rtp_packet_pacer.h (2614B)
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_PACING_RTP_PACKET_PACER_H_ 12 #define MODULES_PACING_RTP_PACKET_PACER_H_ 13 14 #include <optional> 15 #include <vector> 16 17 #include "api/transport/network_types.h" 18 #include "api/units/data_rate.h" 19 #include "api/units/data_size.h" 20 #include "api/units/time_delta.h" 21 #include "api/units/timestamp.h" 22 23 namespace webrtc { 24 25 class RtpPacketPacer { 26 public: 27 virtual ~RtpPacketPacer() = default; 28 29 virtual void CreateProbeClusters( 30 std::vector<ProbeClusterConfig> probe_cluster_configs) = 0; 31 32 // Temporarily pause all sending. 33 virtual void Pause() = 0; 34 35 // Resume sending packets. 36 virtual void Resume() = 0; 37 38 virtual void SetCongested(bool congested) = 0; 39 40 // Sets the pacing rates. Must be called once before packets can be sent. 41 virtual void SetPacingRates(DataRate pacing_rate, DataRate padding_rate) = 0; 42 43 // Time since the oldest packet currently in the queue was added. 44 virtual TimeDelta OldestPacketWaitTime() const = 0; 45 46 // Sum of payload + padding bytes of all packets currently in the pacer queue. 47 virtual DataSize QueueSizeData() const = 0; 48 49 // Returns the time when the first packet was sent. 50 virtual std::optional<Timestamp> FirstSentPacketTime() const = 0; 51 52 // Returns the expected number of milliseconds it will take to send the 53 // current packets in the queue, given the current size and bitrate, ignoring 54 // priority. 55 virtual TimeDelta ExpectedQueueTime() const = 0; 56 57 // Set the average upper bound on pacer queuing delay. The pacer may send at 58 // a higher rate than what was configured via SetPacingRates() in order to 59 // keep ExpectedQueueTimeMs() below `limit_ms` on average. 60 virtual void SetQueueTimeLimit(TimeDelta limit) = 0; 61 62 // Currently audio traffic is not accounted by pacer and passed through. 63 // With the introduction of audio BWE audio traffic will be accounted for 64 // the pacer budget calculation. The audio traffic still will be injected 65 // at high priority. 66 virtual void SetAccountForAudioPackets(bool account_for_audio) = 0; 67 virtual void SetIncludeOverhead() = 0; 68 virtual void SetTransportOverhead(DataSize overhead_per_packet) = 0; 69 }; 70 71 } // namespace webrtc 72 #endif // MODULES_PACING_RTP_PACKET_PACER_H_