1 /* 2 * Copyright (c) 2020 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 API_VOIP_VOIP_ENGINE_H_ 12 #define API_VOIP_VOIP_ENGINE_H_ 13 14 namespace webrtc { 15 16 class VoipBase; 17 class VoipCodec; 18 class VoipNetwork; 19 class VoipDtmf; 20 class VoipStatistics; 21 class VoipVolumeControl; 22 23 // VoipEngine is the main interface serving as the entry point for all VoIP 24 // APIs. A single instance of VoipEngine should suffice the most of the need for 25 // typical VoIP applications as it handles multiple media sessions including a 26 // specialized session type like ad-hoc conference. Below example code 27 // describes the typical sequence of API usage. Each API header contains more 28 // description on what the methods are used for. 29 // 30 // // Caller is responsible of setting desired audio components. 31 // VoipEngineConfig config; 32 // config.encoder_factory = CreateBuiltinAudioEncoderFactory(); 33 // config.decoder_factory = CreateBuiltinAudioDecoderFactory(); 34 // config.task_queue_factory = CreateDefaultTaskQueueFactory(); 35 // config.audio_device = 36 // AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio, 37 // config.task_queue_factory.get()); 38 // config.audio_processing = AudioProcessingBuilder().Create(); 39 // 40 // auto voip_engine = CreateVoipEngine(std::move(config)); 41 // 42 // auto& voip_base = voip_engine->Base(); 43 // auto& voip_codec = voip_engine->Codec(); 44 // auto& voip_network = voip_engine->Network(); 45 // 46 // ChannelId channel = voip_base.CreateChannel(&app_transport_); 47 // 48 // // After SDP offer/answer, set payload type and codecs that have been 49 // // decided through SDP negotiation. 50 // // VoipResult handling omitted here. 51 // voip_codec.SetSendCodec(channel, ...); 52 // voip_codec.SetReceiveCodecs(channel, ...); 53 // 54 // // Start sending and playing RTP on voip channel. 55 // // VoipResult handling omitted here. 56 // voip_base.StartSend(channel); 57 // voip_base.StartPlayout(channel); 58 // 59 // // Inject received RTP/RTCP through VoipNetwork interface. 60 // // VoipResult handling omitted here. 61 // voip_network.ReceivedRTPPacket(channel, ...); 62 // voip_network.ReceivedRTCPPacket(channel, ...); 63 // 64 // // Stop and release voip channel. 65 // // VoipResult handling omitted here. 66 // voip_base.StopSend(channel); 67 // voip_base.StopPlayout(channel); 68 // voip_base.ReleaseChannel(channel); 69 // 70 class VoipEngine { 71 public: 72 virtual ~VoipEngine() = default; 73 74 // VoipBase is the audio session management interface that 75 // creates/releases/starts/stops an one-to-one audio media session. 76 virtual VoipBase& Base() = 0; 77 78 // VoipNetwork provides injection APIs that would enable application 79 // to send and receive RTP/RTCP packets. There is no default network module 80 // that provides RTP transmission and reception. 81 virtual VoipNetwork& Network() = 0; 82 83 // VoipCodec provides codec configuration APIs for encoder and decoders. 84 virtual VoipCodec& Codec() = 0; 85 86 // VoipDtmf provides DTMF event APIs to register and send DTMF events. 87 virtual VoipDtmf& Dtmf() = 0; 88 89 // VoipStatistics provides performance metrics around audio decoding module 90 // and jitter buffer (NetEq). 91 virtual VoipStatistics& Statistics() = 0; 92 93 // VoipVolumeControl provides various input/output volume control. 94 virtual VoipVolumeControl& VolumeControl() = 0; 95 }; 96 97 } // namespace webrtc 98 99 #endif // API_VOIP_VOIP_ENGINE_H_ 100