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Side by Side Diff: webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc

Issue 1469013002: Move ThreadWrapper to ProcessThread in base. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: removed comment Created 5 years ago
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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
(...skipping 22 matching lines...) Expand all
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_);
(...skipping 225 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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