100 lines
3.6 KiB
C
100 lines
3.6 KiB
C
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/*
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* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_VOIP_VOIP_ENGINE_H_
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#define API_VOIP_VOIP_ENGINE_H_
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namespace webrtc {
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class VoipBase;
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class VoipCodec;
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class VoipNetwork;
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class VoipDtmf;
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class VoipStatistics;
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class VoipVolumeControl;
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// VoipEngine is the main interface serving as the entry point for all VoIP
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// APIs. A single instance of VoipEngine should suffice the most of the need for
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// typical VoIP applications as it handles multiple media sessions including a
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// specialized session type like ad-hoc conference. Below example code
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// describes the typical sequence of API usage. Each API header contains more
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// description on what the methods are used for.
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//
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// // Caller is responsible of setting desired audio components.
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// VoipEngineConfig config;
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// config.encoder_factory = CreateBuiltinAudioEncoderFactory();
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// config.decoder_factory = CreateBuiltinAudioDecoderFactory();
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// config.task_queue_factory = CreateDefaultTaskQueueFactory();
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// config.audio_device =
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// AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio,
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// config.task_queue_factory.get());
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// config.audio_processing = AudioProcessingBuilder().Create();
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//
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// auto voip_engine = CreateVoipEngine(std::move(config));
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//
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// auto& voip_base = voip_engine->Base();
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// auto& voip_codec = voip_engine->Codec();
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// auto& voip_network = voip_engine->Network();
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//
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// ChannelId channel = voip_base.CreateChannel(&app_transport_);
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//
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// // After SDP offer/answer, set payload type and codecs that have been
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// // decided through SDP negotiation.
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// // VoipResult handling omitted here.
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// voip_codec.SetSendCodec(channel, ...);
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// voip_codec.SetReceiveCodecs(channel, ...);
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//
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// // Start sending and playing RTP on voip channel.
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// // VoipResult handling omitted here.
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// voip_base.StartSend(channel);
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// voip_base.StartPlayout(channel);
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//
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// // Inject received RTP/RTCP through VoipNetwork interface.
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// // VoipResult handling omitted here.
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// voip_network.ReceivedRTPPacket(channel, ...);
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// voip_network.ReceivedRTCPPacket(channel, ...);
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//
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// // Stop and release voip channel.
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// // VoipResult handling omitted here.
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// voip_base.StopSend(channel);
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// voip_base.StopPlayout(channel);
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// voip_base.ReleaseChannel(channel);
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//
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class VoipEngine {
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public:
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virtual ~VoipEngine() = default;
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// VoipBase is the audio session management interface that
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// creates/releases/starts/stops an one-to-one audio media session.
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virtual VoipBase& Base() = 0;
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// VoipNetwork provides injection APIs that would enable application
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// to send and receive RTP/RTCP packets. There is no default network module
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// that provides RTP transmission and reception.
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virtual VoipNetwork& Network() = 0;
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// VoipCodec provides codec configuration APIs for encoder and decoders.
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virtual VoipCodec& Codec() = 0;
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// VoipDtmf provides DTMF event APIs to register and send DTMF events.
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virtual VoipDtmf& Dtmf() = 0;
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// VoipStatistics provides performance metrics around audio decoding module
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// and jitter buffer (NetEq).
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virtual VoipStatistics& Statistics() = 0;
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// VoipVolumeControl provides various input/output volume control.
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virtual VoipVolumeControl& VolumeControl() = 0;
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};
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} // namespace webrtc
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#endif // API_VOIP_VOIP_ENGINE_H_
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