| Index: webrtc/modules/audio_processing/audio_processing_impl.cc
|
| diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
|
| index 3d62439bd5c0384149b3d101a35695185f06faec..9e0f156c50f076bf01b7fd8884f58d89610024b5 100644
|
| --- a/webrtc/modules/audio_processing/audio_processing_impl.cc
|
| +++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
|
| @@ -484,7 +484,7 @@ bool AudioProcessingImpl::output_will_be_muted() const {
|
| }
|
|
|
| int AudioProcessingImpl::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,
|
| @@ -683,7 +683,7 @@ int AudioProcessingImpl::ProcessStreamLocked() {
|
| }
|
|
|
| int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,
|
| - int samples_per_channel,
|
| + size_t samples_per_channel,
|
| int sample_rate_hz,
|
| ChannelLayout layout) {
|
| const StreamConfig reverse_config = {
|
|
|