16825b1a82
Existing code was inconsistent about whether to use uint16_t, int, unsigned int, or uint32_t, and sometimes silently truncated one to another, or truncated int64_t. Because most core time-handling functions use int64_t, being consistent about using int64_t unless otherwise necessary minimizes the number of explicit or implicit casts. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
87 lines
3.5 KiB
C++
87 lines
3.5 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 "webrtc/modules/interface/module.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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namespace webrtc {
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class BitrateObserver {
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/*
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* Observer class for the encoders, each encoder should implement this class
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* to get the target bitrate. It also get the fraction loss and rtt to
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* optimize its settings for this type of network. |target_bitrate| is the
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* target media/payload bitrate excluding packet headers, measured in bits
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* per second.
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*/
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public:
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virtual void OnNetworkChanged(uint32_t target_bitrate,
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uint8_t fraction_loss, // 0 - 255.
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int64_t rtt) = 0;
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virtual ~BitrateObserver() {}
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};
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class BitrateController : public Module {
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/*
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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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*/
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public:
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// The argument |enforce_min_bitrate| controls the behavior when the available
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// bitrate is lower than the minimum bitrate, or the sum of minimum bitrates.
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// When true, the bitrate will never be set lower than the minimum bitrate(s).
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// When false, the bitrate observers will be allocated rates up to their
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// respective minimum bitrate, satisfying one observer after the other.
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static BitrateController* CreateBitrateController(Clock* clock,
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bool enforce_min_bitrate);
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virtual ~BitrateController() {}
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virtual RtcpBandwidthObserver* CreateRtcpBandwidthObserver() = 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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/*
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* Set the start and max send bitrate used by the bandwidth management.
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*
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* observer, updates bitrates if already in use.
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* min_bitrate_kbit = 0 equals no min bitrate.
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* max_bitrate_kit = 0 equals no max bitrate.
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*/
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virtual void SetBitrateObserver(BitrateObserver* observer,
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uint32_t start_bitrate,
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uint32_t min_bitrate,
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uint32_t max_bitrate) = 0;
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virtual void RemoveBitrateObserver(BitrateObserver* observer) = 0;
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// Changes the mode that was set in the constructor.
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virtual void EnforceMinBitrate(bool enforce_min_bitrate) = 0;
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virtual void SetReservedBitrate(uint32_t reserved_bitrate_bps) = 0;
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virtual void SetBitrateSent(uint32_t bitrate_sent_bps) = 0;
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virtual void SetCodecMode(webrtc::VideoCodecMode mode) = 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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