| Index: webrtc/common_audio/audio_converter.h
|
| diff --git a/webrtc/common_audio/audio_converter.h b/webrtc/common_audio/audio_converter.h
|
| index 772872fcd6d14dfb02be5db06ebbd4b512465191..407b5ff9e7321d28dc619b7a809a213648479160 100644
|
| --- a/webrtc/common_audio/audio_converter.h
|
| +++ b/webrtc/common_audio/audio_converter.h
|
| @@ -27,9 +27,9 @@ class AudioConverter {
|
| // 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,
|
| - int src_frames,
|
| + size_t src_frames,
|
| int dst_channels,
|
| - int dst_frames);
|
| + size_t dst_frames);
|
| virtual ~AudioConverter() {};
|
|
|
| // Convert |src|, containing |src_size| samples, to |dst|, having a sample
|
| @@ -40,23 +40,23 @@ class AudioConverter {
|
| float* const* dst, size_t dst_capacity) = 0;
|
|
|
| int src_channels() const { return src_channels_; }
|
| - int src_frames() const { return src_frames_; }
|
| + size_t src_frames() const { return src_frames_; }
|
| int dst_channels() const { return dst_channels_; }
|
| - int dst_frames() const { return dst_frames_; }
|
| + size_t dst_frames() const { return dst_frames_; }
|
|
|
| protected:
|
| AudioConverter();
|
| - AudioConverter(int src_channels, int src_frames, int dst_channels,
|
| - int dst_frames);
|
| + AudioConverter(int src_channels, size_t src_frames, int 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 int src_frames_;
|
| + const size_t src_frames_;
|
| const int dst_channels_;
|
| - const int dst_frames_;
|
| + const size_t dst_frames_;
|
|
|
| DISALLOW_COPY_AND_ASSIGN(AudioConverter);
|
| };
|
|
|