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 87b82a6a3509131adae9ed698cc0f896fd01d4c0..ee2ae20a7584c07ffda735bdedeee89efc7af0a3 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc |
@@ -473,7 +473,7 @@ 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, |
@@ -665,7 +665,7 @@ 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) { |
CriticalSectionScoped crit_scoped(crit_); |