data_buffer,
+                                  int8_t payload_type,
+                                  bool marker_bit,
+                                  uint32_t capture_timestamp,
+                                  int64_t capture_time_ms) {
   assert(payload_type >= 0);
   rtc::CritScope lock(&send_critsect_);
 
-  if (timestamp_provided) {
-    timestamp_ = start_timestamp_ + capture_timestamp;
-  } else {
-    // Make a unique time stamp.
-    // We can't inc by the actual time, since then we increase the risk of back
-    // timing.
-    timestamp_++;
-  }
+  timestamp_ = start_timestamp_ + capture_timestamp;
   last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
   uint32_t sequence_number = sequence_number_++;
   capture_time_ms_ = capture_time_ms;
   last_packet_marker_bit_ = marker_bit;
   return CreateRtpHeader(data_buffer, payload_type, ssrc_, marker_bit,
                          timestamp_, sequence_number, csrcs_);
 }
 
-uint16_t RTPSender::BuildRTPHeaderExtension(uint8_t* data_buffer,
+uint16_t RTPSender::BuildRtpHeaderExtension(uint8_t* data_buffer,
                                             bool marker_bit) const {
   if (rtp_header_extension_map_.Size() <= 0) {
     return 0;
   }
   // RTP header extension, RFC 3550.
   //   0                   1                   2                   3
   //   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   //  |      defined by profile       |           length              |
   //  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
@@ skipping 708 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc