Index: webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc |
diff --git a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc |
index 076a67430d98a4545e419ac2416465cadc8c3b2d..4dfb073fa9fb64c0c280db149cd0cc150c080acf 100644 |
--- a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc |
+++ b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc |
@@ -11,6 +11,7 @@ |
#include <algorithm> |
#include <limits> |
#include <list> |
+#include <memory> |
#include <numeric> |
#include <string> |
#include <vector> |
@@ -21,7 +22,6 @@ |
#include "webrtc/base/criticalsection.h" |
#include "webrtc/base/format_macros.h" |
#include "webrtc/base/logging.h" |
-#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/base/scoped_ref_ptr.h" |
#include "webrtc/modules/audio_device/audio_device_impl.h" |
#include "webrtc/modules/audio_device/include/audio_device.h" |
@@ -145,7 +145,7 @@ class FileAudioStream : public AudioStreamInterface { |
private: |
size_t file_size_in_bytes_; |
int sample_rate_; |
- rtc::scoped_ptr<int16_t[]> file_; |
+ std::unique_ptr<int16_t[]> file_; |
size_t file_pos_; |
}; |
@@ -233,7 +233,7 @@ class FifoAudioStream : public AudioStreamInterface { |
rtc::CriticalSection lock_; |
const size_t frames_per_buffer_; |
const size_t bytes_per_buffer_; |
- rtc::scoped_ptr<AudioBufferList> fifo_; |
+ std::unique_ptr<AudioBufferList> fifo_; |
size_t largest_size_; |
size_t total_written_elements_; |
size_t write_count_; |
@@ -593,7 +593,7 @@ class AudioDeviceTest : public ::testing::Test { |
EXPECT_FALSE(audio_device()->Recording()); |
} |
- rtc::scoped_ptr<EventWrapper> test_is_done_; |
+ std::unique_ptr<EventWrapper> test_is_done_; |
rtc::scoped_refptr<AudioDeviceModule> audio_device_; |
AudioParameters playout_parameters_; |
AudioParameters record_parameters_; |
@@ -761,7 +761,7 @@ TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) { |
NiceMock<MockAudioTransport> mock(kPlayout); |
const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond; |
std::string file_name = GetFileName(playout_sample_rate()); |
- rtc::scoped_ptr<FileAudioStream> file_audio_stream( |
+ std::unique_ptr<FileAudioStream> file_audio_stream( |
new FileAudioStream(num_callbacks, file_name, playout_sample_rate())); |
mock.HandleCallbacks(test_is_done_.get(), file_audio_stream.get(), |
num_callbacks); |
@@ -795,7 +795,7 @@ TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) { |
EXPECT_EQ(record_channels(), playout_channels()); |
EXPECT_EQ(record_sample_rate(), playout_sample_rate()); |
NiceMock<MockAudioTransport> mock(kPlayout | kRecording); |
- rtc::scoped_ptr<FifoAudioStream> fifo_audio_stream( |
+ std::unique_ptr<FifoAudioStream> fifo_audio_stream( |
new FifoAudioStream(playout_frames_per_10ms_buffer())); |
mock.HandleCallbacks(test_is_done_.get(), fifo_audio_stream.get(), |
kFullDuplexTimeInSec * kNumCallbacksPerSecond); |
@@ -824,7 +824,7 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) { |
EXPECT_EQ(record_channels(), playout_channels()); |
EXPECT_EQ(record_sample_rate(), playout_sample_rate()); |
NiceMock<MockAudioTransport> mock(kPlayout | kRecording); |
- rtc::scoped_ptr<LatencyMeasuringAudioStream> latency_audio_stream( |
+ std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream( |
new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer())); |
mock.HandleCallbacks(test_is_done_.get(), latency_audio_stream.get(), |
kMeasureLatencyTimeInSec * kNumCallbacksPerSecond); |