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 = { |