| 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;
|
|
|