| Index: content/renderer/media/webrtc/webrtc_audio_sink_adapter.h
|
| diff --git a/content/renderer/media/webrtc/webrtc_audio_sink_adapter.h b/content/renderer/media/webrtc/webrtc_audio_sink_adapter.h
|
| index 3b837a31735824a384d9891ab2d90d6d7436c365..4e70d194b5a0eb141b661713ea2dea62b91ad57a 100644
|
| --- a/content/renderer/media/webrtc/webrtc_audio_sink_adapter.h
|
| +++ b/content/renderer/media/webrtc/webrtc_audio_sink_adapter.h
|
| @@ -5,6 +5,9 @@
|
| #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_AUDIO_SINK_ADAPTER_H_
|
| #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_AUDIO_SINK_ADAPTER_H_
|
|
|
| +#include <stdint.h>
|
| +
|
| +#include "base/macros.h"
|
| #include "base/memory/scoped_ptr.h"
|
| #include "content/public/renderer/media_stream_audio_sink.h"
|
| #include "media/audio/audio_parameters.h"
|
| @@ -37,7 +40,7 @@ class WebRtcAudioSinkAdapter : public MediaStreamAudioSink {
|
| webrtc::AudioTrackSinkInterface* const sink_;
|
|
|
| media::AudioParameters params_;
|
| - scoped_ptr<int16[]> interleaved_data_;
|
| + scoped_ptr<int16_t[]> interleaved_data_;
|
|
|
| DISALLOW_COPY_AND_ASSIGN(WebRtcAudioSinkAdapter);
|
| };
|
|
|