| Index: webrtc/examples/unityplugin/unity_plugin_apis.h
|
| diff --git a/webrtc/examples/unityplugin/unity_plugin_apis.h b/webrtc/examples/unityplugin/unity_plugin_apis.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..bcd1af362d9fef6b3eeeada2deaa0ae7ff8353b8
|
| --- /dev/null
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| +++ b/webrtc/examples/unityplugin/unity_plugin_apis.h
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| @@ -0,0 +1,83 @@
|
| +/*
|
| + * Copyright (c) 2017 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.
|
| + */
|
| +
|
| +// This file provides an example of unity native plugin APIs.
|
| +
|
| +#ifndef WEBRTC_EXAMPLES_UNITYPLUGIN_UNITY_PLUGIN_APIS_H_
|
| +#define WEBRTC_EXAMPLES_UNITYPLUGIN_UNITY_PLUGIN_APIS_H_
|
| +
|
| +#include <stdint.h>
|
| +
|
| +// Defintions of callback functions.
|
| +typedef void (*VIDEOFRAMEREADY_CALLBACK)(uint8_t* buffer,
|
| + uint32_t width,
|
| + uint32_t height,
|
| + uint32_t stride);
|
| +typedef void (*LOCALDATACHANNELREADY_CALLBACK)();
|
| +typedef void (*DATAFROMEDATECHANNELREADY_CALLBACK)(const char* msg);
|
| +typedef void (*FAILURE_CALLBACK)(const char* msg);
|
| +typedef void (*LOCALSDPREADYTOSEND_CALLBACK)(const char* msg);
|
| +typedef void (*ICECANDIDATEREADYTOSEND_CALLBACK)(const char* msg);
|
| +typedef void (*AUDIOBUSREADY_CALLBACK)(const void* audio_data,
|
| + int bits_per_sample,
|
| + int sample_rate,
|
| + int number_of_channels,
|
| + int number_of_frames);
|
| +
|
| +#define WEBRTC_PLUGIN_API __declspec(dllexport)
|
| +extern "C" {
|
| +// Create a peerconnection and return a unique peer connection id.
|
| +WEBRTC_PLUGIN_API int CreatePeerConnection();
|
| +// Close a peerconnection.
|
| +WEBRTC_PLUGIN_API bool ClosePeerConnection(int peer_connection_id);
|
| +// Add a audio stream. If audio_only is true, the stream only has an audio
|
| +// track and no video track.
|
| +WEBRTC_PLUGIN_API bool AddStream(int peer_connection_id, bool audio_only);
|
| +// Add a data channel to peer connection.
|
| +WEBRTC_PLUGIN_API bool AddDataChannel(int peer_connection_id);
|
| +// Create a peer connection offer.
|
| +WEBRTC_PLUGIN_API bool CreateOffer(int peer_connection_id);
|
| +// Create a peer connection answer.
|
| +WEBRTC_PLUGIN_API bool CreateAnswer(int peer_connection_id);
|
| +// Send data through data channel.
|
| +WEBRTC_PLUGIN_API bool SendDataViaDataChannel(int peer_connection_id,
|
| + const char* data);
|
| +// Set audio control. If is_mute=true, no audio will playout. If is_record=true,
|
| +// AUDIOBUSREADY_CALLBACK will be called every 10 ms.
|
| +WEBRTC_PLUGIN_API bool SetAudioControl(int peer_connection_id,
|
| + bool is_mute,
|
| + bool is_record);
|
| +
|
| +// Register callback functions.
|
| +WEBRTC_PLUGIN_API bool RegisterOnVideoFramReady(
|
| + int peer_connection_id,
|
| + VIDEOFRAMEREADY_CALLBACK callback);
|
| +WEBRTC_PLUGIN_API bool RegisterOnLocalDataChannelReady(
|
| + int peer_connection_id,
|
| + LOCALDATACHANNELREADY_CALLBACK callback);
|
| +WEBRTC_PLUGIN_API bool RegisterOnDataFromDataChannelReady(
|
| + int peer_connection_id,
|
| + DATAFROMEDATECHANNELREADY_CALLBACK callback);
|
| +WEBRTC_PLUGIN_API bool RegisterOnFailure(int peer_connection_id,
|
| + FAILURE_CALLBACK callback);
|
| +WEBRTC_PLUGIN_API bool RegisterOnAudioBusReady(int peer_connection_id,
|
| + AUDIOBUSREADY_CALLBACK callback);
|
| +WEBRTC_PLUGIN_API bool RegisterOnLocalSdpReadytoSend(
|
| + int peer_connection_id,
|
| + LOCALSDPREADYTOSEND_CALLBACK callback);
|
| +WEBRTC_PLUGIN_API bool RegisterOnIceCandiateReadytoSend(
|
| + int peer_connection_id,
|
| + ICECANDIDATEREADYTOSEND_CALLBACK callback);
|
| +WEBRTC_PLUGIN_API int ReceivedSdp(int peer_connection_id, const char* sdp);
|
| +WEBRTC_PLUGIN_API bool ReceivedIceCandidate(int peer_connection_id,
|
| + const char* ice_candidate);
|
| +}
|
| +
|
| +#endif // WEBRTC_EXAMPLES_UNITYPLUGIN_UNITY_PLUGIN_APIS_H_
|
|
|