| 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 0597cd9518531c2789451933cf24488099fe5d47..74c5bda28422a2136b0af66a16e6d9efc9a41a20 100644
|
| --- a/webrtc/modules/audio_processing/audio_processing_impl.h
|
| +++ b/webrtc/modules/audio_processing/audio_processing_impl.h
|
| @@ -78,7 +78,7 @@ 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,
|
| @@ -90,7 +90,7 @@ 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 AnalyzeReverseStream(const float* const* data,
|
|
|