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88af20356f
Instead of using field trials in BitrateProber for probe duration, use values provided in ProbeClusterConfig from GoogCC. Field trials are instead read in ProbeController. To avoid having to do a thread jump for every ProbeClusterConfig, RtpPacketPacer interface is changed to RtpPacketPacer::CreateProbeClusters(std::vector<ProbeClusterConfig> Deprecates field trial "WebRTC-Bwe-ProbingConfiguration" Change-Id: I3991e4b54770601855a3af2d6a16678f11d41c31 Bug: webrtc:14027 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261265 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36911}
74 lines
2.6 KiB
C++
74 lines
2.6 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_PACING_RTP_PACKET_PACER_H_
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#define MODULES_PACING_RTP_PACKET_PACER_H_
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#include <stdint.h>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/units/data_rate.h"
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#include "api/units/data_size.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
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namespace webrtc {
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class RtpPacketPacer {
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public:
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virtual ~RtpPacketPacer() = default;
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virtual void CreateProbeClusters(
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std::vector<ProbeClusterConfig> probe_cluster_configs) = 0;
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// Temporarily pause all sending.
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virtual void Pause() = 0;
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// Resume sending packets.
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virtual void Resume() = 0;
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virtual void SetCongested(bool congested) = 0;
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// Sets the pacing rates. Must be called once before packets can be sent.
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virtual void SetPacingRates(DataRate pacing_rate, DataRate padding_rate) = 0;
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// Time since the oldest packet currently in the queue was added.
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virtual TimeDelta OldestPacketWaitTime() const = 0;
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// Sum of payload + padding bytes of all packets currently in the pacer queue.
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virtual DataSize QueueSizeData() const = 0;
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// Returns the time when the first packet was sent.
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virtual absl::optional<Timestamp> FirstSentPacketTime() const = 0;
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// Returns the expected number of milliseconds it will take to send the
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// current packets in the queue, given the current size and bitrate, ignoring
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// priority.
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virtual TimeDelta ExpectedQueueTime() const = 0;
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// Set the average upper bound on pacer queuing delay. The pacer may send at
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// a higher rate than what was configured via SetPacingRates() in order to
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// keep ExpectedQueueTimeMs() below `limit_ms` on average.
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virtual void SetQueueTimeLimit(TimeDelta limit) = 0;
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// Currently audio traffic is not accounted by pacer and passed through.
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// With the introduction of audio BWE audio traffic will be accounted for
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// the pacer budget calculation. The audio traffic still will be injected
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// at high priority.
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virtual void SetAccountForAudioPackets(bool account_for_audio) = 0;
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virtual void SetIncludeOverhead() = 0;
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virtual void SetTransportOverhead(DataSize overhead_per_packet) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_PACING_RTP_PACKET_PACER_H_
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