/* * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. * * Usage: this class will register multiple RtcpBitrateObserver's one at each * RTCP module. It will aggregate the results and run one bandwidth estimation * and push the result to the encoders via BitrateObserver(s). */ #ifndef MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_ #define MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_ #include #include "modules/congestion_controller/goog_cc/delay_based_bwe.h" #include "modules/include/module.h" #include "modules/pacing/paced_sender.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { class RtcEventLog; // Deprecated // TODO(perkj): Remove BitrateObserver when no implementations use it. class BitrateObserver { // Observer class for bitrate changes announced due to change in bandwidth // estimate or due to bitrate allocation changes. Fraction loss and rtt is // also part of this callback to allow the obsevrer to optimize its settings // for different types of network environments. The bitrate does not include // packet headers and is measured in bits per second. public: virtual void OnNetworkChanged(uint32_t bitrate_bps, uint8_t fraction_loss, // 0 - 255. int64_t rtt_ms) = 0; // TODO(gnish): Merge these two into one function. virtual void OnNetworkChanged(uint32_t bitrate_for_encoder_bps, uint32_t bitrate_for_pacer_bps, bool in_probe_rtt, int64_t target_set_time, uint64_t congestion_window) {} virtual void OnBytesAcked(size_t bytes) {} virtual size_t pacer_queue_size_in_bytes() { return 0; } virtual ~BitrateObserver() {} }; class BitrateController : public Module, public RtcpBandwidthObserver { // This class collects feedback from all streams sent to a peer (via // RTCPBandwidthObservers). It does one aggregated send side bandwidth // estimation and divide the available bitrate between all its registered // BitrateObservers. public: static const int kDefaultStartBitratebps = 300000; // Deprecated: // TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC. // Remove this method once other other projects does not use it. static BitrateController* CreateBitrateController(const Clock* clock, BitrateObserver* observer, RtcEventLog* event_log); static BitrateController* CreateBitrateController(const Clock* clock, RtcEventLog* event_log); virtual ~BitrateController() {} // Deprecated, use raw pointer to BitrateController instance instead. // Creates RtcpBandwidthObserver caller responsible to delete. RTC_DEPRECATED virtual RtcpBandwidthObserver* CreateRtcpBandwidthObserver() = 0; // Deprecated virtual void SetStartBitrate(int start_bitrate_bps) = 0; // Deprecated virtual void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) = 0; virtual void SetBitrates(int start_bitrate_bps, int min_bitrate_bps, int max_bitrate_bps) = 0; virtual void ResetBitrates(int bitrate_bps, int min_bitrate_bps, int max_bitrate_bps) = 0; virtual void OnDelayBasedBweResult(const DelayBasedBwe::Result& result) = 0; // Gets the available payload bandwidth in bits per second. Note that // this bandwidth excludes packet headers. virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0; virtual void SetReservedBitrate(uint32_t reserved_bitrate_bps) = 0; virtual bool GetNetworkParameters(uint32_t* bitrate, uint8_t* fraction_loss, int64_t* rtt) = 0; }; } // namespace webrtc #endif // MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_