| /external/webrtc/modules/audio_device/android/ |
| D | audio_record_jni.cc | 70 bool AudioRecordJni::JavaAudioRecord::StopRecording() { in StopRecording() function in webrtc::AudioRecordJni::JavaAudioRecord
177 int32_t AudioRecordJni::StopRecording() { in StopRecording() function in webrtc::AudioRecordJni
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| D | aaudio_recorder.cc | 88 int AAudioRecorder::StopRecording() { in StopRecording() function in webrtc::AAudioRecorder
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| D | opensles_recorder.cc | 147 int OpenSLESRecorder::StopRecording() { in StopRecording() function in webrtc::OpenSLESRecorder
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| D | audio_device_template.h | 207 int32_t StopRecording() override { in StopRecording() function
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| /external/webrtc/sdk/android/src/jni/audio_device/ |
| D | aaudio_recorder.cc | 94 int AAudioRecorder::StopRecording() { in StopRecording() function in webrtc::jni::AAudioRecorder
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| D | audio_record_jni.cc | 155 int32_t AudioRecordJni::StopRecording() { in StopRecording() function in webrtc::jni::AudioRecordJni
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| D | opensles_recorder.cc | 153 int OpenSLESRecorder::StopRecording() { in StopRecording() function in webrtc::jni::OpenSLESRecorder
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| D | audio_device_module.cc | 301 int32_t StopRecording() override { in StopRecording() function in webrtc::jni::__anon095936e00111::AndroidAudioDeviceModule
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| /external/webrtc/modules/audio_device/win/ |
| D | core_audio_input_win.cc | 181 int CoreAudioInput::StopRecording() { in StopRecording() function in webrtc::webrtc_win::CoreAudioInput
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| D | audio_device_module_win.cc | 324 int32_t StopRecording() override { in StopRecording() function in webrtc::webrtc_win::__anon3803f92a0111::WindowsAudioDeviceModule
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| /external/webrtc/modules/audio_device/ |
| D | audio_device_buffer.cc | 149 void AudioDeviceBuffer::StopRecording() { in StopRecording() function in webrtc::AudioDeviceBuffer
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| D | audio_device_impl.cc | 832 int32_t AudioDeviceModuleImpl::StopRecording() { in StopRecording() function in webrtc::AudioDeviceModuleImpl
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| D | audio_device_data_observer.cc | 181 int32_t StopRecording() override { return impl_->StopRecording(); } in StopRecording() function in webrtc::__anon6f60d37b0111::ADMWrapper
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| D | audio_device_unittest.cc | 637 void StopRecording() { in StopRecording() function in webrtc::MAYBE_AudioDeviceTest
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| /external/webrtc/modules/audio_device/dummy/ |
| D | audio_device_dummy.cc | 110 int32_t AudioDeviceDummy::StopRecording() { in StopRecording() function in webrtc::AudioDeviceDummy
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| D | file_audio_device.cc | 290 int32_t FileAudioDevice::StopRecording() { in StopRecording() function in webrtc::FileAudioDevice
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| /external/ruy/ruy/ |
| D | pmu.cc | 233 void PmuEvents::StopRecording() { in StopRecording() function in ruy::PmuEvents
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| /external/webrtc/modules/audio_device/include/ |
| D | audio_device_default.h | 47 int32_t StopRecording() override { return 0; } in StopRecording() function
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| D | test_audio_device.cc | 128 int32_t StopRecording() override { in StopRecording() function in webrtc::__anon9ec0e82d0111::TestAudioDeviceModuleImpl
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| /external/tensorflow/tensorflow/core/profiler/internal/cpu/ |
| D | traceme_recorder.cc | 385 TraceMeRecorder::Events TraceMeRecorder::StopRecording() { in StopRecording() function in tensorflow::profiler::TraceMeRecorder
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| /external/protobuf/src/google/protobuf/io/ |
| D | tokenizer.cc | 286 inline void Tokenizer::StopRecording() { in StopRecording() function in google::protobuf::io::Tokenizer
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| /external/webrtc/pc/test/ |
| D | fake_audio_capture_module.cc | 223 int32_t FakeAudioCaptureModule::StopRecording() { in StopRecording() function in FakeAudioCaptureModule
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| /external/webrtc/modules/audio_processing/test/ |
| D | debug_dump_test.cc | 231 void DebugDumpGenerator::StopRecording() { in StopRecording() function in webrtc::test::__anon9b6362310111::DebugDumpGenerator
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| /external/webrtc/modules/audio_device/linux/ |
| D | audio_device_alsa_linux.cc | 1060 int32_t AudioDeviceLinuxALSA::StopRecording() { in StopRecording() function in webrtc::AudioDeviceLinuxALSA
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| D | audio_device_pulse_linux.cc | 1091 int32_t AudioDeviceLinuxPulse::StopRecording() { in StopRecording() function in webrtc::AudioDeviceLinuxPulse
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