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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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 |
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33 return ssrc; | 33 return ssrc; |
34 } | 34 } |
35 } // namespace | 35 } // namespace |
36 | 36 |
37 namespace voetest { | 37 namespace voetest { |
38 | 38 |
39 ConferenceTransport::ConferenceTransport() | 39 ConferenceTransport::ConferenceTransport() |
40 : pq_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), | 40 : pq_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), |
41 stream_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), | 41 stream_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), |
42 packet_event_(webrtc::EventWrapper::Create()), | 42 packet_event_(webrtc::EventWrapper::Create()), |
43 thread_(webrtc::ThreadWrapper::CreateThread(Run, | 43 thread_(webrtc::PlatformThread::CreateThread(Run, |
44 this, | 44 this, |
45 "ConferenceTransport")), | 45 "ConferenceTransport")), |
46 rtt_ms_(0), | 46 rtt_ms_(0), |
47 stream_count_(0), | 47 stream_count_(0), |
48 rtp_header_parser_(webrtc::RtpHeaderParser::Create()) { | 48 rtp_header_parser_(webrtc::RtpHeaderParser::Create()) { |
49 rtp_header_parser_-> | 49 rtp_header_parser_-> |
50 RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, | 50 RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, |
51 kAudioLevelHeaderId); | 51 kAudioLevelHeaderId); |
52 | 52 |
53 local_voe_ = webrtc::VoiceEngine::Create(); | 53 local_voe_ = webrtc::VoiceEngine::Create(); |
54 local_base_ = webrtc::VoEBase::GetInterface(local_voe_); | 54 local_base_ = webrtc::VoEBase::GetInterface(local_voe_); |
55 local_network_ = webrtc::VoENetwork::GetInterface(local_voe_); | 55 local_network_ = webrtc::VoENetwork::GetInterface(local_voe_); |
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281 bool ConferenceTransport::GetReceiverStatistics(unsigned int id, | 281 bool ConferenceTransport::GetReceiverStatistics(unsigned int id, |
282 webrtc::CallStatistics* stats) { | 282 webrtc::CallStatistics* stats) { |
283 int dst = GetReceiverChannelForSsrc(id); | 283 int dst = GetReceiverChannelForSsrc(id); |
284 if (dst == -1) { | 284 if (dst == -1) { |
285 return false; | 285 return false; |
286 } | 286 } |
287 EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats)); | 287 EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats)); |
288 return true; | 288 return true; |
289 } | 289 } |
290 } // namespace voetest | 290 } // namespace voetest |
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