voip_engine.h (3752B)
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_builder = 39 // std::make_unique<BuiltinAudioProcessingBuilder>(); 40 // 41 // auto voip_engine = CreateVoipEngine(std::move(config)); 42 // 43 // auto& voip_base = voip_engine->Base(); 44 // auto& voip_codec = voip_engine->Codec(); 45 // auto& voip_network = voip_engine->Network(); 46 // 47 // ChannelId channel = voip_base.CreateChannel(&app_transport_); 48 // 49 // // After SDP offer/answer, set payload type and codecs that have been 50 // // decided through SDP negotiation. 51 // // VoipResult handling omitted here. 52 // voip_codec.SetSendCodec(channel, ...); 53 // voip_codec.SetReceiveCodecs(channel, ...); 54 // 55 // // Start sending and playing RTP on voip channel. 56 // // VoipResult handling omitted here. 57 // voip_base.StartSend(channel); 58 // voip_base.StartPlayout(channel); 59 // 60 // // Inject received RTP/RTCP through VoipNetwork interface. 61 // // VoipResult handling omitted here. 62 // voip_network.ReceivedRTPPacket(channel, ...); 63 // voip_network.ReceivedRTCPPacket(channel, ...); 64 // 65 // // Stop and release voip channel. 66 // // VoipResult handling omitted here. 67 // voip_base.StopSend(channel); 68 // voip_base.StopPlayout(channel); 69 // voip_base.ReleaseChannel(channel); 70 // 71 class VoipEngine { 72 public: 73 virtual ~VoipEngine() = default; 74 75 // VoipBase is the audio session management interface that 76 // creates/releases/starts/stops an one-to-one audio media session. 77 virtual VoipBase& Base() = 0; 78 79 // VoipNetwork provides injection APIs that would enable application 80 // to send and receive RTP/RTCP packets. There is no default network module 81 // that provides RTP transmission and reception. 82 virtual VoipNetwork& Network() = 0; 83 84 // VoipCodec provides codec configuration APIs for encoder and decoders. 85 virtual VoipCodec& Codec() = 0; 86 87 // VoipDtmf provides DTMF event APIs to register and send DTMF events. 88 virtual VoipDtmf& Dtmf() = 0; 89 90 // VoipStatistics provides performance metrics around audio decoding module 91 // and jitter buffer (NetEq). 92 virtual VoipStatistics& Statistics() = 0; 93 94 // VoipVolumeControl provides various input/output volume control. 95 virtual VoipVolumeControl& VolumeControl() = 0; 96 }; 97 98 } // namespace webrtc 99 100 #endif // API_VOIP_VOIP_ENGINE_H_