Index: webrtc/audio_send_stream.h |
diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h |
index 24c3d77ab27a30eac58e351381178ca90c27f832..18e71253f1158dfe5681c79000d77f2f7faedc46 100644 |
--- a/webrtc/audio_send_stream.h |
+++ b/webrtc/audio_send_stream.h |
@@ -11,6 +11,7 @@ |
#ifndef WEBRTC_AUDIO_SEND_STREAM_H_ |
#define WEBRTC_AUDIO_SEND_STREAM_H_ |
+#include <memory> |
#include <string> |
#include <vector> |
@@ -84,7 +85,7 @@ class AudioSendStream : public SendStream { |
// Ownership of the encoder object is transferred to Call when the config is |
// passed to Call::CreateAudioSendStream(). |
// TODO(solenberg): Implement, once we configure codecs through the new API. |
- // rtc::scoped_ptr<AudioEncoder> encoder; |
+ // std::unique_ptr<AudioEncoder> encoder; |
int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator. |
int red_payload_type = -1; // pt, or -1 to disable REDundant coding. |
}; |