| 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.
|
| };
|
|
|
|
|