| Index: webrtc/api/call/audio_sink.h
|
| diff --git a/webrtc/api/call/audio_sink.h b/webrtc/api/call/audio_sink.h
|
| index 8d38763f9402e6e6a351c3a051ad5cd72c210e76..e865ead365cdc68e2ec50a2b4a2a0c607a26990d 100644
|
| --- a/webrtc/api/call/audio_sink.h
|
| +++ b/webrtc/api/call/audio_sink.h
|
| @@ -27,7 +27,7 @@
|
| virtual ~AudioSinkInterface() {}
|
|
|
| struct Data {
|
| - Data(const int16_t* data,
|
| + Data(int16_t* data,
|
| size_t samples_per_channel,
|
| int sample_rate,
|
| size_t channels,
|
| @@ -38,7 +38,7 @@
|
| channels(channels),
|
| timestamp(timestamp) {}
|
|
|
| - const int16_t* data; // The actual 16bit audio data.
|
| + 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.
|
| size_t channels; // Number of channels in the audio data.
|
|
|