Make FlexFEC packets paceable through RTPSender.

webrtc/modules/rtp_rtcp/source/rtp_sender.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
 
 #include <map>
 #include <memory>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/deprecation.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/base/random.h"
 #include "webrtc/base/rate_statistics.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
@@ skipping 83 matching lines @@
                         const RTPVideoHeader* rtp_header,
                         uint32_t* transport_frame_id_out);
 
   // RTP header extension
   int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
   bool IsRtpHeaderExtensionRegistered(RTPExtensionType type);
   int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
 
   size_t RtpHeaderExtensionLength() const;
 
-  bool TimeToSendPacket(uint16_t sequence_number,
+  bool TimeToSendPacket(uint32_t ssrc,
+                        uint16_t sequence_number,
                         int64_t capture_time_ms,
                         bool retransmission,
                         int probe_cluster_id);
   size_t TimeToSendPadding(size_t bytes, int probe_cluster_id);
 
   // NACK.
   int SelectiveRetransmissions() const;
   int SetSelectiveRetransmissions(uint8_t settings);
   void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers,
                       int64_t avg_rtt);
@@ skipping 23 matching lines @@
   // Save packet's fields to generate padding that doesn't break media stream.
   // Return false if sending was turned off.
   bool AssignSequenceNumber(RtpPacketToSend* packet);
 
   size_t RtpHeaderLength() const;
   uint16_t AllocateSequenceNumber(uint16_t packets_to_send);
   size_t MaxPayloadLength() const;
 
   uint32_t SSRC() const;
 
+  rtc::Optional<uint32_t> FlexfecSsrc() const;
+
   bool SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
                      StorageType storage,
                      RtpPacketSender::Priority priority);
 
   // Audio.
 
   // Send a DTMF tone using RFC 2833 (4733).
   int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
   // Set audio packet size, used to determine when it's time to send a DTMF
@@ skipping 99 matching lines @@
   std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
 
   RtpHeaderExtensionMap rtp_header_extension_map_ GUARDED_BY(send_critsect_);
 
   // Tracks the current request for playout delay limits from application
   // and decides whether the current RTP frame should include the playout
   // delay extension on header.
   PlayoutDelayOracle playout_delay_oracle_;
 
   RtpPacketHistory packet_history_;
+  // TODO(brandtr): Remove |flexfec_packet_history_| when the FlexfecSender
+  // is hooked up to the PacedSender.
+  RtpPacketHistory flexfec_packet_history_;
 
   // Statistics
   rtc::CriticalSection statistics_crit_;
   SendDelayMap send_delays_ GUARDED_BY(statistics_crit_);
   FrameCounts frame_counts_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
   RateStatistics total_bitrate_sent_ GUARDED_BY(statistics_crit_);
   RateStatistics nack_bitrate_sent_ GUARDED_BY(statistics_crit_);
@@ skipping 24 matching lines @@
   std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_);
 
   RateLimiter* const retransmission_rate_limiter_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_