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); |
}; |