This is a follow up of https://webrtc-review.googlesource.com/c/src/+/43201. Issue 43201 didn't do the job properly. 1. The audio rtcp report interval is not properly hooked up. 2. We don't need to propagate audio rtcp interval into video send stream or vice versa. 3. We don't need to propagate rtcp report interval to any receiving streams. Bug: webrtc:8789 Change-Id: I1f637d6e5173608564ef0702d7eda6fc93b3200f Reviewed-on: https://webrtc-review.googlesource.com/c/110105 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#25610}
136 lines
5.4 KiB
C++
136 lines
5.4 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 CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
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#define CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/transport/network_control.h"
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#include "call/rtp_bitrate_configurator.h"
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#include "call/rtp_transport_controller_send_interface.h"
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#include "call/rtp_video_sender.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/congestion_controller/include/send_side_congestion_controller_interface.h"
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#include "modules/pacing/packet_router.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/networkroute.h"
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#include "rtc_base/task_queue.h"
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namespace webrtc {
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class Clock;
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class FrameEncryptorInterface;
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class RtcEventLog;
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// TODO(nisse): When we get the underlying transports here, we should
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// have one object implementing RtpTransportControllerSendInterface
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// per transport, sharing the same congestion controller.
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class RtpTransportControllerSend final
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: public RtpTransportControllerSendInterface,
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public NetworkChangedObserver {
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public:
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RtpTransportControllerSend(
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Clock* clock,
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RtcEventLog* event_log,
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NetworkControllerFactoryInterface* controller_factory,
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const BitrateConstraints& bitrate_config);
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~RtpTransportControllerSend() override;
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RtpVideoSenderInterface* CreateRtpVideoSender(
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const std::vector<uint32_t>& ssrcs,
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std::map<uint32_t, RtpState> suspended_ssrcs,
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const std::map<uint32_t, RtpPayloadState>&
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states, // move states into RtpTransportControllerSend
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const RtpConfig& rtp_config,
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int rtcp_report_interval_ms,
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Transport* send_transport,
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const RtpSenderObservers& observers,
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RtcEventLog* event_log,
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std::unique_ptr<FecController> fec_controller,
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const RtpSenderFrameEncryptionConfig& frame_encryption_config) override;
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void DestroyRtpVideoSender(
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RtpVideoSenderInterface* rtp_video_sender) override;
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// Implements NetworkChangedObserver interface.
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void OnNetworkChanged(uint32_t bitrate_bps,
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uint8_t fraction_loss,
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int64_t rtt_ms,
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int64_t probing_interval_ms) override;
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// Implements RtpTransportControllerSendInterface
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rtc::TaskQueue* GetWorkerQueue() override;
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PacketRouter* packet_router() override;
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TransportFeedbackObserver* transport_feedback_observer() override;
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RtpPacketSender* packet_sender() override;
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const RtpKeepAliveConfig& keepalive_config() const override;
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void SetAllocatedSendBitrateLimits(int min_send_bitrate_bps,
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int max_padding_bitrate_bps,
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int max_total_bitrate_bps) override;
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void SetKeepAliveConfig(const RtpKeepAliveConfig& config);
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void SetPacingFactor(float pacing_factor) override;
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void SetQueueTimeLimit(int limit_ms) override;
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CallStatsObserver* GetCallStatsObserver() override;
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void RegisterPacketFeedbackObserver(
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PacketFeedbackObserver* observer) override;
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void DeRegisterPacketFeedbackObserver(
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PacketFeedbackObserver* observer) override;
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void RegisterTargetTransferRateObserver(
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TargetTransferRateObserver* observer) override;
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void OnNetworkRouteChanged(const std::string& transport_name,
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const rtc::NetworkRoute& network_route) override;
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void OnNetworkAvailability(bool network_available) override;
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RtcpBandwidthObserver* GetBandwidthObserver() override;
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int64_t GetPacerQueuingDelayMs() const override;
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int64_t GetFirstPacketTimeMs() const override;
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void SetPerPacketFeedbackAvailable(bool available) override;
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void EnablePeriodicAlrProbing(bool enable) override;
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void OnSentPacket(const rtc::SentPacket& sent_packet) override;
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void SetSdpBitrateParameters(const BitrateConstraints& constraints) override;
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void SetClientBitratePreferences(const BitrateSettings& preferences) override;
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void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) override;
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void OnTransportOverheadChanged(
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size_t transport_overhead_per_packet) override;
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private:
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const Clock* const clock_;
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PacketRouter packet_router_;
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std::vector<std::unique_ptr<RtpVideoSenderInterface>> video_rtp_senders_;
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PacedSender pacer_;
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RtpKeepAliveConfig keepalive_;
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RtpBitrateConfigurator bitrate_configurator_;
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std::map<std::string, rtc::NetworkRoute> network_routes_;
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const std::unique_ptr<ProcessThread> process_thread_;
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rtc::CriticalSection observer_crit_;
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TargetTransferRateObserver* observer_ RTC_GUARDED_BY(observer_crit_);
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std::unique_ptr<SendSideCongestionControllerInterface> send_side_cc_;
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RateLimiter retransmission_rate_limiter_;
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// TODO(perkj): |task_queue_| is supposed to replace |process_thread_|.
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// |task_queue_| is defined last to ensure all pending tasks are cancelled
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// and deleted before any other members.
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rtc::TaskQueue task_queue_;
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RTC_DISALLOW_COPY_AND_ASSIGN(RtpTransportControllerSend);
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};
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} // namespace webrtc
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#endif // CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
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