Index: webrtc/modules/audio_device/android/opensles_player.h |
diff --git a/webrtc/modules/audio_device/android/opensles_player.h b/webrtc/modules/audio_device/android/opensles_player.h |
index fa9e931218cebfd0332404198d4a82189434d0aa..084546dbf7b3ab2d5fc2a32816d5441aa6388d96 100644 |
--- a/webrtc/modules/audio_device/android/opensles_player.h |
+++ b/webrtc/modules/audio_device/android/opensles_player.h |
@@ -11,11 +11,12 @@ |
#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_ |
#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_ |
+#include <memory> |
+ |
#include <SLES/OpenSLES.h> |
#include <SLES/OpenSLES_Android.h> |
#include <SLES/OpenSLES_AndroidConfiguration.h> |
-#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/base/thread_checker.h" |
#include "webrtc/modules/audio_device/android/audio_common.h" |
#include "webrtc/modules/audio_device/android/audio_manager.h" |
@@ -150,7 +151,7 @@ class OpenSLESPlayer { |
// Queue of audio buffers to be used by the player object for rendering |
// audio. They will be used in a Round-robin way and the size of each buffer |
// is given by FineAudioBuffer::RequiredBufferSizeBytes(). |
- rtc::scoped_ptr<SLint8[]> audio_buffers_[kNumOfOpenSLESBuffers]; |
+ std::unique_ptr<SLint8[]> audio_buffers_[kNumOfOpenSLESBuffers]; |
// FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data |
// in chunks of 10ms. It then allows for this data to be pulled in |
@@ -162,7 +163,7 @@ class OpenSLESPlayer { |
// in each callback (one every 5ms). This class can then ask for 240 and the |
// FineAudioBuffer will ask WebRTC for new data only every second callback |
// and also cach non-utilized audio. |
- rtc::scoped_ptr<FineAudioBuffer> fine_buffer_; |
+ std::unique_ptr<FineAudioBuffer> fine_buffer_; |
// Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue. |
// Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ... |