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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 #include "webrtc/test/channel_transport/channel_transport.h" | |
| 12 | |
| 13 #include <stdio.h> | |
| 14 | |
| 15 #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) | |
| 16 #include "testing/gtest/include/gtest/gtest.h" | |
| 17 #endif | |
| 18 #include "webrtc/test/channel_transport/udp_transport.h" | |
| 19 #include "webrtc/voice_engine/include/voe_network.h" | |
| 20 | |
| 21 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) | |
| 22 #undef NDEBUG | |
| 23 #include <assert.h> | |
| 24 #endif | |
| 25 | |
| 26 namespace webrtc { | |
| 27 namespace test { | |
| 28 | |
| 29 VoiceChannelTransport::VoiceChannelTransport(VoENetwork* voe_network, | |
| 30 int channel) | |
| 31 : channel_(channel), | |
| 32 voe_network_(voe_network) { | |
| 33 uint8_t socket_threads = 1; | |
| 34 socket_transport_ = UdpTransport::Create(channel, socket_threads); | |
| 35 int registered = voe_network_->RegisterExternalTransport(channel, | |
| 36 *socket_transport_); | |
| 37 #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) | |
| 38 EXPECT_EQ(0, registered); | |
| 39 #else | |
| 40 assert(registered == 0); | |
| 41 #endif | |
| 42 } | |
| 43 | |
| 44 VoiceChannelTransport::~VoiceChannelTransport() { | |
| 45 voe_network_->DeRegisterExternalTransport(channel_); | |
| 46 UdpTransport::Destroy(socket_transport_); | |
| 47 } | |
| 48 | |
| 49 void VoiceChannelTransport::IncomingRTPPacket( | |
| 50 const int8_t* incoming_rtp_packet, | |
| 51 const size_t packet_length, | |
| 52 const char* /*from_ip*/, | |
| 53 const uint16_t /*from_port*/) { | |
| 54 voe_network_->ReceivedRTPPacket( | |
| 55 channel_, incoming_rtp_packet, packet_length, PacketTime()); | |
| 56 } | |
| 57 | |
| 58 void VoiceChannelTransport::IncomingRTCPPacket( | |
| 59 const int8_t* incoming_rtcp_packet, | |
| 60 const size_t packet_length, | |
| 61 const char* /*from_ip*/, | |
| 62 const uint16_t /*from_port*/) { | |
| 63 voe_network_->ReceivedRTCPPacket(channel_, incoming_rtcp_packet, | |
| 64 packet_length); | |
| 65 } | |
| 66 | |
| 67 int VoiceChannelTransport::SetLocalReceiver(uint16_t rtp_port) { | |
| 68 static const int kNumReceiveSocketBuffers = 500; | |
| 69 int return_value = socket_transport_->InitializeReceiveSockets(this, | |
| 70 rtp_port); | |
| 71 if (return_value == 0) { | |
| 72 return socket_transport_->StartReceiving(kNumReceiveSocketBuffers); | |
| 73 } | |
| 74 return return_value; | |
| 75 } | |
| 76 | |
| 77 int VoiceChannelTransport::SetSendDestination(const char* ip_address, | |
| 78 uint16_t rtp_port) { | |
| 79 return socket_transport_->InitializeSendSockets(ip_address, rtp_port); | |
| 80 } | |
| 81 | |
| 82 } // namespace test | |
| 83 } // namespace webrtc | |
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