Index: webrtc/modules/audio_processing/audio_processing_impl.h |
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h |
index bbd17191585037e9fa739a1390f5013023a4ba8e..0464fcaee343304f772ec3b9ca6e45d2a0c96577 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.h |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h |
@@ -54,15 +54,16 @@ class AudioRate { |
void set(int rate) { |
rate_ = rate; |
- samples_per_channel_ = AudioProcessing::kChunkSizeMs * rate_ / 1000; |
+ samples_per_channel_ = |
+ static_cast<size_t>(AudioProcessing::kChunkSizeMs * rate_ / 1000); |
} |
int rate() const { return rate_; } |
- int samples_per_channel() const { return samples_per_channel_; } |
+ size_t samples_per_channel() const { return samples_per_channel_; } |
private: |
int rate_; |
- int samples_per_channel_; |
+ size_t samples_per_channel_; |
}; |
class AudioFormat : public AudioRate { |
@@ -112,7 +113,7 @@ class AudioProcessingImpl : public AudioProcessing { |
bool output_will_be_muted() const override; |
int ProcessStream(AudioFrame* frame) override; |
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, |
@@ -120,7 +121,7 @@ class AudioProcessingImpl : public AudioProcessing { |
float* const* dest) override; |
int AnalyzeReverseStream(AudioFrame* frame) override; |
int AnalyzeReverseStream(const float* const* data, |
- int samples_per_channel, |
+ size_t samples_per_channel, |
int sample_rate_hz, |
ChannelLayout layout) override; |
int set_stream_delay_ms(int delay) override; |