5b5d2cdad7
This reverts commit 72d41aa6da94dacb8a8464d1abd4ca7d1afffc65. New change made: Do not reset the BWE when the new network route is not ready to send media. BUG= R=pthatcher@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2041593002 . Cr-Commit-Position: refs/heads/master@{#13280}
93 lines
3.6 KiB
C++
93 lines
3.6 KiB
C++
/*
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* Copyright (c) 2012 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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* Usage: this class will register multiple RtcpBitrateObserver's one at each
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* RTCP module. It will aggregate the results and run one bandwidth estimation
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* and push the result to the encoders via BitrateObserver(s).
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*/
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#ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
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#define WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
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#include <map>
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#include "webrtc/modules/include/module.h"
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#include "webrtc/modules/pacing/paced_sender.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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class RtcEventLog;
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struct PacketInfo;
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// Deprecated
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// TODO(perkj): Remove BitrateObserver when no implementations use it.
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class BitrateObserver {
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// Observer class for bitrate changes announced due to change in bandwidth
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// estimate or due to bitrate allocation changes. Fraction loss and rtt is
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// also part of this callback to allow the obsevrer to optimize its settings
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// for different types of network environments. The bitrate does not include
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// packet headers and is measured in bits per second.
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public:
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virtual void OnNetworkChanged(uint32_t bitrate_bps,
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uint8_t fraction_loss, // 0 - 255.
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int64_t rtt_ms) = 0;
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virtual ~BitrateObserver() {}
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};
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class BitrateController : public Module {
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// This class collects feedback from all streams sent to a peer (via
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// RTCPBandwidthObservers). It does one aggregated send side bandwidth
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// estimation and divide the available bitrate between all its registered
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// BitrateObservers.
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public:
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static const int kDefaultStartBitratebps = 300000;
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// Deprecated:
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// TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
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// Remove this method once other other projects does not use it.
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static BitrateController* CreateBitrateController(Clock* clock,
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BitrateObserver* observer);
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static BitrateController* CreateBitrateController(Clock* clock);
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virtual ~BitrateController() {}
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virtual RtcpBandwidthObserver* CreateRtcpBandwidthObserver() = 0;
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// Deprecated
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virtual void SetStartBitrate(int start_bitrate_bps) = 0;
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// Deprecated
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virtual void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) = 0;
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virtual void SetBitrates(int start_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 ResetBitrates(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 UpdateDelayBasedEstimate(uint32_t bitrate_bps) = 0;
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virtual void SetEventLog(RtcEventLog* event_log) = 0;
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// Gets the available payload bandwidth in bits per second. Note that
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// this bandwidth excludes packet headers.
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virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
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virtual void SetReservedBitrate(uint32_t reserved_bitrate_bps) = 0;
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virtual bool GetNetworkParameters(uint32_t* bitrate,
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uint8_t* fraction_loss,
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int64_t* rtt) = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
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