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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #include "webrtc/test/channel_transport/include/channel_transport.h" | 11 #include "webrtc/test/channel_transport/include/channel_transport.h" |
12 | 12 |
13 #include <stdio.h> | 13 #include <stdio.h> |
14 | 14 |
15 #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) | 15 #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) |
16 #include "testing/gtest/include/gtest/gtest.h" | 16 #include "testing/gtest/include/gtest/gtest.h" |
17 #endif | 17 #endif |
18 #include "webrtc/test/channel_transport/udp_transport.h" | 18 #include "webrtc/test/channel_transport/udp_transport.h" |
19 #include "webrtc/video_engine/vie_defines.h" | |
20 #include "webrtc/voice_engine/include/voe_network.h" | 19 #include "webrtc/voice_engine/include/voe_network.h" |
21 | 20 |
22 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) | 21 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) |
23 #undef NDEBUG | 22 #undef NDEBUG |
24 #include <assert.h> | 23 #include <assert.h> |
25 #endif | 24 #endif |
26 | 25 |
27 namespace webrtc { | 26 namespace webrtc { |
28 namespace test { | 27 namespace test { |
29 | 28 |
(...skipping 29 matching lines...) Expand all Loading... | |
59 void VoiceChannelTransport::IncomingRTCPPacket( | 58 void VoiceChannelTransport::IncomingRTCPPacket( |
60 const int8_t* incoming_rtcp_packet, | 59 const int8_t* incoming_rtcp_packet, |
61 const size_t packet_length, | 60 const size_t packet_length, |
62 const char* /*from_ip*/, | 61 const char* /*from_ip*/, |
63 const uint16_t /*from_port*/) { | 62 const uint16_t /*from_port*/) { |
64 voe_network_->ReceivedRTCPPacket(channel_, incoming_rtcp_packet, | 63 voe_network_->ReceivedRTCPPacket(channel_, incoming_rtcp_packet, |
65 packet_length); | 64 packet_length); |
66 } | 65 } |
67 | 66 |
68 int VoiceChannelTransport::SetLocalReceiver(uint16_t rtp_port) { | 67 int VoiceChannelTransport::SetLocalReceiver(uint16_t rtp_port) { |
68 const int kNumReceiveSocketBuffers = 500; | |
pbos-webrtc
2015/10/29 15:39:40
static
mflodman
2015/11/06 11:38:36
Done.
| |
69 int return_value = socket_transport_->InitializeReceiveSockets(this, | 69 int return_value = socket_transport_->InitializeReceiveSockets(this, |
70 rtp_port); | 70 rtp_port); |
71 if (return_value == 0) { | 71 if (return_value == 0) { |
72 return socket_transport_->StartReceiving(kViENumReceiveSocketBuffers); | 72 return socket_transport_->StartReceiving(kNumReceiveSocketBuffers); |
73 } | 73 } |
74 return return_value; | 74 return return_value; |
75 } | 75 } |
76 | 76 |
77 int VoiceChannelTransport::SetSendDestination(const char* ip_address, | 77 int VoiceChannelTransport::SetSendDestination(const char* ip_address, |
78 uint16_t rtp_port) { | 78 uint16_t rtp_port) { |
79 return socket_transport_->InitializeSendSockets(ip_address, rtp_port); | 79 return socket_transport_->InitializeSendSockets(ip_address, rtp_port); |
80 } | 80 } |
81 | 81 |
82 } // namespace test | 82 } // namespace test |
83 } // namespace webrtc | 83 } // namespace webrtc |
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