822fbd8b68
Together with Stefan's http://review.webrtc.org/1960004/. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2048004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
89 lines
3.2 KiB
C++
89 lines
3.2 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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#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_
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#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_
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// This sub-API supports the following functionalities:
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// - Configuring send and receive addresses.
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// - External transport support.
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// - Port and address filters.
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// - Windows GQoS functions and ToS functions.
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// - Packet timeout notification.
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// - Dead‐or‐Alive connection observations.
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#include "webrtc/common_types.h"
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namespace webrtc {
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class Transport;
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class VideoEngine;
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// This enumerator describes VideoEngine packet timeout states.
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enum ViEPacketTimeout {
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NoPacket = 0,
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PacketReceived = 1
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};
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class WEBRTC_DLLEXPORT ViENetwork {
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public:
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// Default values.
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enum { KDefaultSampleTimeSeconds = 2 };
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// Factory for the ViENetwork sub‐API and increases an internal reference
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// counter if successful. Returns NULL if the API is not supported or if
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// construction fails.
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static ViENetwork* GetInterface(VideoEngine* video_engine);
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// Releases the ViENetwork sub-API and decreases an internal reference
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// counter.Returns the new reference count. This value should be zero
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// for all sub-API:s before the VideoEngine object can be safely deleted.
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virtual int Release() = 0;
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// Inform the engine about if the network adapter is currently transmitting
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// packets or not.
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virtual void SetNetworkTransmissionState(const int video_channel,
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const bool is_transmitting) = 0;
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// This function registers a user implementation of Transport to use for
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// sending RTP and RTCP packets on this channel.
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virtual int RegisterSendTransport(const int video_channel,
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Transport& transport) = 0;
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// This function deregisters a used Transport for a specified channel.
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virtual int DeregisterSendTransport(const int video_channel) = 0;
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// When using external transport for a channel, received RTP packets should
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// be passed to VideoEngine using this function. The input should contain
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// the RTP header and payload.
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virtual int ReceivedRTPPacket(const int video_channel,
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const void* data,
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const int length) = 0;
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// When using external transport for a channel, received RTCP packets should
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// be passed to VideoEngine using this function.
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virtual int ReceivedRTCPPacket(const int video_channel,
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const void* data,
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const int length) = 0;
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// This function sets the Maximum Transition Unit (MTU) for a channel. The
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// RTP packet will be packetized based on this MTU to optimize performance
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// over the network.
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virtual int SetMTU(int video_channel, unsigned int mtu) = 0;
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protected:
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ViENetwork() {}
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virtual ~ViENetwork() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_
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