Index: webrtc/modules/audio_processing/include/audio_processing.h |
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h |
index 6fa1c96c0771c14d141836dc0c88bf69ec9f5aea..85300a568d2fbdec46c7de6380700106d8aacad1 100644 |
--- a/webrtc/modules/audio_processing/include/audio_processing.h |
+++ b/webrtc/modules/audio_processing/include/audio_processing.h |
@@ -295,7 +295,7 @@ class AudioProcessing { |
// The output layout may only remove channels, not add. |src| and |dest| |
// may use the same memory, if desired. |
virtual int ProcessStream(const float* const* src, |
- int samples_per_channel, |
+ size_t samples_per_channel, |
int input_sample_rate_hz, |
ChannelLayout input_layout, |
int output_sample_rate_hz, |
@@ -322,7 +322,7 @@ class AudioProcessing { |
// Accepts deinterleaved float audio with the range [-1, 1]. Each element |
// of |data| points to a channel buffer, arranged according to |layout|. |
virtual int AnalyzeReverseStream(const float* const* data, |
- int samples_per_channel, |
+ size_t samples_per_channel, |
int sample_rate_hz, |
ChannelLayout layout) = 0; |