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 <stdint.h> 15 16 #include <vector> 17 18 #include "absl/types/optional.h" 19 #include "api/units/data_rate.h" 20 #include "api/units/data_size.h" 21 #include "api/units/time_delta.h" 22 #include "api/units/timestamp.h" 23 #include "modules/rtp_rtcp/include/rtp_packet_sender.h" 24 25 namespace webrtc { 26 27 class RtpPacketPacer { 28 public: 29 virtual ~RtpPacketPacer() = default; 30 31 virtual void CreateProbeClusters( 32 std::vector<ProbeClusterConfig> probe_cluster_configs) = 0; 33 34 // Temporarily pause all sending. 35 virtual void Pause() = 0; 36 37 // Resume sending packets. 38 virtual void Resume() = 0; 39 40 virtual void SetCongested(bool congested) = 0; 41 42 // Sets the pacing rates. Must be called once before packets can be sent. 43 virtual void SetPacingRates(DataRate pacing_rate, DataRate padding_rate) = 0; 44 45 // Time since the oldest packet currently in the queue was added. 46 virtual TimeDelta OldestPacketWaitTime() const = 0; 47 48 // Sum of payload + padding bytes of all packets currently in the pacer queue. 49 virtual DataSize QueueSizeData() const = 0; 50 51 // Returns the time when the first packet was sent. 52 virtual absl::optional<Timestamp> FirstSentPacketTime() const = 0; 53 54 // Returns the expected number of milliseconds it will take to send the 55 // current packets in the queue, given the current size and bitrate, ignoring 56 // priority. 57 virtual TimeDelta ExpectedQueueTime() const = 0; 58 59 // Set the average upper bound on pacer queuing delay. The pacer may send at 60 // a higher rate than what was configured via SetPacingRates() in order to 61 // keep ExpectedQueueTimeMs() below `limit_ms` on average. 62 virtual void SetQueueTimeLimit(TimeDelta limit) = 0; 63 64 // Currently audio traffic is not accounted by pacer and passed through. 65 // With the introduction of audio BWE audio traffic will be accounted for 66 // the pacer budget calculation. The audio traffic still will be injected 67 // at high priority. 68 virtual void SetAccountForAudioPackets(bool account_for_audio) = 0; 69 virtual void SetIncludeOverhead() = 0; 70 virtual void SetTransportOverhead(DataSize overhead_per_packet) = 0; 71 }; 72 73 } // namespace webrtc 74 #endif // MODULES_PACING_RTP_PACKET_PACER_H_ 75