webrtc/modules/audio_processing/aec/aec_resampler.h - Issue 1227213002: Update audio code to use size_t more correctly, webrtc/modules/audio_processing/

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Unified Diff: webrtc/modules/audio_processing/aec/aec_resampler.h

Issue 1227213002: Update audio code to use size_t more correctly, webrtc/modules/audio_processing/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master

Patch Set: Resync Created 5 years, 5 months ago

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Index: webrtc/modules/audio_processing/aec/aec_resampler.h

diff --git a/webrtc/modules/audio_processing/aec/aec_resampler.h b/webrtc/modules/audio_processing/aec/aec_resampler.h

index a37499258f317217a94806658aac5fc3ce5af5da..a5002c155a4a5fff6f338223f01aef4dc2aa8593 100644

--- a/webrtc/modules/audio_processing/aec/aec_resampler.h

+++ b/webrtc/modules/audio_processing/aec/aec_resampler.h

@@ -31,9 +31,9 @@ int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst);

// Resamples input using linear interpolation.

void WebRtcAec_ResampleLinear(void* resampInst,

const float* inspeech,

- int size,

+ size_t size,

float skew,

float* outspeech,

- int* size_out);

+ size_t* size_out);

#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_

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