| 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;
|
|
|
|
|