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Unified Diff: webrtc/common_audio/signal_processing/filter_ar_fast_q12.c

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/common_audio/signal_processing/filter_ar_fast_q12.c
diff --git a/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c b/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
index cfd82ca8cf3e7020b44b3596bdb8604be0425adb..70001a088272ab450501e7d49637d27f41508f40 100644
--- a/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
+++ b/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
@@ -16,10 +16,10 @@
void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
int16_t* data_out,
const int16_t* __restrict coefficients,
- int coefficients_length,
- int data_length) {
- int i = 0;
- int j = 0;
+ size_t coefficients_length,
+ size_t data_length) {
+ size_t i = 0;
+ size_t j = 0;
assert(data_length > 0);
assert(coefficients_length > 1);
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