webrtc_m130/modules/rtp_rtcp/source/rtp_sender_egress.h

/*
 *  Copyright (c) 2019 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 MODULES_RTP_RTCP_SOURCE_RTP_SENDER_EGRESS_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_EGRESS_H_

#include <map>
#include <memory>
#include <vector>

#include "absl/types/optional.h"
#include "api/call/transport.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/units/data_rate.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_history.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/rate_statistics.h"
#include "rtc_base/thread_annotations.h"

namespace webrtc {

class RtpSenderEgress {
 public:
  // Helper class that redirects packets directly to the send part of this class
  // without passing through an actual paced sender.
  class NonPacedPacketSender : public RtpPacketSender {
   public:
    explicit NonPacedPacketSender(RtpSenderEgress* sender);
    virtual ~NonPacedPacketSender();

    void EnqueuePackets(
        std::vector<std::unique_ptr<RtpPacketToSend>> packets) override;

   private:
    uint16_t transport_sequence_number_;
    RtpSenderEgress* const sender_;
  };

  RtpSenderEgress(const RtpRtcp::Configuration& config,
                  RtpPacketHistory* packet_history);
  ~RtpSenderEgress() = default;

  void SendPacket(RtpPacketToSend* packet, const PacedPacketInfo& pacing_info);
  uint32_t Ssrc() const { return ssrc_; }
  absl::optional<uint32_t> RtxSsrc() const { return rtx_ssrc_; }
  absl::optional<uint32_t> FlexFecSsrc() const { return flexfec_ssrc_; }

  void ProcessBitrateAndNotifyObservers();
  DataRate SendBitrate() const;
  DataRate NackOverheadRate() const;
  void GetDataCounters(StreamDataCounters* rtp_stats,
                       StreamDataCounters* rtx_stats) const;

  void ForceIncludeSendPacketsInAllocation(bool part_of_allocation);
  bool MediaHasBeenSent() const;
  void SetMediaHasBeenSent(bool media_sent);

 private:
  // Maps capture time in milliseconds to send-side delay in milliseconds.
  // Send-side delay is the difference between transmission time and capture
  // time.
  typedef std::map<int64_t, int> SendDelayMap;

  bool HasCorrectSsrc(const RtpPacketToSend& packet) const;
  void AddPacketToTransportFeedback(uint16_t packet_id,
                                    const RtpPacketToSend& packet,
                                    const PacedPacketInfo& pacing_info);
  void UpdateDelayStatistics(int64_t capture_time_ms,
                             int64_t now_ms,
                             uint32_t ssrc);
  void RecomputeMaxSendDelay() RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
  void UpdateOnSendPacket(int packet_id,
                          int64_t capture_time_ms,
                          uint32_t ssrc);
  // Sends packet on to |transport_|, leaving the RTP module.
  bool SendPacketToNetwork(const RtpPacketToSend& packet,
                           const PacketOptions& options,
                           const PacedPacketInfo& pacing_info);
  void UpdateRtpOverhead(const RtpPacketToSend& packet);
  void UpdateRtpStats(const RtpPacketToSend& packet)
      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);

  const uint32_t ssrc_;
  const absl::optional<uint32_t> rtx_ssrc_;
  const absl::optional<uint32_t> flexfec_ssrc_;
  const bool populate_network2_timestamp_;
  const bool send_side_bwe_with_overhead_;
  Clock* const clock_;
  RtpPacketHistory* const packet_history_;
  Transport* const transport_;
  RtcEventLog* const event_log_;
  const bool is_audio_;

  TransportFeedbackObserver* const transport_feedback_observer_;
  SendSideDelayObserver* const send_side_delay_observer_;
  SendPacketObserver* const send_packet_observer_;
  OverheadObserver* const overhead_observer_;
  StreamDataCountersCallback* const rtp_stats_callback_;
  BitrateStatisticsObserver* const bitrate_callback_;

  rtc::CriticalSection lock_;
  bool media_has_been_sent_ RTC_GUARDED_BY(lock_);
  bool force_part_of_allocation_ RTC_GUARDED_BY(lock_);

  SendDelayMap send_delays_ RTC_GUARDED_BY(lock_);
  SendDelayMap::const_iterator max_delay_it_ RTC_GUARDED_BY(lock_);
  // The sum of delays over a kSendSideDelayWindowMs sliding window.
  int64_t sum_delays_ms_ RTC_GUARDED_BY(lock_);
  uint64_t total_packet_send_delay_ms_ RTC_GUARDED_BY(lock_);
  size_t rtp_overhead_bytes_per_packet_ RTC_GUARDED_BY(lock_);
  StreamDataCounters rtp_stats_ RTC_GUARDED_BY(lock_);
  StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(lock_);
  RateStatistics total_bitrate_sent_ RTC_GUARDED_BY(lock_);
  RateStatistics nack_bitrate_sent_ RTC_GUARDED_BY(lock_);
};

}  // namespace webrtc

#endif  // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_EGRESS_H_