1e51a388bc
Some clients will not count audio packets into the bandwidth estimate despite negotiating e.g. abs-send-time for that SSRC. If padding is sent on such an RTP module, we might get stuck in a low resolution. This CL works around that by preferring to send padding on video SSRCs. Bug: webrtc:11196 Change-Id: I1ff503a31a85bc32315006a4f15f8b08e5d4e883 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161941 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30066}
132 lines
5.2 KiB
C++
132 lines
5.2 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 <utility>
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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/rtcp_packet.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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// 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 RemoteBitrateObserver,
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public TransportFeedbackSenderInterface {
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public:
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PacketRouter();
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explicit PacketRouter(uint16_t start_transport_seq);
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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 void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
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const PacedPacketInfo& cluster_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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uint16_t CurrentTransportSequenceNumber() const;
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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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// Sends |packets| in one or more IP packets.
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bool SendCombinedRtcpPacket(
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std::vector<std::unique_ptr<rtcp::RtcpPacket>> packets) override;
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private:
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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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void AddSendRtpModuleToMap(RtpRtcp* rtp_module, uint32_t ssrc)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
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void RemoveSendRtpModuleFromMap(uint32_t ssrc)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
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rtc::CriticalSection modules_crit_;
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// Ssrc to RtpRtcp module;
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std::unordered_map<uint32_t, RtpRtcp*> send_modules_map_
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RTC_GUARDED_BY(modules_crit_);
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std::list<RtpRtcp*> send_modules_list_ 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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uint64_t transport_seq_ RTC_GUARDED_BY(modules_crit_);
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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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