blob: a60446dbdccccae22ee82d974b5b0496dca447a7 [file] [log] [blame]
* 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.
#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 {
ViESender(int engine_id, int channel_id);
// 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);
int engine_id_;
int channel_id_;
scoped_ptr<CriticalSectionWrapper> critsect_;
Encryption* external_encryption_;
WebRtc_UWord8* encryption_buffer_;
Transport* transport_;
RtpDump* rtp_dump_;
} // namespace webrtc