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- /*
- * 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 <stdint.h>
- #include "absl/types/optional.h"
- #include "api/units/data_rate.h"
- #include "api/units/data_size.h"
- #include "api/units/time_delta.h"
- #include "api/units/timestamp.h"
- #include "modules/rtp_rtcp/include/rtp_packet_sender.h"
- namespace webrtc {
- class RtpPacketPacer {
- public:
- virtual ~RtpPacketPacer() = default;
- virtual void CreateProbeCluster(DataRate bitrate, int cluster_id) = 0;
- // Temporarily pause all sending.
- virtual void Pause() = 0;
- // Resume sending packets.
- virtual void Resume() = 0;
- virtual void SetCongestionWindow(DataSize congestion_window_size) = 0;
- virtual void UpdateOutstandingData(DataSize outstanding_data) = 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 absl::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_
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