Replace scoped_ptr with unique_ptr in webrtc/modules/audio_device/

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1722083002

Cr-Commit-Position: refs/heads/master@{#11740}
This commit is contained in:
kwiberg
2016-02-24 05:00:36 -08:00
committed by Commit bot
parent 58e08cbea8
commit f01633e667
24 changed files with 111 additions and 90 deletions

View File

@ -11,7 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
#include "webrtc/base/scoped_ptr.h"
#include <memory>
#include "webrtc/typedefs.h"
namespace webrtc {
@ -86,14 +87,14 @@ class FineAudioBuffer {
// Number of audio bytes per 10ms.
const size_t bytes_per_10_ms_;
// Storage for output samples that are not yet asked for.
rtc::scoped_ptr<int8_t[]> playout_cache_buffer_;
std::unique_ptr<int8_t[]> playout_cache_buffer_;
// Location of first unread output sample.
size_t playout_cached_buffer_start_;
// Number of bytes stored in output (contain samples to be played out) cache.
size_t playout_cached_bytes_;
// Storage for input samples that are about to be delivered to the WebRTC
// ADB or remains from the last successful delivery of a 10ms audio buffer.
rtc::scoped_ptr<int8_t[]> record_cache_buffer_;
std::unique_ptr<int8_t[]> record_cache_buffer_;
// Required (max) size in bytes of the |record_cache_buffer_|.
const size_t required_record_buffer_size_bytes_;
// Number of bytes in input (contains recorded samples) cache.