ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
79 lines
2.3 KiB
C++
79 lines
2.3 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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#ifndef WEBRTC_VIDEO_ENGINE_VIE_REMB_H_
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#define WEBRTC_VIDEO_ENGINE_VIE_REMB_H_
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#include <list>
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#include <utility>
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#include <vector>
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/include/module.h"
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#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.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 ProcessThread;
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class RtpRtcp;
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class VieRemb : public RemoteBitrateObserver {
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public:
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VieRemb();
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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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// 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 a receive channel
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// group. This call will trigger a new RTCP REMB packet if the bitrate
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// 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(const std::vector<unsigned int>& ssrcs,
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unsigned int bitrate);
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private:
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typedef std::list<RtpRtcp*> RtpModules;
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rtc::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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unsigned 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 that can send REMB RTCP.
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RtpModules rtcp_sender_;
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// The last bitrate update.
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unsigned int bitrate_;
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_VIE_REMB_H_
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