| /* |
| * Copyright (c) 2011 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. |
| */ |
| |
| // ViESender is responsible for encrypting, if enabled, packets and send to |
| // network. |
| |
| #ifndef WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_ |
| #define WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_ |
| |
| #include "common_types.h" |
| #include "engine_configurations.h" |
| #include "system_wrappers/interface/scoped_ptr.h" |
| #include "typedefs.h" |
| #include "vie_defines.h" |
| |
| namespace webrtc { |
| |
| class CriticalSectionWrapper; |
| class RtpDump; |
| class Transport; |
| class VideoCodingModule; |
| |
| class ViESender: public Transport { |
| public: |
| ViESender(int engine_id, int channel_id); |
| ~ViESender(); |
| |
| // Registers an encryption class to use before sending packets. |
| int RegisterExternalEncryption(Encryption* encryption); |
| int DeregisterExternalEncryption(); |
| |
| // Registers transport to use for sending RTP and RTCP. |
| int RegisterSendTransport(Transport* transport); |
| int DeregisterSendTransport(); |
| |
| // Stores all incoming packets to file. |
| int StartRTPDump(const char file_nameUTF8[1024]); |
| int StopRTPDump(); |
| |
| // Implements Transport. |
| virtual int SendPacket(int vie_id, const void* data, int len); |
| virtual int SendRTCPPacket(int vie_id, const void* data, int len); |
| |
| private: |
| int engine_id_; |
| int channel_id_; |
| |
| scoped_ptr<CriticalSectionWrapper> critsect_; |
| |
| Encryption* external_encryption_; |
| WebRtc_UWord8* encryption_buffer_; |
| Transport* transport_; |
| RtpDump* rtp_dump_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_ |