9e03c3b372
Reason for revert: Reverting all CLs related to moving the eventlog, as they break Chromium tests. Original issue's description: > Move RtcEventLog object from inside VoiceEngine to Call. > > In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced. > The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface. > > BUG=webrtc:4741,webrtc:5603,chromium:609749 > R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org > > Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016 > Cr-Commit-Position: refs/heads/master@{#13321} TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4741,webrtc:5603,chromium:609749 Review-Url: https://codereview.webrtc.org/2111813002 Cr-Commit-Position: refs/heads/master@{#13340}
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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