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Unified Diff: webrtc/modules/audio_device/dummy/file_audio_device.h

Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Resync Created 5 years, 4 months ago
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Index: webrtc/modules/audio_device/dummy/file_audio_device.h
diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.h b/webrtc/modules/audio_device/dummy/file_audio_device.h
index ffc8adc0162acb27df686d665001a6ba695bb263..91a7d22f64fcec1526cbd6c16a81995d7aa8f103 100644
--- a/webrtc/modules/audio_device/dummy/file_audio_device.h
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.h
@@ -174,9 +174,9 @@ class FileAudioDevice : public AudioDeviceGeneric {
uint32_t _playoutFramesLeft;
CriticalSectionWrapper& _critSect;
- uint32_t _recordingBufferSizeIn10MS;
- uint32_t _recordingFramesIn10MS;
- uint32_t _playoutFramesIn10MS;
+ size_t _recordingBufferSizeIn10MS;
+ size_t _recordingFramesIn10MS;
+ size_t _playoutFramesIn10MS;
rtc::scoped_ptr<ThreadWrapper> _ptrThreadRec;
rtc::scoped_ptr<ThreadWrapper> _ptrThreadPlay;
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