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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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