Chromium Code Reviews| Index: webrtc/api/call/audio_sink.h |
| diff --git a/webrtc/api/call/audio_sink.h b/webrtc/api/call/audio_sink.h |
| index e865ead365cdc68e2ec50a2b4a2a0c607a26990d..8d38763f9402e6e6a351c3a051ad5cd72c210e76 100644 |
| --- a/webrtc/api/call/audio_sink.h |
| +++ b/webrtc/api/call/audio_sink.h |
| @@ -27,7 +27,7 @@ class AudioSinkInterface { |
| virtual ~AudioSinkInterface() {} |
| struct Data { |
| - Data(int16_t* data, |
| + Data(const int16_t* data, |
| size_t samples_per_channel, |
| int sample_rate, |
| size_t channels, |
| @@ -38,7 +38,7 @@ class AudioSinkInterface { |
| channels(channels), |
| timestamp(timestamp) {} |
| - int16_t* data; // The actual 16bit audio data. |
| + const int16_t* data; // The actual 16bit audio data. |
|
kwiberg-webrtc
2017/05/10 08:19:51
Seems like a good change to me---it can't break ca
the sun
2017/05/11 07:56:15
Agree with Karl on all points (let's not address t
|
| 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. |