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Side by Side Diff: webrtc/test/channel_transport/channel_transport.cc

Issue 1411573007: Removed vie_defines.h (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 1 month ago
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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
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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