| Index: webrtc/common_audio/audio_converter.h
|
| diff --git a/webrtc/common_audio/audio_converter.h b/webrtc/common_audio/audio_converter.h
|
| index 407b5ff9e7321d28dc619b7a809a213648479160..09e4267847cda7fd24dc2483d800cc747fdd9588 100644
|
| --- a/webrtc/common_audio/audio_converter.h
|
| +++ b/webrtc/common_audio/audio_converter.h
|
| @@ -26,9 +26,9 @@ class AudioConverter {
|
| public:
|
| // Returns a new AudioConverter, which will use the supplied format for its
|
| // lifetime. Caller is responsible for the memory.
|
| - static rtc::scoped_ptr<AudioConverter> Create(int src_channels,
|
| + static rtc::scoped_ptr<AudioConverter> Create(size_t src_channels,
|
| size_t src_frames,
|
| - int dst_channels,
|
| + size_t dst_channels,
|
| size_t dst_frames);
|
| virtual ~AudioConverter() {};
|
|
|
| @@ -39,23 +39,23 @@ class AudioConverter {
|
| virtual void Convert(const float* const* src, size_t src_size,
|
| float* const* dst, size_t dst_capacity) = 0;
|
|
|
| - int src_channels() const { return src_channels_; }
|
| + size_t src_channels() const { return src_channels_; }
|
| size_t src_frames() const { return src_frames_; }
|
| - int dst_channels() const { return dst_channels_; }
|
| + size_t dst_channels() const { return dst_channels_; }
|
| size_t dst_frames() const { return dst_frames_; }
|
|
|
| protected:
|
| AudioConverter();
|
| - AudioConverter(int src_channels, size_t src_frames, int dst_channels,
|
| + AudioConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
|
| size_t dst_frames);
|
|
|
| // Helper to CHECK that inputs are correctly sized.
|
| void CheckSizes(size_t src_size, size_t dst_capacity) const;
|
|
|
| private:
|
| - const int src_channels_;
|
| + const size_t src_channels_;
|
| const size_t src_frames_;
|
| - const int dst_channels_;
|
| + const size_t dst_channels_;
|
| const size_t dst_frames_;
|
|
|
| DISALLOW_COPY_AND_ASSIGN(AudioConverter);
|
|
|