Reason for revert: Fails in the waterfall here: https://build.chromium.org/p/client.webrtc/builders/Win32%20Debug/builds/7832/steps/rtc_media_unittests/logs/stdio Original issue's description: > Remove SendPacer from ViEEncoder > > This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator) > > BUG=webrtc:5687 > > Committed: https://crrev.com/857c5ccdb56e4c94196f7c6227abd5993c95abe2 > Cr-Commit-Position: refs/heads/master@{#12620} TBR=stefan@webrtc.org,mflodman@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5687 Review-Url: https://codereview.webrtc.org/1947853002 Cr-Commit-Position: refs/heads/master@{#12621}
77 lines
2.9 KiB
C++
77 lines
2.9 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/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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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 kDefaultStartBitrateKbps = 300;
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static BitrateController* CreateBitrateController(Clock* clock,
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BitrateObserver* observer);
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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 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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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
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