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Unified Diff: webrtc/modules/audio_processing/audio_processing_impl.cc

Issue 1227213002: Update audio code to use size_t more correctly, webrtc/modules/audio_processing/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Resync Created 5 years, 5 months ago
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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 bbfb771182cc7196f0e8e0ca9fb63860617d6b8f..b1ea6e30033f546c9b84a9e262fd240f2691f08a 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -483,7 +483,7 @@ 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,
@@ -682,7 +682,7 @@ 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 = {

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