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| -rw-r--r-- | third_party/libwebrtc/api/voip/voip_engine.h | 99 |
1 files changed, 99 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/voip/voip_engine.h b/third_party/libwebrtc/api/voip/voip_engine.h new file mode 100644 index 0000000000..d223f6ad6c --- /dev/null +++ b/third_party/libwebrtc/api/voip/voip_engine.h @@ -0,0 +1,99 @@ +/* + * 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_ |