Make FlexFEC packets paceable through 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 23 matching lines @@
 
 namespace {
 // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
 constexpr size_t kMaxPaddingLength = 224;
 constexpr int kSendSideDelayWindowMs = 1000;
 constexpr size_t kRtpHeaderLength = 12;
 constexpr uint16_t kMaxInitRtpSeqNumber = 32767;  // 2^15 -1.
 constexpr uint32_t kTimestampTicksPerMs = 90;
 constexpr int kBitrateStatisticsWindowMs = 1000;
 
+constexpr size_t kMinFlexfecPacketsToStoreForPacing = 50;
+
 const char* FrameTypeToString(FrameType frame_type) {
   switch (frame_type) {
     case kEmptyFrame:
       return "empty";
     case kAudioFrameSpeech: return "audio_speech";
     case kAudioFrameCN: return "audio_cn";
     case kVideoFrameKey: return "video_key";
     case kVideoFrameDelta: return "video_delta";
   }
   return "";
@@ skipping 33 matching lines @@
       transport_sequence_number_allocator_(sequence_number_allocator),
       transport_feedback_observer_(transport_feedback_observer),
       last_capture_time_ms_sent_(0),
       transport_(transport),
       sending_media_(true),                      // Default to sending media.
       max_payload_length_(IP_PACKET_SIZE - 28),  // Default is IP-v4/UDP.
       payload_type_(-1),
       payload_type_map_(),
       rtp_header_extension_map_(),
       packet_history_(clock),
+      flexfec_packet_history_(clock),
       // Statistics
       rtp_stats_callback_(nullptr),
       total_bitrate_sent_(kBitrateStatisticsWindowMs,
                           RateStatistics::kBpsScale),
       nack_bitrate_sent_(kBitrateStatisticsWindowMs, RateStatistics::kBpsScale),
       frame_count_observer_(frame_count_observer),
       send_side_delay_observer_(send_side_delay_observer),
       event_log_(event_log),
       send_packet_observer_(send_packet_observer),
       bitrate_callback_(bitrate_callback),
@@ skipping 13 matching lines @@
   ssrc_ = ssrc_db_->CreateSSRC();
   RTC_DCHECK(ssrc_ != 0);
   ssrc_rtx_ = ssrc_db_->CreateSSRC();
   RTC_DCHECK(ssrc_rtx_ != 0);
 
   // This random initialization is not intended to be cryptographic strong.
   timestamp_offset_ = random_.Rand<uint32_t>();
   // Random start, 16 bits. Can't be 0.
   sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
   sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
+
+  // Store FlexFEC packets in the packet history data structure, so they can
+  // be found when paced.
+  if (flexfec_sender) {
+    flexfec_packet_history_.SetStorePacketsStatus(
+        true, kMinFlexfecPacketsToStoreForPacing);
+  }
 }
 
 RTPSender::~RTPSender() {
   // TODO(tommi): Use a thread checker to ensure the object is created and
   // deleted on the same thread.  At the moment this isn't possible due to
   // voe::ChannelOwner in voice engine.  To reproduce, run:
   // voe_auto_test --automated --gtest_filter=*MixManyChannelsForStressOpus
 
   // TODO(tommi,holmer): We don't grab locks in the dtor before accessing member
   // variables but we grab them in all other methods. (what's the design?)
@@ skipping 538 matching lines @@
     }
   }
 }
 
 void RTPSender::OnReceivedRtcpReportBlocks(
     const ReportBlockList& report_blocks) {
   playout_delay_oracle_.OnReceivedRtcpReportBlocks(report_blocks);
 }
 
 // Called from pacer when we can send the packet.
-bool RTPSender::TimeToSendPacket(uint16_t sequence_number,
+bool RTPSender::TimeToSendPacket(uint32_t ssrc,
+                                 uint16_t sequence_number,
                                  int64_t capture_time_ms,
                                  bool retransmission,
                                  int probe_cluster_id) {
-  std::unique_ptr<RtpPacketToSend> packet =
-      packet_history_.GetPacketAndSetSendTime(sequence_number, 0,
-                                              retransmission);
+  if (!SendingMedia())
+    return true;
+
+  std::unique_ptr<RtpPacketToSend> packet;
+  if (ssrc == SSRC()) {
+    packet = packet_history_.GetPacketAndSetSendTime(sequence_number, 0,
+                                                     retransmission);
+  } else if (ssrc == FlexfecSsrc()) {
+    packet = flexfec_packet_history_.GetPacketAndSetSendTime(sequence_number, 0,
+                                                             retransmission);
+  }
+
   if (!packet) {
-    // Packet cannot be found. Allow sending to continue.
+    // Packet cannot be found.
     return true;
   }
 
   return PrepareAndSendPacket(
       std::move(packet),
       retransmission && (RtxStatus() & kRtxRetransmitted) > 0, retransmission,
       probe_cluster_id);
 }
 
 bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
@@ skipping 122 matching lines @@
                                     FecOverheadRate() / 1000, packet->Ssrc());
     BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoNackBitrate_kbps", now_ms,
                                     NackOverheadRate() / 1000, packet->Ssrc());
   } else {
     BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
                                     ActualSendBitrateKbit(), packet->Ssrc());
     BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioNackBitrate_kbps", now_ms,
                                     NackOverheadRate() / 1000, packet->Ssrc());
   }
 
+  uint32_t ssrc = packet->Ssrc();
+  rtc::Optional<uint32_t> flexfec_ssrc = FlexfecSsrc();
   if (paced_sender_) {
     uint16_t seq_no = packet->SequenceNumber();
-    uint32_t ssrc = packet->Ssrc();
     // Correct offset between implementations of millisecond time stamps in
     // TickTime and Clock.
     int64_t corrected_time_ms = packet->capture_time_ms() + clock_delta_ms_;
     size_t payload_length = packet->payload_size();
-    packet_history_.PutRtpPacket(std::move(packet), storage, false);
+    if (ssrc == flexfec_ssrc) {
+      // Store FlexFEC packets in the history here, so they can be found
+      // when the pacer calls TimeToSendPacket.
+      flexfec_packet_history_.PutRtpPacket(std::move(packet), storage, false);
+    } else {
+      packet_history_.PutRtpPacket(std::move(packet), storage, false);
+    }
 
     paced_sender_->InsertPacket(priority, ssrc, seq_no, corrected_time_ms,
                                 payload_length, false);
     if (last_capture_time_ms_sent_ == 0 ||
         corrected_time_ms > last_capture_time_ms_sent_) {
       last_capture_time_ms_sent_ = corrected_time_ms;
       TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                                "PacedSend", corrected_time_ms,
                                "capture_time_ms", corrected_time_ms);
     }
@@ skipping 15 matching lines @@
   bool sent = SendPacketToNetwork(*packet, options);
 
   if (sent) {
     {
       rtc::CritScope lock(&send_critsect_);
       media_has_been_sent_ = true;
     }
     UpdateRtpStats(*packet, false, false);
   }
 
-  // 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.
-  packet_history_.PutRtpPacket(std::move(packet), storage, true);
+  // To support retransmissions, we store the media packet as sent in the
+  // packet history (even if send failed).
+  if (storage == kAllowRetransmission) {

[brandtr, 2016/11/14 12:36:20]

I thought it was nice to check the complement (i.e. == kAllowRetransmission, not
!= kDontRetransmit) here.

+    RTC_DCHECK_EQ(ssrc, SSRC());
+    packet_history_.PutRtpPacket(std::move(packet), storage, true);
+  }
 
   return sent;
 }
 
 void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
   if (!send_side_delay_observer_ || capture_time_ms <= 0)
     return;
 
   uint32_t ssrc;
   int avg_delay_ms = 0;
@@ skipping 186 matching lines @@
   if (!sequence_number_forced_) {
     sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
   }
 }
 
 uint32_t RTPSender::SSRC() const {
   rtc::CritScope lock(&send_critsect_);
   return ssrc_;
 }
 
+rtc::Optional<uint32_t> RTPSender::FlexfecSsrc() const {
+  if (video_) {
+    return video_->FlexfecSsrc();
+  }
+  return rtc::Optional<uint32_t>();
+}
+
 void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
   assert(csrcs.size() <= kRtpCsrcSize);
   rtc::CritScope lock(&send_critsect_);
   csrcs_ = csrcs;
 }
 
 void RTPSender::SetSequenceNumber(uint16_t seq) {
   rtc::CritScope lock(&send_critsect_);
   sequence_number_forced_ = true;
   sequence_number_ = seq;
@@ skipping 132 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 }  // namespace webrtc