Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData

webrtc/modules/rtp_rtcp/source/rtp_sender_video.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 58 matching lines @@
   strncpy(payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1);
   payload->typeSpecific.Video.videoCodecType = video_type;
   payload->audio = false;
   return payload;
 }
 
 void RTPSenderVideo::SendVideoPacket(uint8_t* data_buffer,
                                      size_t payload_length,
                                      size_t rtp_header_length,
                                      uint16_t seq_num,
-                                     uint32_t capture_timestamp,
+                                     uint32_t rtp_timestamp,
                                      int64_t capture_time_ms,
                                      StorageType storage) {
   if (!rtp_sender_->SendToNetwork(data_buffer, payload_length,
                                   rtp_header_length, capture_time_ms, storage,
                                   RtpPacketSender::kLowPriority)) {
     LOG(LS_WARNING) << "Failed to send video packet " << seq_num;
     return;
   }
   rtc::CritScope cs(&stats_crit_);
   video_bitrate_.Update(payload_length + rtp_header_length,
                         clock_->TimeInMilliseconds());
   TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                       "Video::PacketNormal", "timestamp", capture_timestamp,
+                       "Video::PacketNormal", "timestamp", rtp_timestamp,
                        "seqnum", seq_num);
 }
 
 void RTPSenderVideo::SendVideoPacketAsRed(uint8_t* data_buffer,
                                           size_t payload_length,
                                           size_t rtp_header_length,
                                           uint16_t media_seq_num,
-                                          uint32_t capture_timestamp,
+                                          uint32_t rtp_timestamp,
                                           int64_t capture_time_ms,
                                           StorageType media_packet_storage,
                                           bool protect) {
   std::unique_ptr<RedPacket> red_packet;
   std::vector<std::unique_ptr<RedPacket>> fec_packets;
   StorageType fec_storage = kDontRetransmit;
   uint16_t next_fec_sequence_number = 0;
   {
     // Only protect while creating RED and FEC packets, not when sending.
     rtc::CritScope cs(&crit_);
@@ skipping 15 matching lines @@
         fec_storage = kAllowRetransmission;
     }
   }
   if (rtp_sender_->SendToNetwork(
           red_packet->data(), red_packet->length() - rtp_header_length,
           rtp_header_length, capture_time_ms, media_packet_storage,
           RtpPacketSender::kLowPriority)) {
     rtc::CritScope cs(&stats_crit_);
     video_bitrate_.Update(red_packet->length(), clock_->TimeInMilliseconds());
     TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                         "Video::PacketRed", "timestamp", capture_timestamp,
+                         "Video::PacketRed", "timestamp", rtp_timestamp,
                          "seqnum", media_seq_num);
   } else {
     LOG(LS_WARNING) << "Failed to send RED packet " << media_seq_num;
   }
   for (const auto& fec_packet : fec_packets) {
     if (rtp_sender_->SendToNetwork(
             fec_packet->data(), fec_packet->length() - rtp_header_length,
             rtp_header_length, capture_time_ms, fec_storage,
             RtpPacketSender::kLowPriority)) {
       rtc::CritScope cs(&stats_crit_);
       fec_bitrate_.Update(fec_packet->length(), clock_->TimeInMilliseconds());
       TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                           "Video::PacketFec", "timestamp", capture_timestamp,
+                           "Video::PacketFec", "timestamp", rtp_timestamp,
                            "seqnum", next_fec_sequence_number);
     } else {
       LOG(LS_WARNING) << "Failed to send FEC packet "
                       << next_fec_sequence_number;
     }
     ++next_fec_sequence_number;
   }
 }
 
 void RTPSenderVideo::SetGenericFECStatus(bool enable,
@@ skipping 41 matching lines @@
   RTC_DCHECK(key_params);
   if (fec_enabled_) {
     delta_fec_params_ = *delta_params;
     key_fec_params_ = *key_params;
   }
 }
 
 bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
                                FrameType frame_type,
                                int8_t payload_type,
-                               uint32_t capture_timestamp,
+                               uint32_t rtp_timestamp,
                                int64_t capture_time_ms,
                                const uint8_t* payload_data,
                                size_t payload_size,
                                const RTPFragmentationHeader* fragmentation,
                                const RTPVideoHeader* video_header) {
   if (payload_size == 0)
     return false;
 
   std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
       video_type, rtp_sender_->MaxDataPayloadLength(),
@@ skipping 40 matching lines @@
     uint8_t dataBuffer[IP_PACKET_SIZE] = {0};
     size_t payload_bytes_in_packet = 0;
 
     if (!packetizer->NextPacket(&dataBuffer[rtp_header_length],
                                 &payload_bytes_in_packet, &last)) {
       return false;
     }
 
     // Write RTP header.
     int32_t header_length = rtp_sender_->BuildRtpHeader(
-        dataBuffer, payload_type, last, capture_timestamp, capture_time_ms);
+        dataBuffer, payload_type, last, rtp_timestamp, capture_time_ms);
     if (header_length <= 0)
       return false;
 
     // According to
     // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
     // ts_126114v120700p.pdf Section 7.4.5:
     // The MTSI client shall add the payload bytes as defined in this clause
     // onto the last RTP packet in each group of packets which make up a key
     // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
     // (HEVC)). The MTSI client may also add the payload bytes onto the last RTP
@@ skipping 13 matching lines @@
       size_t packetSize = payload_size + rtp_header_length;
       RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
       RTPHeader rtp_header;
       rtp_parser.Parse(&rtp_header);
       rtp_sender_->UpdateVideoRotation(dataBuffer, packetSize, rtp_header,
                                        video_header->rotation);
     }
     if (red_payload_type != 0) {
       SendVideoPacketAsRed(dataBuffer, payload_bytes_in_packet,
                            rtp_header_length, rtp_sender_->SequenceNumber(),
-                           capture_timestamp, capture_time_ms, storage,
+                           rtp_timestamp, capture_time_ms, storage,
                            packetizer->GetProtectionType() == kProtectedPacket);
     } else {
       SendVideoPacket(dataBuffer, payload_bytes_in_packet, rtp_header_length,
-                      rtp_sender_->SequenceNumber(), capture_timestamp,
+                      rtp_sender_->SequenceNumber(), rtp_timestamp,
                       capture_time_ms, storage);
     }
 
     if (first_frame) {
       if (first) {
         LOG(LS_INFO)
             << "Sent first RTP packet of the first video frame (pre-pacer)";
       }
       if (last) {
         LOG(LS_INFO)
             << "Sent last RTP packet of the first video frame (pre-pacer)";
       }
     }
     first = false;
   }
 
   TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms, "timestamp",
-                         rtp_sender_->Timestamp());
+                         rtp_timestamp);
   return true;
 }
 
 uint32_t RTPSenderVideo::VideoBitrateSent() const {
   rtc::CritScope cs(&stats_crit_);
   return video_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0);
 }
 
 uint32_t RTPSenderVideo::FecOverheadRate() const {
   rtc::CritScope cs(&stats_crit_);
   return fec_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0);
 }
 
 int RTPSenderVideo::SelectiveRetransmissions() const {
   rtc::CritScope cs(&crit_);
   return retransmission_settings_;
 }
 
 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
   rtc::CritScope cs(&crit_);
   retransmission_settings_ = settings;
 }
 
 }  // namespace webrtc