9ec883e8bd
BUG=312 TEST=ViE standard autotest and API test. Review URL: https://webrtc-codereview.appspot.com/432005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1836 4adac7df-926f-26a2-2b94-8c16560cd09d
104 lines
3.3 KiB
C++
104 lines
3.3 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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// 1. Register a RtpRtcp module to include in the REMB packet.
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// 2. When UpdateBitrateEstimate is called for the first time for a SSRC, add it
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// to the map.
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// 3. Send a new REMB every kRembSendIntervallMs or if a lower bitrate estimate
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// for a specified SSRC.
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#ifndef WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_REMB_H_
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#define WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_REMB_H_
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#include <list>
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#include <map>
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#include "modules/interface/module.h"
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#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "system_wrappers/interface/scoped_ptr.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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class ProcessThread;
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class RtpRtcp;
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class VieRemb : public RtpRemoteBitrateObserver, public Module {
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public:
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VieRemb(ProcessThread* process_thread);
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~VieRemb();
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// Called to add a receive channel to include in the REMB packet.
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void AddReceiveChannel(RtpRtcp* rtp_rtcp);
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// Removes the specified channel from REMB estimate.
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void RemoveReceiveChannel(RtpRtcp* rtp_rtcp);
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// Called to add a module that can generate and send REMB RTCP.
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void AddRembSender(RtpRtcp* rtp_rtcp);
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// Removes a REMB RTCP sender.
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void RemoveRembSender(RtpRtcp* rtp_rtcp);
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// Called to add a send channel encoding and sending data, affected by
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// received REMB packets.
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void AddSendChannel(RtpRtcp* rtp_rtcp);
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// Removes the specified channel from receiving REMB packet estimates.
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void RemoveSendChannel(RtpRtcp* rtp_rtcp);
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// Returns true if the instance is in use, false otherwise.
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bool InUse() const;
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// Called every time there is a new bitrate estimate for the received stream
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// with given SSRC. This call will trigger a new RTCP REMB packet if the
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// bitrate estimate has decreased or if no RTCP REMB packet has been sent for
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// a certain time interval.
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// Implements RtpReceiveBitrateUpdate.
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virtual void OnReceiveBitrateChanged(unsigned int ssrc, unsigned int bitrate);
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// Called for every new receive REMB packet and distributes the estmate
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// between all sending modules.
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virtual void OnReceivedRemb(unsigned int bitrate);
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// Implements Module.
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virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
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virtual WebRtc_Word32 TimeUntilNextProcess();
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virtual WebRtc_Word32 Process();
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private:
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typedef std::list<RtpRtcp*> RtpModules;
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typedef std::map<unsigned int, unsigned int> SsrcBitrate;
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ProcessThread* process_thread_;
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scoped_ptr<CriticalSectionWrapper> list_crit_;
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// The last time a REMB was sent.
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int64_t last_remb_time_;
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int last_send_bitrate_;
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// All RtpRtcp modules to include in the REMB packet.
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RtpModules receive_modules_;
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// All modules encoding and sending data.
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RtpModules send_modules_;
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// All modules that can send REMB RTCP.
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RtpModules rtcp_sender_;
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// The last bitrate update for each SSRC.
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SsrcBitrate bitrates_;
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_REMB_H_
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