| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| locked_bandwidth_info.cc | 16 : lock_(CriticalSectionWrapper::CreateCriticalSection()) {
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| /external/webrtc/webrtc/system_wrappers/source/ |
| critical_section.cc | 20 CriticalSectionWrapper* CriticalSectionWrapper::CreateCriticalSection() {
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| condition_variable_unittest.cc | 38 : giver_sect_(CriticalSectionWrapper::CreateCriticalSection()),
39 crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
190 CriticalSectionWrapper::CreateCriticalSection());
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| rw_lock_generic.cc | 23 critical_section_ = CriticalSectionWrapper::CreateCriticalSection();
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| critical_section_unittest.cc | 79 CriticalSectionWrapper::CreateCriticalSection();
106 CriticalSectionWrapper::CreateCriticalSection();
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| logging_unittest.cc | 40 : crit_(CriticalSectionWrapper::CreateCriticalSection()),
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| trace_posix.cc | 23 : crit_sect_(*CriticalSectionWrapper::CreateCriticalSection()) {
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| data_log.cc | 26 CriticalSectionWrapper::CreateCriticalSection());
108 cells_lock_(CriticalSectionWrapper::CreateCriticalSection()) {
149 table_lock_(CriticalSectionWrapper::CreateCriticalSection()) {
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| /external/webrtc/webrtc/system_wrappers/include/ |
| critical_section_wrapper.h | 24 static CriticalSectionWrapper* CreateCriticalSection();
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| static_instance.h | 49 CriticalSectionWrapper::CreateCriticalSection());
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_receiver_strategy.cc | 20 : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
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| ssrc_database.cc | 55 : crit_(CriticalSectionWrapper::CreateCriticalSection()), random_(Seed()) {}
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| dtmf_queue.cc | 17 : dtmf_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
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| rtp_header_parser.cc | 42 : critical_section_(CriticalSectionWrapper::CreateCriticalSection()) {}
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| /external/webrtc/webrtc/voice_engine/ |
| level_indicator.cc | 28 _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
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| monitor_module.cc | 21 _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
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| dtmf_inband_queue.cc | 18 _DtmfCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
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| statistics.cc | 24 _critPtr(CriticalSectionWrapper::CreateCriticalSection()),
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| shared_data.cc | 28 _apiCritPtr(CriticalSectionWrapper::CreateCriticalSection()),
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| /external/webrtc/webrtc/video/ |
| video_capture_input.cc | 38 : capture_cs_(CriticalSectionWrapper::CreateCriticalSection()),
43 incoming_frame_cs_(CriticalSectionWrapper::CreateCriticalSection()),
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| payload_router.cc | 21 : crit_(CriticalSectionWrapper::CreateCriticalSection()),
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| /external/webrtc/webrtc/modules/audio_conference_mixer/source/ |
| time_scheduler.cc | 16 : _crit(CriticalSectionWrapper::CreateCriticalSection()),
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| /external/webrtc/webrtc/modules/video_coding/codecs/test/ |
| packet_manipulator.cc | 27 critsect_(CriticalSectionWrapper::CreateCriticalSection()),
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| /external/webrtc/webrtc/voice_engine/test/auto_test/standard/ |
| rtp_rtcp_test.cc | 21 : crit_(voetest::CriticalSectionWrapper::CreateCriticalSection()),
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| /external/webrtc/webrtc/common_video/ |
| incoming_video_stream.cc | 39 stream_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
40 thread_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
41 buffer_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
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