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| 1 /* | |
| 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #ifndef WEBRTC_VOICE_ENGINE_TEST_CHANNEL_TRANSPORT_CHANNEL_TRANSPORT_H_ | |
| 12 #define WEBRTC_VOICE_ENGINE_TEST_CHANNEL_TRANSPORT_CHANNEL_TRANSPORT_H_ | |
| 13 | |
| 14 #include "webrtc/voice_engine/test/channel_transport/udp_transport.h" | |
| 15 | |
| 16 namespace webrtc { | |
| 17 | |
| 18 class VoENetwork; | |
| 19 | |
| 20 namespace test { | |
| 21 | |
| 22 // Helper class for VoiceEngine tests. | |
| 23 class VoiceChannelTransport : public UdpTransportData { | |
| 24 public: | |
| 25 VoiceChannelTransport(VoENetwork* voe_network, int channel); | |
| 26 | |
| 27 virtual ~VoiceChannelTransport(); | |
| 28 | |
| 29 // Start implementation of UdpTransportData. | |
| 30 void IncomingRTPPacket(const int8_t* incoming_rtp_packet, | |
| 31 const size_t packet_length, | |
| 32 const char* /*from_ip*/, | |
| 33 const uint16_t /*from_port*/) override; | |
| 34 | |
| 35 void IncomingRTCPPacket(const int8_t* incoming_rtcp_packet, | |
| 36 const size_t packet_length, | |
| 37 const char* /*from_ip*/, | |
| 38 const uint16_t /*from_port*/) override; | |
| 39 // End implementation of UdpTransportData. | |
| 40 | |
| 41 // Specifies the ports to receive RTP packets on. | |
| 42 int SetLocalReceiver(uint16_t rtp_port); | |
| 43 | |
| 44 // Specifies the destination port and IP address for a specified channel. | |
| 45 int SetSendDestination(const char* ip_address, uint16_t rtp_port); | |
| 46 | |
| 47 private: | |
| 48 int channel_; | |
| 49 VoENetwork* voe_network_; | |
| 50 UdpTransport* socket_transport_; | |
| 51 }; | |
| 52 | |
| 53 } // namespace test | |
| 54 } // namespace webrtc | |
| 55 | |
| 56 #endif // WEBRTC_VOICE_ENGINE_TEST_CHANNEL_TRANSPORT_CHANNEL_TRANSPORT_H_ | |
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