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 @@ |
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 @@ |
channels(channels), |
timestamp(timestamp) {} |
- int16_t* data; // The actual 16bit audio data. |
+ const 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. |