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67ac9e8ecb
The post-pacing part of the RTP sender has been moved from RTPSender into the new RtpSenderEgress class. However, that class is not directly used and instead a subset of method calls are passed through RTPSender. This CL prepares for removing dependencies between RTPSender and RtpSenderEgress. All current behavior is preserved, and unit tests are unchanged to verify this. For more context, see patch set 2. Change-Id: If795f2603aeb6302ac1565d9efaea514af240dc7 Bug: webrtc:11036 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158020 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29616}
129 lines
5 KiB
C++
129 lines
5 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_RTP_RTCP_SOURCE_RTP_SENDER_EGRESS_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_EGRESS_H_
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#include <map>
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#include <memory>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/call/transport.h"
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#include "api/rtc_event_log/rtc_event_log.h"
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#include "api/units/data_rate.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_packet_history.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "rtc_base/critical_section.h"
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#include "rtc_base/rate_statistics.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class RtpSenderEgress {
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public:
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// Helper class that redirects packets directly to the send part of this class
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// without passing through an actual paced sender.
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class NonPacedPacketSender : public RtpPacketSender {
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public:
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explicit NonPacedPacketSender(RtpSenderEgress* sender);
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virtual ~NonPacedPacketSender();
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void EnqueuePackets(
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std::vector<std::unique_ptr<RtpPacketToSend>> packets) override;
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private:
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uint16_t transport_sequence_number_;
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RtpSenderEgress* const sender_;
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};
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RtpSenderEgress(const RtpRtcp::Configuration& config,
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RtpPacketHistory* packet_history);
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~RtpSenderEgress() = default;
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void SendPacket(RtpPacketToSend* packet, const PacedPacketInfo& pacing_info);
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uint32_t Ssrc() const { return ssrc_; }
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absl::optional<uint32_t> RtxSsrc() const { return rtx_ssrc_; }
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absl::optional<uint32_t> FlexFecSsrc() const { return flexfec_ssrc_; }
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void ProcessBitrateAndNotifyObservers();
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DataRate SendBitrate() const;
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DataRate NackOverheadRate() const;
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void GetDataCounters(StreamDataCounters* rtp_stats,
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StreamDataCounters* rtx_stats) const;
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void ForceIncludeSendPacketsInAllocation(bool part_of_allocation);
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bool MediaHasBeenSent() const;
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void SetMediaHasBeenSent(bool media_sent);
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private:
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// Maps capture time in milliseconds to send-side delay in milliseconds.
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// Send-side delay is the difference between transmission time and capture
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// time.
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typedef std::map<int64_t, int> SendDelayMap;
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bool HasCorrectSsrc(const RtpPacketToSend& packet) const;
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void AddPacketToTransportFeedback(uint16_t packet_id,
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const RtpPacketToSend& packet,
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const PacedPacketInfo& pacing_info);
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void UpdateDelayStatistics(int64_t capture_time_ms,
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int64_t now_ms,
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uint32_t ssrc);
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void RecomputeMaxSendDelay() RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
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void UpdateOnSendPacket(int packet_id,
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int64_t capture_time_ms,
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uint32_t ssrc);
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// Sends packet on to |transport_|, leaving the RTP module.
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bool SendPacketToNetwork(const RtpPacketToSend& packet,
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const PacketOptions& options,
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const PacedPacketInfo& pacing_info);
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void UpdateRtpOverhead(const RtpPacketToSend& packet);
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void UpdateRtpStats(const RtpPacketToSend& packet)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
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const uint32_t ssrc_;
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const absl::optional<uint32_t> rtx_ssrc_;
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const absl::optional<uint32_t> flexfec_ssrc_;
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const bool populate_network2_timestamp_;
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const bool send_side_bwe_with_overhead_;
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Clock* const clock_;
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RtpPacketHistory* const packet_history_;
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Transport* const transport_;
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RtcEventLog* const event_log_;
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const bool is_audio_;
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TransportFeedbackObserver* const transport_feedback_observer_;
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SendSideDelayObserver* const send_side_delay_observer_;
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SendPacketObserver* const send_packet_observer_;
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OverheadObserver* const overhead_observer_;
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StreamDataCountersCallback* const rtp_stats_callback_;
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BitrateStatisticsObserver* const bitrate_callback_;
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rtc::CriticalSection lock_;
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bool media_has_been_sent_ RTC_GUARDED_BY(lock_);
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bool force_part_of_allocation_ RTC_GUARDED_BY(lock_);
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SendDelayMap send_delays_ RTC_GUARDED_BY(lock_);
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SendDelayMap::const_iterator max_delay_it_ RTC_GUARDED_BY(lock_);
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// The sum of delays over a kSendSideDelayWindowMs sliding window.
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int64_t sum_delays_ms_ RTC_GUARDED_BY(lock_);
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uint64_t total_packet_send_delay_ms_ RTC_GUARDED_BY(lock_);
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size_t rtp_overhead_bytes_per_packet_ RTC_GUARDED_BY(lock_);
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StreamDataCounters rtp_stats_ RTC_GUARDED_BY(lock_);
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StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(lock_);
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RateStatistics total_bitrate_sent_ RTC_GUARDED_BY(lock_);
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RateStatistics nack_bitrate_sent_ RTC_GUARDED_BY(lock_);
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_EGRESS_H_
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