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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 <vector>
#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 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 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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