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TBR=stefan@webrtc.org,alexnarest@webrtc.org Bug: webrtc:8243 Change-Id: Ie68e4f414b2ac32ba4e64877cb250fabcb089a07 Reviewed-on: https://webrtc-review.googlesource.com/13940 Commit-Queue: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Alex Narest <alexnarest@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20369}
93 lines
3 KiB
C++
93 lines
3 KiB
C++
/*
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* Copyright (c) 2017 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_REMOTE_BITRATE_ESTIMATOR_TEST_BBR_PACED_SENDER_H_
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#define MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BBR_PACED_SENDER_H_
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#include <list>
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#include <memory>
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#include "modules/pacing/paced_sender.h"
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#include "modules/pacing/pacer.h"
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namespace webrtc {
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namespace testing {
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namespace bwe {
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class CongestionWindow;
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}
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} // namespace testing
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struct Packet {
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Packet(RtpPacketSender::Priority priority,
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uint32_t ssrc,
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uint16_t seq_number,
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int64_t capture_time_ms,
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int64_t enqueue_time_ms,
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size_t size_in_bytes,
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bool retransmission)
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: priority(priority),
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ssrc(ssrc),
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sequence_number(seq_number),
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capture_time_ms(capture_time_ms),
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enqueue_time_ms(enqueue_time_ms),
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size_in_bytes(size_in_bytes),
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retransmission(retransmission) {}
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RtpPacketSender::Priority priority;
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uint32_t ssrc;
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uint16_t sequence_number;
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int64_t capture_time_ms;
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int64_t enqueue_time_ms;
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size_t size_in_bytes;
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bool retransmission;
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};
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class Clock;
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class RtcEventLog;
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struct Packet;
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class BbrPacedSender : public Pacer {
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public:
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BbrPacedSender(const Clock* clock,
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PacedSender::PacketSender* packet_sender,
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RtcEventLog* event_log);
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~BbrPacedSender() override;
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void SetEstimatedBitrateAndCongestionWindow(
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uint32_t bitrate_bps,
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bool in_probe_rtt,
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uint64_t congestion_window) override;
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void SetMinBitrate(int min_send_bitrate_bps);
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void InsertPacket(RtpPacketSender::Priority priority,
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uint32_t ssrc,
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uint16_t sequence_number,
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int64_t capture_time_ms,
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size_t bytes,
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bool retransmission) override;
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void SetAccountForAudioPackets(bool account_for_audio) override {}
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int64_t TimeUntilNextProcess() override;
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void OnBytesAcked(size_t bytes) override;
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void Process() override;
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bool TryToSendPacket(Packet* packet);
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private:
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const Clock* const clock_;
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PacedSender::PacketSender* const packet_sender_;
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uint32_t estimated_bitrate_bps_;
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uint32_t min_send_bitrate_kbps_;
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uint32_t pacing_bitrate_kbps_;
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int64_t time_last_update_us_;
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int64_t time_last_update_ms_;
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int64_t next_packet_send_time_;
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float rounding_error_time_ms_;
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std::list<Packet*> packets_;
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// TODO(gnish): integrate |max_data_inflight| into congestion window class.
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size_t max_data_inflight_bytes_;
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std::unique_ptr<testing::bwe::CongestionWindow> congestion_window_;
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};
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} // namespace webrtc
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#endif // MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BBR_PACED_SENDER_H_
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