c63bf10790
Bug: webrtc:11251 Change-Id: I26b6915d3ad6bb5a50f9898a6866889867fd53f5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169000 Commit-Queue: Tim Na <natim@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30594}
80 lines
2.6 KiB
C++
80 lines
2.6 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_
|
|
|
|
#include <memory>
|
|
|
|
#include "api/voip/voip_base.h"
|
|
#include "api/voip/voip_codec.h"
|
|
#include "api/voip/voip_network.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// VoipEngine interfaces
|
|
//
|
|
// These pointer interfaces are valid as long as VoipEngine is available.
|
|
// Therefore, application must synchronize the usage within the life span of
|
|
// created VoipEngine instance.
|
|
//
|
|
// auto voip_engine =
|
|
// webrtc::VoipEngineBuilder()
|
|
// .SetAudioEncoderFactory(CreateBuiltinAudioEncoderFactory())
|
|
// .SetAudioDecoderFactory(CreateBuiltinAudioDecoderFactory())
|
|
// .Create();
|
|
//
|
|
// auto* voip_base = voip_engine->Base();
|
|
// auto* voip_codec = voip_engine->Codec();
|
|
// auto* voip_network = voip_engine->Network();
|
|
//
|
|
// VoipChannel::Config config = { &app_transport_, 0xdeadc0de };
|
|
// int channel = voip_base->CreateChannel(config);
|
|
//
|
|
// // After SDP offer/answer, payload type and codec usage have been
|
|
// // decided through negotiation.
|
|
// voip_codec->SetSendCodec(channel, ...);
|
|
// voip_codec->SetReceiveCodecs(channel, ...);
|
|
//
|
|
// // Start Send/Playout on voip channel.
|
|
// voip_base->StartSend(channel);
|
|
// voip_base->StartPlayout(channel);
|
|
//
|
|
// // Inject received rtp/rtcp thru voip network interface.
|
|
// voip_network->ReceivedRTPPacket(channel, rtp_data, rtp_size);
|
|
// voip_network->ReceivedRTCPPacket(channel, rtcp_data, rtcp_size);
|
|
//
|
|
// // Stop and release voip channel.
|
|
// voip_base->StopSend(channel);
|
|
// voip_base->StopPlayout(channel);
|
|
//
|
|
// voip_base->ReleaseChannel(channel);
|
|
//
|
|
class VoipEngine {
|
|
public:
|
|
// VoipBase is the audio session management interface that
|
|
// create/release/start/stop 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;
|
|
|
|
virtual ~VoipEngine() = default;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // API_VOIP_VOIP_ENGINE_H_
|