Add flag enabling more packets to be retransmittable.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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 13 matching lines @@
 #ifdef _WIN32
 // Disable warning C4355: 'this' : used in base member initializer list.
 #pragma warning(disable : 4355)
 #endif
 
 namespace webrtc {
 namespace {
 const int64_t kRtpRtcpMaxIdleTimeProcessMs = 5;
 const int64_t kRtpRtcpRttProcessTimeMs = 1000;
 const int64_t kRtpRtcpBitrateProcessTimeMs = 10;
+const int64_t kDefaultExpectedRetransmissionTimeMs = 125;
 }  // namespace
 
 RTPExtensionType StringToRtpExtensionType(const std::string& extension) {
   if (extension == RtpExtension::kTimestampOffsetUri)
     return kRtpExtensionTransmissionTimeOffset;
   if (extension == RtpExtension::kAudioLevelUri)
     return kRtpExtensionAudioLevel;
   if (extension == RtpExtension::kAbsSendTimeUri)
     return kRtpExtensionAbsoluteSendTime;
   if (extension == RtpExtension::kVideoRotationUri)
@@ skipping 379 matching lines @@
     const uint8_t* payload_data,
     size_t payload_size,
     const RTPFragmentationHeader* fragmentation,
     const RTPVideoHeader* rtp_video_header,
     uint32_t* transport_frame_id_out) {
   rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
   // Make sure an RTCP report isn't queued behind a key frame.
   if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
       rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
   }
+  int64_t expected_retransmission_time_ms = rtt_ms();
+  if (expected_retransmission_time_ms == 0) {
+    if (rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), nullptr,
+                           &expected_retransmission_time_ms, nullptr,

[danilchap, 2017/08/30 13:52:29]

can add comment which of many rtts you are taking here, e.g.
RTT(remote_ssrc, nullptr, /*avg_rtt=*/&expected_retransmission_time_ms, nullptr,
...);

[sprang_webrtc, 2017/08/31 15:54:28]

Done.

+                           nullptr) == -1) {
+      expected_retransmission_time_ms = kDefaultExpectedRetransmissionTimeMs;
+    }
+  }
   return rtp_sender_->SendOutgoingData(
       frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
-      payload_size, fragmentation, rtp_video_header, transport_frame_id_out);
+      payload_size, fragmentation, rtp_video_header, transport_frame_id_out,
+      expected_retransmission_time_ms);
 }
 
 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
                                          uint16_t sequence_number,
                                          int64_t capture_time_ms,
                                          bool retransmission,
                                          const PacedPacketInfo& pacing_info) {
   return rtp_sender_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
                                       retransmission, pacing_info);
 }
@@ skipping 465 matching lines @@
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_->GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc