voip_engine.h - mozsearch

/*

 *  Copyright (c) 2020 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 API_VOIP_VOIP_ENGINE_H_

#define API_VOIP_VOIP_ENGINE_H_

namespace webrtc {

class VoipBase;

class VoipCodec;

class VoipNetwork;

class VoipDtmf;

class VoipStatistics;

class VoipVolumeControl;

// VoipEngine is the main interface serving as the entry point for all VoIP

// APIs. A single instance of VoipEngine should suffice the most of the need for

// typical VoIP applications as it handles multiple media sessions including a

// specialized session type like ad-hoc conference. Below example code

// describes the typical sequence of API usage. Each API header contains more

// description on what the methods are used for.

//

//   // Caller is responsible of setting desired audio components.

//   VoipEngineConfig config;

//   config.encoder_factory = CreateBuiltinAudioEncoderFactory();

//   config.decoder_factory = CreateBuiltinAudioDecoderFactory();

//   config.task_queue_factory = CreateDefaultTaskQueueFactory();

//   config.audio_device =

//       AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio,

//                                 config.task_queue_factory.get());

//   config.audio_processing = AudioProcessingBuilder().Create();

//

//   auto voip_engine = CreateVoipEngine(std::move(config));

//

//   auto& voip_base = voip_engine->Base();

//   auto& voip_codec = voip_engine->Codec();

//   auto& voip_network = voip_engine->Network();

//

//   ChannelId channel = voip_base.CreateChannel(&app_transport_);

//

//   // After SDP offer/answer, set payload type and codecs that have been

//   // decided through SDP negotiation.

//   // VoipResult handling omitted here.

//   voip_codec.SetSendCodec(channel, ...);

//   voip_codec.SetReceiveCodecs(channel, ...);

//

//   // Start sending and playing RTP on voip channel.

//   // VoipResult handling omitted here.

//   voip_base.StartSend(channel);

//   voip_base.StartPlayout(channel);

//

//   // Inject received RTP/RTCP through VoipNetwork interface.

//   // VoipResult handling omitted here.

//   voip_network.ReceivedRTPPacket(channel, ...);

//   voip_network.ReceivedRTCPPacket(channel, ...);

//

//   // Stop and release voip channel.

//   // VoipResult handling omitted here.

//   voip_base.StopSend(channel);

//   voip_base.StopPlayout(channel);

//   voip_base.ReleaseChannel(channel);

//

class VoipEngine {

 public:

  virtual ~VoipEngine() = default;

  // VoipBase is the audio session management interface that

  // creates/releases/starts/stops an one-to-one audio media session.

  virtual VoipBase& Base() = 0;

  // VoipNetwork provides injection APIs that would enable application

  // to send and receive RTP/RTCP packets. There is no default network module

  // that provides RTP transmission and reception.

  virtual VoipNetwork& Network() = 0;

  // VoipCodec provides codec configuration APIs for encoder and decoders.

  virtual VoipCodec& Codec() = 0;

  // VoipDtmf provides DTMF event APIs to register and send DTMF events.

  virtual VoipDtmf& Dtmf() = 0;

  // VoipStatistics provides performance metrics around audio decoding module

  // and jitter buffer (NetEq).

  virtual VoipStatistics& Statistics() = 0;

  // VoipVolumeControl provides various input/output volume control.

  virtual VoipVolumeControl& VolumeControl() = 0;

};

}  // namespace webrtc

#endif  // API_VOIP_VOIP_ENGINE_H_