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Side by Side Diff: talk/media/webrtc/webrtcvoiceengine.cc

Issue 1666843003: Bugfix to stop chrome from crashing. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added namespace. Created 4 years, 10 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2004 Google Inc. 3 * Copyright 2004 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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1146 if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() != 1146 if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() !=
1147 webrtc::AudioProcessing::kNoError) { 1147 webrtc::AudioProcessing::kNoError) {
1148 LOG_RTCERR0(StopDebugRecording); 1148 LOG_RTCERR0(StopDebugRecording);
1149 } 1149 }
1150 is_dumping_aec_ = false; 1150 is_dumping_aec_ = false;
1151 } 1151 }
1152 } 1152 }
1153 1153
1154 bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) { 1154 bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) {
1155 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1155 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1156 return voe_wrapper_->codec()->GetEventLog()->StartLogging(file); 1156 webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
1157 if (event_log) {
1158 return event_log->StartLogging(file);
1159 }
1160 LOG_RTCERR0(StartRtcEventLog);
1161 return false;
1157 } 1162 }
1158 1163
1159 void WebRtcVoiceEngine::StopRtcEventLog() { 1164 void WebRtcVoiceEngine::StopRtcEventLog() {
1160 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1165 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1161 voe_wrapper_->codec()->GetEventLog()->StopLogging(); 1166 webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
1167 if (event_log) {
1168 event_log->StopLogging();
1169 return;
1170 }
1171 LOG_RTCERR0(StopRtcEventLog);
1162 } 1172 }
1163 1173
1164 int WebRtcVoiceEngine::CreateVoEChannel() { 1174 int WebRtcVoiceEngine::CreateVoEChannel() {
1165 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); 1175 RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
1166 return voe_wrapper_->base()->CreateChannel(voe_config_); 1176 return voe_wrapper_->base()->CreateChannel(voe_config_);
1167 } 1177 }
1168 1178
1169 class WebRtcVoiceMediaChannel::WebRtcAudioSendStream 1179 class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
1170 : public AudioRenderer::Sink { 1180 : public AudioRenderer::Sink {
1171 public: 1181 public:
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2560 } 2570 }
2561 } else { 2571 } else {
2562 LOG(LS_INFO) << "Stopping playout for channel #" << channel; 2572 LOG(LS_INFO) << "Stopping playout for channel #" << channel;
2563 engine()->voe()->base()->StopPlayout(channel); 2573 engine()->voe()->base()->StopPlayout(channel);
2564 } 2574 }
2565 return true; 2575 return true;
2566 } 2576 }
2567 } // namespace cricket 2577 } // namespace cricket
2568 2578
2569 #endif // HAVE_WEBRTC_VOICE 2579 #endif // HAVE_WEBRTC_VOICE
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