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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
// This sub-API supports the following functionalities:
// - Configuring send and receive addresses.
// - External transport support.
// - Port and address filters.
// - Windows GQoS functions and ToS functions.
// - Packet timeout notification.
// - Dead‐or‐Alive connection observations.
#include "webrtc/common_types.h"
namespace webrtc {
class Transport;
class VideoEngine;
// This enumerator describes VideoEngine packet timeout states.
enum ViEPacketTimeout {
NoPacket = 0,
PacketReceived = 1
// Default values.
enum { KDefaultSampleTimeSeconds = 2 };
// Factory for the ViENetwork sub‐API and increases an internal reference
// counter if successful. Returns NULL if the API is not supported or if
// construction fails.
static ViENetwork* GetInterface(VideoEngine* video_engine);
// Releases the ViENetwork sub-API and decreases an internal reference
// counter.Returns the new reference count. This value should be zero
// for all sub-API:s before the VideoEngine object can be safely deleted.
virtual int Release() = 0;
virtual void SetBitrateConfig(int video_channel,
int min_bitrate_bps,
int start_bitrate_bps,
int max_bitrate_bps) = 0;
// Inform the engine about if the network adapter is currently transmitting
// packets or not.
virtual void SetNetworkTransmissionState(const int video_channel,
const bool is_transmitting) = 0;
// This function registers a user implementation of Transport to use for
// sending RTP and RTCP packets on this channel.
virtual int RegisterSendTransport(const int video_channel,
Transport& transport) = 0;
// This function deregisters a used Transport for a specified channel.
virtual int DeregisterSendTransport(const int video_channel) = 0;
// When using external transport for a channel, received RTP packets should
// be passed to VideoEngine using this function. The input should contain
// the RTP header and payload.
virtual int ReceivedRTPPacket(const int video_channel,
const void* data,
const size_t length,
const PacketTime& packet_time) = 0;
// When using external transport for a channel, received RTCP packets should
// be passed to VideoEngine using this function.
virtual int ReceivedRTCPPacket(const int video_channel,
const void* data,
const size_t length) = 0;
// This function sets the Maximum Transition Unit (MTU) for a channel. The
// RTP packet will be packetized based on this MTU to optimize performance
// over the network.
virtual int SetMTU(int video_channel, unsigned int mtu) = 0;
// Forward (audio) packet to bandwidth estimator for the given video channel,
// for aggregated audio+video BWE.
virtual int ReceivedBWEPacket(const int video_channel,
int64_t arrival_time_ms, size_t payload_size, const RTPHeader& header) {
return 0;
ViENetwork() {}
virtual ~ViENetwork() {}
} // namespace webrtc