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bcf91808a2
This will allow experimenting with audio bitrate allocation in video calls without increasing transport overhead. Bug: webrtc:8243 Change-Id: If961780921d53bdce95b68c26641df6875509c1f Reviewed-on: https://webrtc-review.googlesource.com/84501 Commit-Queue: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23755}
73 lines
3.1 KiB
C++
73 lines
3.1 KiB
C++
/*
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* Copyright (c) 2018 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_CONGESTION_CONTROLLER_INCLUDE_SEND_SIDE_CONGESTION_CONTROLLER_INTERFACE_H_
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#define MODULES_CONGESTION_CONTROLLER_INCLUDE_SEND_SIDE_CONGESTION_CONTROLLER_INTERFACE_H_
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#include <memory>
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#include <vector>
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#include "modules/congestion_controller/include/network_changed_observer.h"
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#include "modules/congestion_controller/transport_feedback_adapter.h"
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#include "modules/include/module.h"
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#include "modules/include/module_common_types.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/networkroute.h"
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namespace rtc {
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struct SentPacket;
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}
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namespace webrtc {
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class BitrateController;
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class Clock;
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class AcknowledgedBitrateEstimator;
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class ProbeController;
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class RateLimiter;
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class RtcEventLog;
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class SendSideCongestionControllerInterface : public CallStatsObserver,
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public Module,
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public TransportFeedbackObserver {
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public:
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SendSideCongestionControllerInterface() = default;
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~SendSideCongestionControllerInterface() override = default;
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virtual void RegisterPacketFeedbackObserver(
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PacketFeedbackObserver* observer) = 0;
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virtual void DeRegisterPacketFeedbackObserver(
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PacketFeedbackObserver* observer) = 0;
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virtual void RegisterNetworkObserver(NetworkChangedObserver* observer) = 0;
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virtual void SetBweBitrates(int min_bitrate_bps,
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int start_bitrate_bps,
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int max_bitrate_bps) = 0;
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virtual void SetAllocatedSendBitrateLimits(int64_t min_send_bitrate_bps,
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int64_t max_padding_bitrate_bps,
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int64_t max_total_bitrate_bps) = 0;
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virtual void OnNetworkRouteChanged(const rtc::NetworkRoute& network_route,
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int bitrate_bps,
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int min_bitrate_bps,
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int max_bitrate_bps) = 0;
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virtual void SignalNetworkState(NetworkState state) = 0;
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virtual RtcpBandwidthObserver* GetBandwidthObserver() = 0;
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virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
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virtual TransportFeedbackObserver* GetTransportFeedbackObserver() = 0;
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virtual void SetPerPacketFeedbackAvailable(bool available) = 0;
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virtual void EnablePeriodicAlrProbing(bool enable) = 0;
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virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
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virtual void SetPacingFactor(float pacing_factor) = 0;
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virtual void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) = 0;
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RTC_DISALLOW_COPY_AND_ASSIGN(SendSideCongestionControllerInterface);
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};
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} // namespace webrtc
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#endif // MODULES_CONGESTION_CONTROLLER_INCLUDE_SEND_SIDE_CONGESTION_CONTROLLER_INTERFACE_H_
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