| Index: webrtc/modules/audio_processing/audio_processing_impl.h
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| diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
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| index 0597cd9518531c2789451933cf24488099fe5d47..74c5bda28422a2136b0af66a16e6d9efc9a41a20 100644
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| --- a/webrtc/modules/audio_processing/audio_processing_impl.h
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| +++ b/webrtc/modules/audio_processing/audio_processing_impl.h
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| @@ -78,7 +78,7 @@ class AudioProcessingImpl : public AudioProcessing {
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|    bool output_will_be_muted() const override;
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|    int ProcessStream(AudioFrame* frame) override;
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|    int ProcessStream(const float* const* src,
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| -                    int samples_per_channel,
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| +                    size_t samples_per_channel,
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|                      int input_sample_rate_hz,
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|                      ChannelLayout input_layout,
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|                      int output_sample_rate_hz,
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| @@ -90,7 +90,7 @@ class AudioProcessingImpl : public AudioProcessing {
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|                      float* const* dest) override;
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|    int AnalyzeReverseStream(AudioFrame* frame) override;
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|    int AnalyzeReverseStream(const float* const* data,
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| -                           int samples_per_channel,
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| +                           size_t samples_per_channel,
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|                             int sample_rate_hz,
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|                             ChannelLayout layout) override;
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|    int AnalyzeReverseStream(const float* const* data,
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| 
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