| /* |
| * 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 WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_ |
| #define WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_ |
| |
| #include <map> |
| |
| #include "webrtc/modules/congestion_controller/delay_based_bwe.h" |
| #include "webrtc/modules/include/module.h" |
| #include "webrtc/modules/pacing/paced_sender.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| |
| namespace webrtc { |
| |
| class CriticalSectionWrapper; |
| class RtcEventLog; |
| struct PacketInfo; |
| |
| // 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; |
| |
| virtual ~BitrateObserver() {} |
| }; |
| |
| class BitrateController : public Module { |
| // 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(Clock* clock, |
| BitrateObserver* observer, |
| RtcEventLog* event_log); |
| |
| static BitrateController* CreateBitrateController(Clock* clock, |
| RtcEventLog* event_log); |
| |
| virtual ~BitrateController() {} |
| |
| 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 // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_ |