Index: webrtc/audio/audio_sink.h |
diff --git a/webrtc/audio/audio_sink.h b/webrtc/audio/audio_sink.h |
index d022b32855daf553482437db9a9dd87c56df163c..999644f4ce161946e55e59645d29d043b723c0c8 100644 |
--- a/webrtc/audio/audio_sink.h |
+++ b/webrtc/audio/audio_sink.h |
@@ -30,7 +30,7 @@ class AudioSinkInterface { |
Data(int16_t* data, |
size_t samples_per_channel, |
int sample_rate, |
- int channels, |
+ size_t channels, |
uint32_t timestamp) |
: data(data), |
samples_per_channel(samples_per_channel), |
@@ -41,7 +41,7 @@ class AudioSinkInterface { |
int16_t* data; // The actual 16bit audio data. |
size_t samples_per_channel; // Number of frames in the buffer. |
int sample_rate; // Sample rate in Hz. |
- int channels; // Number of channels in the audio data. |
+ size_t channels; // Number of channels in the audio data. |
uint32_t timestamp; // The RTP timestamp of the first sample. |
}; |