| /* |
| * 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_PACING_RTP_PACKET_PACER_H_ |
| #define MODULES_PACING_RTP_PACKET_PACER_H_ |
| |
| |
| #include <optional> |
| #include <vector> |
| |
| #include "api/transport/network_types.h" |
| #include "api/units/data_rate.h" |
| #include "api/units/data_size.h" |
| #include "api/units/time_delta.h" |
| #include "api/units/timestamp.h" |
| |
| namespace webrtc { |
| |
| class RtpPacketPacer { |
| public: |
| virtual ~RtpPacketPacer() = default; |
| |
| virtual void CreateProbeClusters( |
| std::vector<ProbeClusterConfig> probe_cluster_configs) = 0; |
| |
| // Temporarily pause all sending. |
| virtual void Pause() = 0; |
| |
| // Resume sending packets. |
| virtual void Resume() = 0; |
| |
| virtual void SetCongested(bool congested) = 0; |
| |
| // Sets the pacing rates. Must be called once before packets can be sent. |
| virtual void SetPacingRates(DataRate pacing_rate, DataRate padding_rate) = 0; |
| |
| // Time since the oldest packet currently in the queue was added. |
| virtual TimeDelta OldestPacketWaitTime() const = 0; |
| |
| // Sum of payload + padding bytes of all packets currently in the pacer queue. |
| virtual DataSize QueueSizeData() const = 0; |
| |
| // Returns the time when the first packet was sent. |
| virtual std::optional<Timestamp> FirstSentPacketTime() const = 0; |
| |
| // Returns the expected number of milliseconds it will take to send the |
| // current packets in the queue, given the current size and bitrate, ignoring |
| // priority. |
| virtual TimeDelta ExpectedQueueTime() const = 0; |
| |
| // Set the average upper bound on pacer queuing delay. The pacer may send at |
| // a higher rate than what was configured via SetPacingRates() in order to |
| // keep ExpectedQueueTimeMs() below `limit_ms` on average. |
| virtual void SetQueueTimeLimit(TimeDelta limit) = 0; |
| |
| // Currently audio traffic is not accounted by pacer and passed through. |
| // With the introduction of audio BWE audio traffic will be accounted for |
| // the pacer budget calculation. The audio traffic still will be injected |
| // at high priority. |
| virtual void SetAccountForAudioPackets(bool account_for_audio) = 0; |
| virtual void SetIncludeOverhead() = 0; |
| virtual void SetTransportOverhead(DataSize overhead_per_packet) = 0; |
| }; |
| |
| } // namespace webrtc |
| #endif // MODULES_PACING_RTP_PACKET_PACER_H_ |