1 /* 2 * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_RTP_RTCP_SOURCE_DEPRECATED_DEPRECATED_RTP_SENDER_EGRESS_H_ 12 #define MODULES_RTP_RTCP_SOURCE_DEPRECATED_DEPRECATED_RTP_SENDER_EGRESS_H_ 13 14 #include <map> 15 #include <memory> 16 #include <vector> 17 18 #include "absl/types/optional.h" 19 #include "api/call/transport.h" 20 #include "api/rtc_event_log/rtc_event_log.h" 21 #include "api/units/data_rate.h" 22 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" 23 #include "modules/rtp_rtcp/source/packet_sequencer.h" 24 #include "modules/rtp_rtcp/source/rtp_packet_history.h" 25 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" 26 #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" 27 #include "modules/rtp_rtcp/source/rtp_sequence_number_map.h" 28 #include "rtc_base/rate_statistics.h" 29 #include "rtc_base/synchronization/mutex.h" 30 #include "rtc_base/thread_annotations.h" 31 32 namespace webrtc { 33 34 class DEPRECATED_RtpSenderEgress { 35 public: 36 // Helper class that redirects packets directly to the send part of this class 37 // without passing through an actual paced sender. 38 class NonPacedPacketSender : public RtpPacketSender { 39 public: 40 NonPacedPacketSender(DEPRECATED_RtpSenderEgress* sender, 41 PacketSequencer* sequence_number_assigner); 42 virtual ~NonPacedPacketSender(); 43 44 void EnqueuePackets( 45 std::vector<std::unique_ptr<RtpPacketToSend>> packets) override; 46 47 private: 48 uint16_t transport_sequence_number_; 49 DEPRECATED_RtpSenderEgress* const sender_; 50 PacketSequencer* sequence_number_assigner_; 51 }; 52 53 DEPRECATED_RtpSenderEgress(const RtpRtcpInterface::Configuration& config, 54 RtpPacketHistory* packet_history); 55 ~DEPRECATED_RtpSenderEgress() = default; 56 57 void SendPacket(RtpPacketToSend* packet, const PacedPacketInfo& pacing_info) 58 RTC_LOCKS_EXCLUDED(lock_); Ssrc()59 uint32_t Ssrc() const { return ssrc_; } RtxSsrc()60 absl::optional<uint32_t> RtxSsrc() const { return rtx_ssrc_; } FlexFecSsrc()61 absl::optional<uint32_t> FlexFecSsrc() const { return flexfec_ssrc_; } 62 63 void ProcessBitrateAndNotifyObservers() RTC_LOCKS_EXCLUDED(lock_); 64 RtpSendRates GetSendRates() const RTC_LOCKS_EXCLUDED(lock_); 65 void GetDataCounters(StreamDataCounters* rtp_stats, 66 StreamDataCounters* rtx_stats) const 67 RTC_LOCKS_EXCLUDED(lock_); 68 69 void ForceIncludeSendPacketsInAllocation(bool part_of_allocation) 70 RTC_LOCKS_EXCLUDED(lock_); 71 bool MediaHasBeenSent() const RTC_LOCKS_EXCLUDED(lock_); 72 void SetMediaHasBeenSent(bool media_sent) RTC_LOCKS_EXCLUDED(lock_); 73 void SetTimestampOffset(uint32_t timestamp) RTC_LOCKS_EXCLUDED(lock_); 74 75 // For each sequence number in `sequence_number`, recall the last RTP packet 76 // which bore it - its timestamp and whether it was the first and/or last 77 // packet in that frame. If all of the given sequence numbers could be 78 // recalled, return a vector with all of them (in corresponding order). 79 // If any could not be recalled, return an empty vector. 80 std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos( 81 rtc::ArrayView<const uint16_t> sequence_numbers) const 82 RTC_LOCKS_EXCLUDED(lock_); 83 84 private: 85 // Maps capture time in milliseconds to send-side delay in milliseconds. 86 // Send-side delay is the difference between transmission time and capture 87 // time. 88 typedef std::map<int64_t, int> SendDelayMap; 89 90 RtpSendRates GetSendRatesLocked() const RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_); 91 bool HasCorrectSsrc(const RtpPacketToSend& packet) const; 92 void AddPacketToTransportFeedback(uint16_t packet_id, 93 const RtpPacketToSend& packet, 94 const PacedPacketInfo& pacing_info); 95 void UpdateDelayStatistics(int64_t capture_time_ms, 96 int64_t now_ms, 97 uint32_t ssrc); 98 void RecomputeMaxSendDelay() RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_); 99 void UpdateOnSendPacket(int packet_id, 100 int64_t capture_time_ms, 101 uint32_t ssrc); 102 // Sends packet on to `transport_`, leaving the RTP module. 103 bool SendPacketToNetwork(const RtpPacketToSend& packet, 104 const PacketOptions& options, 105 const PacedPacketInfo& pacing_info); 106 void UpdateRtpStats(const RtpPacketToSend& packet) 107 RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_); 108 109 const uint32_t ssrc_; 110 const absl::optional<uint32_t> rtx_ssrc_; 111 const absl::optional<uint32_t> flexfec_ssrc_; 112 const bool populate_network2_timestamp_; 113 Clock* const clock_; 114 RtpPacketHistory* const packet_history_; 115 Transport* const transport_; 116 RtcEventLog* const event_log_; 117 const bool is_audio_; 118 const bool need_rtp_packet_infos_; 119 120 TransportFeedbackObserver* const transport_feedback_observer_; 121 SendSideDelayObserver* const send_side_delay_observer_; 122 SendPacketObserver* const send_packet_observer_; 123 StreamDataCountersCallback* const rtp_stats_callback_; 124 BitrateStatisticsObserver* const bitrate_callback_; 125 126 mutable Mutex lock_; 127 bool media_has_been_sent_ RTC_GUARDED_BY(lock_); 128 bool force_part_of_allocation_ RTC_GUARDED_BY(lock_); 129 uint32_t timestamp_offset_ RTC_GUARDED_BY(lock_); 130 131 SendDelayMap send_delays_ RTC_GUARDED_BY(lock_); 132 SendDelayMap::const_iterator max_delay_it_ RTC_GUARDED_BY(lock_); 133 // The sum of delays over a kSendSideDelayWindowMs sliding window. 134 int64_t sum_delays_ms_ RTC_GUARDED_BY(lock_); 135 StreamDataCounters rtp_stats_ RTC_GUARDED_BY(lock_); 136 StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(lock_); 137 // One element per value in RtpPacketMediaType, with index matching value. 138 std::vector<RateStatistics> send_rates_ RTC_GUARDED_BY(lock_); 139 140 // Maps sent packets' sequence numbers to a tuple consisting of: 141 // 1. The timestamp, without the randomizing offset mandated by the RFC. 142 // 2. Whether the packet was the first in its frame. 143 // 3. Whether the packet was the last in its frame. 144 const std::unique_ptr<RtpSequenceNumberMap> rtp_sequence_number_map_ 145 RTC_GUARDED_BY(lock_); 146 }; 147 148 } // namespace webrtc 149 150 #endif // MODULES_RTP_RTCP_SOURCE_DEPRECATED_DEPRECATED_RTP_SENDER_EGRESS_H_ 151