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); |
} |
} |