xref: /external/chromium_org/third_party/webrtc/video_engine/include/vie_network.h

/*
 *  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.
 */

#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_
#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_

// 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
};

class WEBRTC_DLLEXPORT ViENetwork {
 public:
  // 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;

  // 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 int 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 int 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, int payload_size, const RTPHeader& header) {
    return 0;
  }

  // TODO(holmer): Remove the default implementation when this has been fixed
  // in fakewebrtcvideoengine.cc.
  virtual bool SetBandwidthEstimationConfig(int video_channel,
                                            const webrtc::Config& config) {
    return false;
  }

 protected:
  ViENetwork() {}
  virtual ~ViENetwork() {}
};

}  // namespace webrtc

#endif  // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_