Deprecate RtpSender::SendPadData with provided timestamps.

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 515 matching lines @@
     if (!packet)
       break;
     size_t payload_size = packet->payload_size();
     if (!PrepareAndSendPacket(std::move(packet), true, false, probe_cluster_id))
       break;
     bytes_left -= payload_size;
   }
   return bytes_to_send - bytes_left;
 }
 
+size_t RTPSender::SendPadData(size_t bytes, int probe_cluster_id) {
+  return DeprecatedSendPadData(bytes, false, 0, 0, probe_cluster_id);
+}
+
 size_t RTPSender::SendPadData(size_t bytes,
                               bool timestamp_provided,
                               uint32_t timestamp,
                               int64_t capture_time_ms) {
-  return SendPadData(bytes, timestamp_provided, timestamp, capture_time_ms,
-                     PacketInfo::kNotAProbe);
+  return DeprecatedSendPadData(bytes, timestamp_provided, timestamp,
+                               capture_time_ms, PacketInfo::kNotAProbe);
 }
 
-size_t RTPSender::SendPadData(size_t bytes,
-                              bool timestamp_provided,
-                              uint32_t timestamp,
-                              int64_t capture_time_ms,
-                              int probe_cluster_id) {
+size_t RTPSender::DeprecatedSendPadData(size_t bytes,
+                                        bool timestamp_provided,
+                                        uint32_t timestamp,
+                                        int64_t capture_time_ms,
+                                        int probe_cluster_id) {
   // Always send full padding packets. This is accounted for by the
   // RtpPacketSender,
   // which will make sure we don't send too much padding even if a single packet
   // is larger than requested.
   size_t padding_bytes_in_packet =
       std::min(MaxDataPayloadLength(), kMaxPaddingLength);
   size_t bytes_sent = 0;
   bool using_transport_seq =
       IsRtpHeaderExtensionRegistered(kRtpExtensionTransportSequenceNumber) &&
       transport_sequence_number_allocator_;
@@ skipping 306 matching lines @@
   video_->GenericFECStatus(&fec_enabled, &pt_red, &pt_fec);
   return fec_enabled && packet.PayloadType() == pt_red &&
          packet.payload()[0] == pt_fec;
 }
 
 size_t RTPSender::TimeToSendPadding(size_t bytes, int probe_cluster_id) {
   if (audio_configured_ || bytes == 0)
     return 0;
   size_t bytes_sent = TrySendRedundantPayloads(bytes, probe_cluster_id);
   if (bytes_sent < bytes)
-    bytes_sent +=
-        SendPadData(bytes - bytes_sent, false, 0, 0, probe_cluster_id);
+    bytes_sent += SendPadData(bytes - bytes_sent, probe_cluster_id);
   return bytes_sent;
 }
 
 bool RTPSender::SendToNetwork(uint8_t* buffer,
                               size_t payload_length,
                               size_t rtp_header_length,
                               int64_t capture_time_ms,
                               StorageType storage,
                               RtpPacketSender::Priority priority) {
   size_t length = payload_length + rtp_header_length;
@@ skipping 881 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 }  // namespace webrtc