xref: /aosp_15_r20/external/webrtc/modules/rtp_rtcp/source/rtp_sender_egress.h (revision d9f758449e529ab9291ac668be2861e7a55c2422)
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_RTP_SENDER_EGRESS_H_
12 #define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_EGRESS_H_
13 
14 #include <map>
15 #include <memory>
16 #include <utility>
17 #include <vector>
18 
19 #include "absl/types/optional.h"
20 #include "api/call/transport.h"
21 #include "api/rtc_event_log/rtc_event_log.h"
22 #include "api/sequence_checker.h"
23 #include "api/task_queue/pending_task_safety_flag.h"
24 #include "api/task_queue/task_queue_base.h"
25 #include "api/units/data_rate.h"
26 #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
27 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
28 #include "modules/rtp_rtcp/source/packet_sequencer.h"
29 #include "modules/rtp_rtcp/source/rtp_packet_history.h"
30 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
31 #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
32 #include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
33 #include "rtc_base/rate_statistics.h"
34 #include "rtc_base/synchronization/mutex.h"
35 #include "rtc_base/system/no_unique_address.h"
36 #include "rtc_base/task_utils/repeating_task.h"
37 #include "rtc_base/thread_annotations.h"
38 
39 namespace webrtc {
40 
41 class RtpSenderEgress {
42  public:
43   // Helper class that redirects packets directly to the send part of this class
44   // without passing through an actual paced sender.
45   class NonPacedPacketSender : public RtpPacketSender {
46    public:
47     NonPacedPacketSender(RtpSenderEgress* sender, PacketSequencer* sequencer);
48     virtual ~NonPacedPacketSender();
49 
50     void EnqueuePackets(
51         std::vector<std::unique_ptr<RtpPacketToSend>> packets) override;
52 
53    private:
54     void PrepareForSend(RtpPacketToSend* packet);
55     uint16_t transport_sequence_number_;
56     RtpSenderEgress* const sender_;
57     PacketSequencer* sequencer_;
58   };
59 
60   RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
61                   RtpPacketHistory* packet_history);
62   ~RtpSenderEgress();
63 
64   void SendPacket(RtpPacketToSend* packet, const PacedPacketInfo& pacing_info)
65       RTC_LOCKS_EXCLUDED(lock_);
Ssrc()66   uint32_t Ssrc() const { return ssrc_; }
RtxSsrc()67   absl::optional<uint32_t> RtxSsrc() const { return rtx_ssrc_; }
FlexFecSsrc()68   absl::optional<uint32_t> FlexFecSsrc() const { return flexfec_ssrc_; }
69 
70   RtpSendRates GetSendRates() const RTC_LOCKS_EXCLUDED(lock_);
71   void GetDataCounters(StreamDataCounters* rtp_stats,
72                        StreamDataCounters* rtx_stats) const
73       RTC_LOCKS_EXCLUDED(lock_);
74 
75   void ForceIncludeSendPacketsInAllocation(bool part_of_allocation)
76       RTC_LOCKS_EXCLUDED(lock_);
77   bool MediaHasBeenSent() const RTC_LOCKS_EXCLUDED(lock_);
78   void SetMediaHasBeenSent(bool media_sent) RTC_LOCKS_EXCLUDED(lock_);
79   void SetTimestampOffset(uint32_t timestamp) RTC_LOCKS_EXCLUDED(lock_);
80 
81   // For each sequence number in `sequence_number`, recall the last RTP packet
82   // which bore it - its timestamp and whether it was the first and/or last
83   // packet in that frame. If all of the given sequence numbers could be
84   // recalled, return a vector with all of them (in corresponding order).
85   // If any could not be recalled, return an empty vector.
86   std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
87       rtc::ArrayView<const uint16_t> sequence_numbers) const
88       RTC_LOCKS_EXCLUDED(lock_);
89 
90   void SetFecProtectionParameters(const FecProtectionParams& delta_params,
91                                   const FecProtectionParams& key_params);
92   std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets();
93 
94   // Clears pending status for these sequence numbers in the packet history.
95   void OnAbortedRetransmissions(
96       rtc::ArrayView<const uint16_t> sequence_numbers);
97 
98  private:
99   // Maps capture time in milliseconds to send-side delay in milliseconds.
100   // Send-side delay is the difference between transmission time and capture
101   // time.
102   typedef std::map<int64_t, int> SendDelayMap;
103 
104   RtpSendRates GetSendRatesLocked(int64_t now_ms) const
105       RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
106   bool HasCorrectSsrc(const RtpPacketToSend& packet) const;
107   void AddPacketToTransportFeedback(uint16_t packet_id,
108                                     const RtpPacketToSend& packet,
109                                     const PacedPacketInfo& pacing_info);
110   void UpdateDelayStatistics(int64_t capture_time_ms,
111                              int64_t now_ms,
112                              uint32_t ssrc);
113   void RecomputeMaxSendDelay() RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
114   void UpdateOnSendPacket(int packet_id,
115                           int64_t capture_time_ms,
116                           uint32_t ssrc);
117   // Sends packet on to `transport_`, leaving the RTP module.
118   bool SendPacketToNetwork(const RtpPacketToSend& packet,
119                            const PacketOptions& options,
120                            const PacedPacketInfo& pacing_info);
121 
122   void UpdateRtpStats(int64_t now_ms,
123                       uint32_t packet_ssrc,
124                       RtpPacketMediaType packet_type,
125                       RtpPacketCounter counter,
126                       size_t packet_size);
127 #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
128   void BweTestLoggingPlot(int64_t now_ms, uint32_t packet_ssrc);
129 #endif
130 
131   // Called on a timer, once a second, on the worker_queue_.
132   void PeriodicUpdate();
133 
134   TaskQueueBase* const worker_queue_;
135   RTC_NO_UNIQUE_ADDRESS SequenceChecker pacer_checker_;
136   const uint32_t ssrc_;
137   const absl::optional<uint32_t> rtx_ssrc_;
138   const absl::optional<uint32_t> flexfec_ssrc_;
139   const bool populate_network2_timestamp_;
140   Clock* const clock_;
141   RtpPacketHistory* const packet_history_;
142   Transport* const transport_;
143   RtcEventLog* const event_log_;
144 #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
145   const bool is_audio_;
146 #endif
147   const bool need_rtp_packet_infos_;
148   VideoFecGenerator* const fec_generator_ RTC_GUARDED_BY(pacer_checker_);
149   absl::optional<uint16_t> last_sent_seq_ RTC_GUARDED_BY(pacer_checker_);
150   absl::optional<uint16_t> last_sent_rtx_seq_ RTC_GUARDED_BY(pacer_checker_);
151 
152   TransportFeedbackObserver* const transport_feedback_observer_;
153   SendSideDelayObserver* const send_side_delay_observer_;
154   SendPacketObserver* const send_packet_observer_;
155   StreamDataCountersCallback* const rtp_stats_callback_;
156   BitrateStatisticsObserver* const bitrate_callback_;
157 
158   mutable Mutex lock_;
159   bool media_has_been_sent_ RTC_GUARDED_BY(pacer_checker_);
160   bool force_part_of_allocation_ RTC_GUARDED_BY(lock_);
161   uint32_t timestamp_offset_ RTC_GUARDED_BY(worker_queue_);
162 
163   SendDelayMap send_delays_ RTC_GUARDED_BY(lock_);
164   SendDelayMap::const_iterator max_delay_it_ RTC_GUARDED_BY(lock_);
165   // The sum of delays over a kSendSideDelayWindowMs sliding window.
166   int64_t sum_delays_ms_ RTC_GUARDED_BY(lock_);
167   StreamDataCounters rtp_stats_ RTC_GUARDED_BY(lock_);
168   StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(lock_);
169   // One element per value in RtpPacketMediaType, with index matching value.
170   std::vector<RateStatistics> send_rates_ RTC_GUARDED_BY(lock_);
171   absl::optional<std::pair<FecProtectionParams, FecProtectionParams>>
172       pending_fec_params_ RTC_GUARDED_BY(lock_);
173 
174   // Maps sent packets' sequence numbers to a tuple consisting of:
175   // 1. The timestamp, without the randomizing offset mandated by the RFC.
176   // 2. Whether the packet was the first in its frame.
177   // 3. Whether the packet was the last in its frame.
178   const std::unique_ptr<RtpSequenceNumberMap> rtp_sequence_number_map_
179       RTC_GUARDED_BY(worker_queue_);
180   RepeatingTaskHandle update_task_ RTC_GUARDED_BY(worker_queue_);
181   ScopedTaskSafety task_safety_;
182 };
183 
184 }  // namespace webrtc
185 
186 #endif  // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_EGRESS_H_
187