1 /* 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_ 12 #define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_ 13 14 // This sub-API supports the following functionalities: 15 // - Configuring send and receive addresses. 16 // - External transport support. 17 // - Port and address filters. 18 // - Windows GQoS functions and ToS functions. 19 // - Packet timeout notification. 20 // - DeadorAlive connection observations. 21 22 #include "webrtc/common_types.h" 23 24 namespace webrtc { 25 26 class Transport; 27 class VideoEngine; 28 29 // This enumerator describes VideoEngine packet timeout states. 30 enum ViEPacketTimeout { 31 NoPacket = 0, 32 PacketReceived = 1 33 }; 34 35 class WEBRTC_DLLEXPORT ViENetwork { 36 public: 37 // Default values. 38 enum { KDefaultSampleTimeSeconds = 2 }; 39 40 // Factory for the ViENetwork subAPI and increases an internal reference 41 // counter if successful. Returns NULL if the API is not supported or if 42 // construction fails. 43 static ViENetwork* GetInterface(VideoEngine* video_engine); 44 45 // Releases the ViENetwork sub-API and decreases an internal reference 46 // counter.Returns the new reference count. This value should be zero 47 // for all sub-API:s before the VideoEngine object can be safely deleted. 48 virtual int Release() = 0; 49 50 // Inform the engine about if the network adapter is currently transmitting 51 // packets or not. 52 virtual void SetNetworkTransmissionState(const int video_channel, 53 const bool is_transmitting) = 0; 54 55 // This function registers a user implementation of Transport to use for 56 // sending RTP and RTCP packets on this channel. 57 virtual int RegisterSendTransport(const int video_channel, 58 Transport& transport) = 0; 59 60 // This function deregisters a used Transport for a specified channel. 61 virtual int DeregisterSendTransport(const int video_channel) = 0; 62 63 // When using external transport for a channel, received RTP packets should 64 // be passed to VideoEngine using this function. The input should contain 65 // the RTP header and payload. 66 virtual int ReceivedRTPPacket(const int video_channel, 67 const void* data, 68 const int length, 69 const PacketTime& packet_time) = 0; 70 71 // When using external transport for a channel, received RTCP packets should 72 // be passed to VideoEngine using this function. 73 virtual int ReceivedRTCPPacket(const int video_channel, 74 const void* data, 75 const int length) = 0; 76 77 // This function sets the Maximum Transition Unit (MTU) for a channel. The 78 // RTP packet will be packetized based on this MTU to optimize performance 79 // over the network. 80 virtual int SetMTU(int video_channel, unsigned int mtu) = 0; 81 82 // Forward (audio) packet to bandwidth estimator for the given video channel, 83 // for aggregated audio+video BWE. 84 virtual int ReceivedBWEPacket(const int video_channel, 85 int64_t arrival_time_ms, int payload_size, const RTPHeader& header) { 86 return 0; 87 } 88 89 // TODO(holmer): Remove the default implementation when this has been fixed 90 // in fakewebrtcvideoengine.cc. 91 virtual bool SetBandwidthEstimationConfig(int video_channel, 92 const webrtc::Config& config) { 93 return false; 94 } 95 96 protected: 97 ViENetwork() {} 98 virtual ~ViENetwork() {} 99 }; 100 101 } // namespace webrtc 102 103 #endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_ 104
