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 20 // VoipEngine is the main interface serving as the entry point for all VoIP 21 // APIs. A single instance of VoipEngine should suffice the most of the need for 22 // typical VoIP applications as it handles multiple media sessions including a 23 // specialized session type like ad-hoc mesh conferencing. Below example code 24 // describes the typical sequence of API usage. Each API header contains more 25 // description on what the methods are used for. 26 // 27 // // Caller is responsible of setting desired audio components. 28 // VoipEngineConfig config; 29 // config.encoder_factory = CreateBuiltinAudioEncoderFactory(); 30 // config.decoder_factory = CreateBuiltinAudioDecoderFactory(); 31 // config.task_queue_factory = CreateDefaultTaskQueueFactory(); 32 // config.audio_device = 33 // AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio, 34 // config.task_queue_factory.get()); 35 // config.audio_processing = AudioProcessingBuilder().Create(); 36 // 37 // auto voip_engine = CreateVoipEngine(std::move(config)); 38 // if (!voip_engine) return some_failure; 39 // 40 // auto& voip_base = voip_engine->Base(); 41 // auto& voip_codec = voip_engine->Codec(); 42 // auto& voip_network = voip_engine->Network(); 43 // 44 // absl::optional<ChannelId> channel = 45 // voip_base.CreateChannel(&app_transport_); 46 // if (!channel) return some_failure; 47 // 48 // // After SDP offer/answer, set payload type and codecs that have been 49 // // decided through SDP negotiation. 50 // voip_codec.SetSendCodec(*channel, ...); 51 // voip_codec.SetReceiveCodecs(*channel, ...); 52 // 53 // // Start sending and playing RTP on voip channel. 54 // voip_base.StartSend(*channel); 55 // voip_base.StartPlayout(*channel); 56 // 57 // // Inject received RTP/RTCP through VoipNetwork interface. 58 // voip_network.ReceivedRTPPacket(*channel, ...); 59 // voip_network.ReceivedRTCPPacket(*channel, ...); 60 // 61 // // Stop and release voip channel. 62 // voip_base.StopSend(*channel); 63 // voip_base.StopPlayout(*channel); 64 // voip_base.ReleaseChannel(*channel); 65 // 66 // Current VoipEngine defines three sub-API classes and is subject to expand in 67 // near future. 68 class VoipEngine { 69 public: 70 virtual ~VoipEngine() = default; 71 72 // VoipBase is the audio session management interface that 73 // creates/releases/starts/stops an one-to-one audio media session. 74 virtual VoipBase& Base() = 0; 75 76 // VoipNetwork provides injection APIs that would enable application 77 // to send and receive RTP/RTCP packets. There is no default network module 78 // that provides RTP transmission and reception. 79 virtual VoipNetwork& Network() = 0; 80 81 // VoipCodec provides codec configuration APIs for encoder and decoders. 82 virtual VoipCodec& Codec() = 0; 83 }; 84 85 } // namespace webrtc 86 87 #endif // API_VOIP_VOIP_ENGINE_H_ 88