Propagate probing cluster id to SendTimeHistory.

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 525 matching lines @@
   } else if (frame_type == kVideoFrameDelta) {
     ++frame_counts_.delta_frames;
   }
   if (frame_count_observer_) {
     frame_count_observer_->FrameCountUpdated(frame_counts_, ssrc);
   }
 
   return ret_val;
 }
 
-size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send) {
+size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send,
+                                           int probe_cluster_id) {
   {
     rtc::CritScope lock(&send_critsect_);
     if (!sending_media_)
       return 0;
     if ((rtx_ & kRtxRedundantPayloads) == 0)
       return 0;
   }
 
   uint8_t buffer[IP_PACKET_SIZE];
   int bytes_left = static_cast<int>(bytes_to_send);
   while (bytes_left > 0) {
     size_t length = bytes_left;
     int64_t capture_time_ms;
     if (!packet_history_.GetBestFittingPacket(buffer, &length,
                                               &capture_time_ms)) {
       break;
     }
-    if (!PrepareAndSendPacket(buffer, length, capture_time_ms, true, false))
+    if (!PrepareAndSendPacket(buffer, length, capture_time_ms, true, false,
+                              probe_cluster_id))
       break;
     RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
     RTPHeader rtp_header;
     rtp_parser.Parse(&rtp_header);
     bytes_left -= static_cast<int>(length - rtp_header.headerLength);
   }
   return bytes_to_send - bytes_left;
 }
 
 void RTPSender::BuildPaddingPacket(uint8_t* packet,
@@ skipping 95 matching lines @@
     }
 
     UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms);
 
     PacketOptions options;
     if (AllocateTransportSequenceNumber(&options.packet_id)) {
       if (UpdateTransportSequenceNumber(options.packet_id, padding_packet,
                                         length, rtp_header)) {
         if (transport_feedback_observer_)
           transport_feedback_observer_->AddPacket(options.packet_id, length,
-                                                  true);

[stefan-webrtc, 2016/05/26 16:29:50]

Was this a bug? Should it have been covered by a test?

+                                                  PacketInfo::kNotAProbe);
       }
     }
 
     if (!SendPacketToNetwork(padding_packet, length, options))
       break;
 
     bytes_sent += padding_bytes_in_packet;
     UpdateRtpStats(padding_packet, length, rtp_header, over_rtx, false);
   }
 
@@ skipping 35 matching lines @@
         corrected_capture_tims_ms, length - header.headerLength, true);
 
     return length;
   }
   int rtx = kRtxOff;
   {
     rtc::CritScope lock(&send_critsect_);
     rtx = rtx_;
   }
   if (!PrepareAndSendPacket(data_buffer, length, capture_time_ms,
-                            (rtx & kRtxRetransmitted) > 0, true)) {
+                            (rtx & kRtxRetransmitted) > 0, true,
+                            PacketInfo::kNotAProbe)) {
     return -1;
   }
   return static_cast<int32_t>(length);
 }
 
 bool RTPSender::SendPacketToNetwork(const uint8_t* packet,
                                     size_t size,
                                     const PacketOptions& options) {
   int bytes_sent = -1;
   if (transport_) {
@@ skipping 115 matching lines @@
     nack_byte_count_[i + 1] = nack_byte_count_[i];
     nack_byte_count_times_[i + 1] = nack_byte_count_times_[i];
   }
   nack_byte_count_[0] = bytes;
   nack_byte_count_times_[0] = now;
 }
 
 // Called from pacer when we can send the packet.
 bool RTPSender::TimeToSendPacket(uint16_t sequence_number,
                                  int64_t capture_time_ms,
-                                 bool retransmission) {
+                                 bool retransmission,
+                                 int probe_cluster_id) {
   size_t length = IP_PACKET_SIZE;
   uint8_t data_buffer[IP_PACKET_SIZE];
   int64_t stored_time_ms;
 
   if (!packet_history_.GetPacketAndSetSendTime(sequence_number,
                                                0,
                                                retransmission,
                                                data_buffer,
                                                &length,
                                                &stored_time_ms)) {
     // Packet cannot be found. Allow sending to continue.
     return true;
   }
 
   int rtx;
   {
     rtc::CritScope lock(&send_critsect_);
     rtx = rtx_;
   }
-  return PrepareAndSendPacket(data_buffer,
-                              length,
-                              capture_time_ms,
+  return PrepareAndSendPacket(data_buffer, length, capture_time_ms,
                               retransmission && (rtx & kRtxRetransmitted) > 0,
-                              retransmission);
+                              retransmission, probe_cluster_id);
 }
 
 bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
                                      size_t length,
                                      int64_t capture_time_ms,
                                      bool send_over_rtx,
-                                     bool is_retransmit) {
+                                     bool is_retransmit,
+                                     int probe_cluster_id) {
   uint8_t* buffer_to_send_ptr = buffer;
 
   RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
   RTPHeader rtp_header;
   rtp_parser.Parse(&rtp_header);
   if (!is_retransmit && rtp_header.markerBit) {
     TRACE_EVENT_ASYNC_END0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PacedSend",
                            capture_time_ms);
   }
 
@@ skipping 12 matching lines @@
   UpdateTransmissionTimeOffset(buffer_to_send_ptr, length, rtp_header,
                                diff_ms);
   UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms);
 
   PacketOptions options;
   if (AllocateTransportSequenceNumber(&options.packet_id)) {
     if (UpdateTransportSequenceNumber(options.packet_id, buffer_to_send_ptr,
                                       length, rtp_header)) {
       if (transport_feedback_observer_)
         transport_feedback_observer_->AddPacket(options.packet_id, length,
-                                                true);

[stefan-webrtc, 2016/05/26 16:29:50]

Same here

+                                                probe_cluster_id);
     }
   }
 
   if (!is_retransmit && !send_over_rtx) {
     UpdateDelayStatistics(capture_time_ms, now_ms);
     UpdateOnSendPacket(options.packet_id, capture_time_ms, rtp_header.ssrc);
   }
 
   bool ret = SendPacketToNetwork(buffer_to_send_ptr, length, options);
   if (ret) {
@@ skipping 46 matching lines @@
   }
   bool fec_enabled;
   uint8_t pt_red;
   uint8_t pt_fec;
   video_->GenericFECStatus(&fec_enabled, &pt_red, &pt_fec);
   return fec_enabled &&
       header.payloadType == pt_red &&
       buffer[header.headerLength] == pt_fec;
 }
 
-size_t RTPSender::TimeToSendPadding(size_t bytes) {
+size_t RTPSender::TimeToSendPadding(size_t bytes, int probe_cluster_id) {
   if (audio_configured_ || bytes == 0)
     return 0;
-  size_t bytes_sent = TrySendRedundantPayloads(bytes);
+  size_t bytes_sent = TrySendRedundantPayloads(bytes, probe_cluster_id);
   if (bytes_sent < bytes)
     bytes_sent += SendPadData(bytes - bytes_sent, false, 0, 0);

[danilchap, 2016/05/31 08:58:24]

Padding can use cluster_id too.

   return bytes_sent;
 }
 
 // TODO(pwestin): send in the RtpHeaderParser to avoid parsing it again.
 int32_t RTPSender::SendToNetwork(uint8_t* buffer,
                                  size_t payload_length,
                                  size_t rtp_header_length,
                                  int64_t capture_time_ms,
                                  StorageType storage,
                                  RtpPacketSender::Priority priority) {
@@ skipping 37 matching lines @@
     }
     return 0;
   }
 
   PacketOptions options;
   if (AllocateTransportSequenceNumber(&options.packet_id)) {
     if (UpdateTransportSequenceNumber(options.packet_id, buffer, length,
                                       rtp_header)) {
       if (transport_feedback_observer_)
         transport_feedback_observer_->AddPacket(options.packet_id, length,
-                                                true);
+                                                PacketInfo::kNotAProbe);
     }
   }
   UpdateDelayStatistics(capture_time_ms, now_ms);
   UpdateOnSendPacket(options.packet_id, capture_time_ms, rtp_header.ssrc);
 
   bool sent = SendPacketToNetwork(buffer, length, options);
 
   // Mark the packet as sent in the history even if send failed. Dropping a
   // packet here should be treated as any other packet drop so we should be
   // ready for a retransmission.
@@ skipping 834 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc