88 lines
3.3 KiB
C++
88 lines
3.3 KiB
C++
/*
|
|
* 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;
|
|
|
|
// 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 mesh conferencing. 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));
|
|
// if (!voip_engine) return some_failure;
|
|
//
|
|
// auto& voip_base = voip_engine->Base();
|
|
// auto& voip_codec = voip_engine->Codec();
|
|
// auto& voip_network = voip_engine->Network();
|
|
//
|
|
// absl::optional<ChannelId> channel =
|
|
// voip_base.CreateChannel(&app_transport_);
|
|
// if (!channel) return some_failure;
|
|
//
|
|
// // After SDP offer/answer, set payload type and codecs that have been
|
|
// // decided through SDP negotiation.
|
|
// voip_codec.SetSendCodec(*channel, ...);
|
|
// voip_codec.SetReceiveCodecs(*channel, ...);
|
|
//
|
|
// // Start sending and playing RTP on voip channel.
|
|
// voip_base.StartSend(*channel);
|
|
// voip_base.StartPlayout(*channel);
|
|
//
|
|
// // Inject received RTP/RTCP through VoipNetwork interface.
|
|
// voip_network.ReceivedRTPPacket(*channel, ...);
|
|
// voip_network.ReceivedRTCPPacket(*channel, ...);
|
|
//
|
|
// // Stop and release voip channel.
|
|
// voip_base.StopSend(*channel);
|
|
// voip_base.StopPlayout(*channel);
|
|
// voip_base.ReleaseChannel(*channel);
|
|
//
|
|
// Current VoipEngine defines three sub-API classes and is subject to expand in
|
|
// near future.
|
|
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;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // API_VOIP_VOIP_ENGINE_H_
|