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This reverts commit66147e892d
. Reason for revert: The culprit was https://webrtc-review.googlesource.com/c/src/+/133169. Original change's description: > Revert "Optimize PacketRouter/RTPSender interactions." > > This reverts commit6f129b3b76
. > > Reason for revert: Speculative revert (some perf test are failing) > > Original change's description: > > Optimize PacketRouter/RTPSender interactions. > > > > The legacy code-path uses a hashmap as cache in order to speed up > > finding the right rtp module to send on. The new path should use that > > as well. > > In addition, there are checks that verify if an RTP module can send > > padding, in some cases payload based. These result in a number of > > calls to methods in RTPSender requiring its lock to be taken. This CL > > introduces a combined SupportsPadding() check method which performs > > all those checks in one go. > > > > Bug: None > > Change-Id: I2d18d0d6e7d8cfe92c81d08cef248a4daa7dcd4b > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144780 > > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Commit-Queue: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#28535} > > TBR=asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org > > Change-Id: I8499dc0fd6e6d0b9fa7a0886c8754655e5589780 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: None > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145326 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28552} TBR=mbonadei@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org Change-Id: I3bff3ecb2b776e30f77c1884f6faa72b21788017 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: None Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145401 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28563}
146 lines
5.6 KiB
C++
146 lines
5.6 KiB
C++
/*
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* Copyright (c) 2015 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_PACKET_ROUTER_H_
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#define MODULES_PACING_PACKET_ROUTER_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <list>
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#include <memory>
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#include <unordered_map>
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#include <vector>
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#include "api/transport/network_types.h"
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#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.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_to_send.h"
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#include "rtc_base/constructor_magic.h"
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#include "rtc_base/critical_section.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class RtpRtcp;
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namespace rtcp {
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class TransportFeedback;
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} // namespace rtcp
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// PacketRouter keeps track of rtp send modules to support the pacer.
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// In addition, it handles feedback messages, which are sent on a send
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// module if possible (sender report), otherwise on receive module
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// (receiver report). For the latter case, we also keep track of the
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// receive modules.
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class PacketRouter : public TransportSequenceNumberAllocator,
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public RemoteBitrateObserver,
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public TransportFeedbackSenderInterface {
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public:
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PacketRouter();
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~PacketRouter() override;
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void AddSendRtpModule(RtpRtcp* rtp_module, bool remb_candidate);
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void RemoveSendRtpModule(RtpRtcp* rtp_module);
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void AddReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender,
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bool remb_candidate);
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void RemoveReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender);
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virtual RtpPacketSendResult TimeToSendPacket(
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uint32_t ssrc,
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uint16_t sequence_number,
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int64_t capture_timestamp,
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bool retransmission,
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const PacedPacketInfo& packet_info);
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virtual void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
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const PacedPacketInfo& cluster_info);
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virtual size_t TimeToSendPadding(size_t bytes,
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const PacedPacketInfo& packet_info);
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virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
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size_t target_size_bytes);
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void SetTransportWideSequenceNumber(uint16_t sequence_number);
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uint16_t AllocateSequenceNumber() override;
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// Called every time there is a new bitrate estimate for a receive channel
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// group. This call will trigger a new RTCP REMB packet if the bitrate
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// estimate has decreased or if no RTCP REMB packet has been sent for
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// a certain time interval.
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// Implements RtpReceiveBitrateUpdate.
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void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
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uint32_t bitrate_bps) override;
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// Ensures remote party notified of the receive bitrate limit no larger than
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// |bitrate_bps|.
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void SetMaxDesiredReceiveBitrate(int64_t bitrate_bps);
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// Send REMB feedback.
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bool SendRemb(int64_t bitrate_bps, const std::vector<uint32_t>& ssrcs);
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// Send transport feedback packet to send-side.
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bool SendTransportFeedback(rtcp::TransportFeedback* packet) override;
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private:
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RtpRtcp* FindRtpModule(uint32_t ssrc)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
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void AddRembModuleCandidate(RtcpFeedbackSenderInterface* candidate_module,
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bool media_sender)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
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void MaybeRemoveRembModuleCandidate(
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RtcpFeedbackSenderInterface* candidate_module,
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bool media_sender) RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
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void UnsetActiveRembModule() RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
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void DetermineActiveRembModule() RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
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bool TrySendPacket(RtpPacketToSend* packet,
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const PacedPacketInfo& cluster_info,
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RtpRtcp* rtp_module)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
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rtc::CriticalSection modules_crit_;
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// Rtp and Rtcp modules of the rtp senders.
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std::list<RtpRtcp*> rtp_send_modules_ RTC_GUARDED_BY(modules_crit_);
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// Ssrc to RtpRtcp module cache.
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std::unordered_map<uint32_t, RtpRtcp*> rtp_module_cache_map_
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RTC_GUARDED_BY(modules_crit_);
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// The last module used to send media.
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RtpRtcp* last_send_module_ RTC_GUARDED_BY(modules_crit_);
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// Rtcp modules of the rtp receivers.
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std::vector<RtcpFeedbackSenderInterface*> rtcp_feedback_senders_
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RTC_GUARDED_BY(modules_crit_);
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// TODO(eladalon): remb_crit_ only ever held from one function, and it's not
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// clear if that function can actually be called from more than one thread.
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rtc::CriticalSection remb_crit_;
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// The last time a REMB was sent.
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int64_t last_remb_time_ms_ RTC_GUARDED_BY(remb_crit_);
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int64_t last_send_bitrate_bps_ RTC_GUARDED_BY(remb_crit_);
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// The last bitrate update.
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int64_t bitrate_bps_ RTC_GUARDED_BY(remb_crit_);
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int64_t max_bitrate_bps_ RTC_GUARDED_BY(remb_crit_);
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// Candidates for the REMB module can be RTP sender/receiver modules, with
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// the sender modules taking precedence.
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std::vector<RtcpFeedbackSenderInterface*> sender_remb_candidates_
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RTC_GUARDED_BY(modules_crit_);
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std::vector<RtcpFeedbackSenderInterface*> receiver_remb_candidates_
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RTC_GUARDED_BY(modules_crit_);
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RtcpFeedbackSenderInterface* active_remb_module_
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RTC_GUARDED_BY(modules_crit_);
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volatile int transport_seq_;
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RTC_DISALLOW_COPY_AND_ASSIGN(PacketRouter);
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};
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} // namespace webrtc
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#endif // MODULES_PACING_PACKET_ROUTER_H_
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