Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 4a1b3e3a69d349b0d3e91f607f24e02d8b975688

This commit was generated by merge_from_chromium.py.

Change-Id: Iada7abd78f123301a98db982a6272cd9487de72f

1 files changed, 3 insertions, 3 deletions

diff --git a/common_audio/include/audio_util.h b/common_audio/include/audio_util.h
index 18fdbe2a..9972a0e0 100644
--- a/common_audio/include/audio_util.h
+++ b/common_audio/include/audio_util.h
@@ -47,14 +47,14 @@ static inline float ScaleToFloat(int16_t v) {
 }
 
 // Round |size| elements of |src| to int16 with clamping and write to |dest|.
-void RoundToInt16(const float* src, int size, int16_t* dest);
+void RoundToInt16(const float* src, size_t size, int16_t* dest);
 
 // Scale (from [-1, 1]) and round |size| elements of |src| to full-range int16
 // with clamping and write to |dest|.
-void ScaleAndRoundToInt16(const float* src, int size, int16_t* dest);
+void ScaleAndRoundToInt16(const float* src, size_t size, int16_t* dest);
 
 // Scale |size| elements of |src| to float [-1, 1] and write to |dest|.
-void ScaleToFloat(const int16_t* src, int size, float* dest);
+void ScaleToFloat(const int16_t* src, size_t size, float* dest);
 
 // Deinterleave audio from |interleaved| to the channel buffers pointed to
 // by |deinterleaved|. There must be sufficient space allocated in the