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c0df5fc25b
This is one last CL that includes the rest of VoIP API implementation. Bug: webrtc:11251 Change-Id: I3f1b0bf2fd48be864ffc73482105f9514f75f9e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173860 Commit-Queue: Tim Na <natim@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31168}
87 lines
3.3 KiB
C++
87 lines
3.3 KiB
C++
/*
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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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// 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 mesh conferencing. 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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// if (!voip_engine) return some_failure;
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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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// absl::optional<ChannelId> channel =
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// voip_base.CreateChannel(&app_transport_);
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// if (!channel) return some_failure;
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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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// 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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// 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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// 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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// 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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// Current VoipEngine defines three sub-API classes and is subject to expand in
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// near future.
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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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};
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} // namespace webrtc
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#endif // API_VOIP_VOIP_ENGINE_H_
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