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Unified Diff: webrtc/modules/audio_device/android/audio_record_jni.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/android/audio_record_jni.h
diff --git a/webrtc/modules/audio_device/android/audio_record_jni.h b/webrtc/modules/audio_device/android/audio_record_jni.h
index 1a2bd9de22280939861392ce5c83417aafc552d0..6a17eb3059a47d1ee712811419177442471a95e9 100644
--- a/webrtc/modules/audio_device/android/audio_record_jni.h
+++ b/webrtc/modules/audio_device/android/audio_record_jni.h
@@ -135,13 +135,13 @@ class AudioRecordJni {
void* direct_buffer_address_;
// Number of bytes in the direct audio buffer owned by |j_audio_record_|.
- int direct_buffer_capacity_in_bytes_;
+ size_t direct_buffer_capacity_in_bytes_;
// Number audio frames per audio buffer. Each audio frame corresponds to
// one sample of PCM mono data at 16 bits per sample. Hence, each audio
// frame contains 2 bytes (given that the Java layer only supports mono).
// Example: 480 for 48000 Hz or 441 for 44100 Hz.
- int frames_per_buffer_;
+ size_t frames_per_buffer_;
bool initialized_;
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