| Index: webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
|
| diff --git a/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc b/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
|
| index 535f16816cb4998230d6394d14b2af9c88007ea6..7db7c300b2bc53c812fe2f32416dcca8694b7454 100644
|
| --- a/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
|
| +++ b/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
|
| @@ -13,6 +13,7 @@
|
| #include <limits.h>
|
| #include <memory>
|
|
|
| +#include "webrtc/base/array_view.h"
|
| #include "webrtc/modules/audio_device/mock_audio_device_buffer.h"
|
| #include "webrtc/test/gmock.h"
|
| #include "webrtc/test/gtest.h"
|
| @@ -114,18 +115,18 @@ void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {
|
| .Times(kNumberOfUpdateBufferCalls - 1)
|
| .WillRepeatedly(Return(kSamplesPer10Ms));
|
|
|
| - FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes,
|
| - sample_rate);
|
| + FineAudioBuffer fine_buffer(&audio_device_buffer, sample_rate,
|
| + kFrameSizeBytes);
|
|
|
| - std::unique_ptr<int8_t[]> out_buffer;
|
| - out_buffer.reset(new int8_t[kFrameSizeBytes]);
|
| - std::unique_ptr<int8_t[]> in_buffer;
|
| - in_buffer.reset(new int8_t[kFrameSizeBytes]);
|
| + int8_t out_buffer[kFrameSizeBytes];
|
| + int8_t in_buffer[kFrameSizeBytes];
|
| for (int i = 0; i < kNumberOfFrames; ++i) {
|
| - fine_buffer.GetPlayoutData(out_buffer.get());
|
| - EXPECT_TRUE(VerifyBuffer(out_buffer.get(), i, kFrameSizeBytes));
|
| - UpdateInputBuffer(in_buffer.get(), i, kFrameSizeBytes);
|
| - fine_buffer.DeliverRecordedData(in_buffer.get(), kFrameSizeBytes, 0, 0);
|
| + fine_buffer.GetPlayoutData(
|
| + rtc::ArrayView<int8_t>(out_buffer, kFrameSizeBytes));
|
| + EXPECT_TRUE(VerifyBuffer(out_buffer, i, kFrameSizeBytes));
|
| + UpdateInputBuffer(in_buffer, i, kFrameSizeBytes);
|
| + fine_buffer.DeliverRecordedData(
|
| + rtc::ArrayView<const int8_t>(in_buffer, kFrameSizeBytes), 0, 0);
|
| }
|
| }
|
|
|
|
|