6d26ef76ea
In a later CL: - References -> const or ptr. Review URL: http://webrtc-codereview.appspot.com/291003 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1011 4adac7df-926f-26a2-2b94-8c16560cd09d
65 lines
1.7 KiB
C++
65 lines
1.7 KiB
C++
/*
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* Copyright (c) 2011 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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// ViESender is responsible for encrypting, if enabled, packets and send to
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// network.
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#ifndef WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
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#define WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
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#include "common_types.h"
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#include "engine_configurations.h"
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#include "typedefs.h"
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#include "vie_defines.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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class RtpDump;
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class Transport;
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class VideoCodingModule;
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class ViESender: public Transport {
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public:
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ViESender(int engine_id, int channel_id);
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~ViESender();
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// Registers an encryption class to use before sending packets.
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int RegisterExternalEncryption(Encryption* encryption);
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int DeregisterExternalEncryption();
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// Registers transport to use for sending RTP and RTCP.
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int RegisterSendTransport(Transport* transport);
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int DeregisterSendTransport();
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// Stores all incoming packets to file.
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int StartRTPDump(const char file_nameUTF8[1024]);
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int StopRTPDump();
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// Implements Transport.
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virtual int SendPacket(int vie_id, const void* data, int len);
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virtual int SendRTCPPacket(int vie_id, const void* data, int len);
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private:
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int engine_id_;
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int channel_id_;
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CriticalSectionWrapper& critsect_;
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Encryption* external_encryption_;
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WebRtc_UWord8* encryption_buffer_;
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Transport* transport_;
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RtpDump* rtp_dump_;
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
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