45087cd23f
Ownership of the retransmission rate limiter for video is moved from send side congestion controller to Call. This is to reduce the interface on the rtp transport controller send. Bug: webrtc:8415 Change-Id: Ie9c7317400a9eb61a3c8325b9e527844ffc13769 Reviewed-on: https://webrtc-review.googlesource.com/58745 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22254}
89 lines
3.6 KiB
C++
89 lines
3.6 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 "call/rtp_bitrate_configurator.h"
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#include "call/rtp_transport_controller_send_interface.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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namespace webrtc {
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class Clock;
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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 : public RtpTransportControllerSendInterface {
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public:
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RtpTransportControllerSend(Clock* clock,
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RtcEventLog* event_log,
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const BitrateConstraints& bitrate_config);
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~RtpTransportControllerSend() override;
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// Implements RtpTransportControllerSendInterface
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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 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 RegisterNetworkObserver(NetworkChangedObserver* 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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bool AvailableBandwidth(uint32_t* bandwidth) const override;
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int64_t GetPacerQueuingDelayMs() const override;
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int64_t GetFirstPacketTimeMs() const 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(
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const BitrateConstraintsMask& preferences) override;
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private:
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PacketRouter packet_router_;
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PacedSender pacer_;
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const std::unique_ptr<SendSideCongestionControllerInterface> send_side_cc_;
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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_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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