ccefde95b3
Bug: webrtc:11251 Change-Id: Iecde33b86856b14db5abade3301a842d5007568d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169034 Commit-Queue: Tim Na <natim@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30675}
78 lines
2.5 KiB
C++
78 lines
2.5 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 interfaces
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//
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// These pointer interfaces are valid as long as VoipEngine is available.
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// Therefore, application must synchronize the usage within the life span of
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// created VoipEngine instance.
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//
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// auto voip_engine =
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// webrtc::VoipEngineBuilder()
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// .SetAudioEncoderFactory(CreateBuiltinAudioEncoderFactory())
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// .SetAudioDecoderFactory(CreateBuiltinAudioDecoderFactory())
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// .Create();
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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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// VoipChannel::Config config = { &app_transport_, 0xdeadc0de };
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// int channel = voip_base.CreateChannel(config);
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//
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// // After SDP offer/answer, payload type and codec usage have been
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// // decided through 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 Send/Playout 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 thru voip network interface.
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// voip_network.ReceivedRTPPacket(channel, rtp_data, rtp_size);
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// voip_network.ReceivedRTCPPacket(channel, rtcp_data, rtcp_size);
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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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//
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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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// create/release/start/stop 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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