Improved audio buffer handling for iOS

webrtc/modules/audio_device/fine_audio_buffer_unittest.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/audio_device/fine_audio_buffer.h"
 
 #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"
 
 using ::testing::_;
 using ::testing::AtLeast;
 using ::testing::InSequence;
 using ::testing::Return;
 
 namespace webrtc {
@@ skipping 81 matching lines @@
           .WillOnce(VerifyInputBuffer(j, kSamplesPer10Ms))
           .RetiresOnSaturation();
     }
   }
   EXPECT_CALL(audio_device_buffer, SetVQEData(_, _, _))
       .Times(kNumberOfUpdateBufferCalls - 1);
   EXPECT_CALL(audio_device_buffer, DeliverRecordedData())
       .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);
   }
 }
 
 TEST(FineBufferTest, BufferLessThan10ms) {
   const int kSampleRate = 44100;
   const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
   const int kFrameSizeSamples = kSamplesPer10Ms - 50;
   RunFineBufferTest(kSampleRate, kFrameSizeSamples);
 }
 
 TEST(FineBufferTest, GreaterThan10ms) {
   const int kSampleRate = 44100;
   const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
   const int kFrameSizeSamples = kSamplesPer10Ms + 50;
   RunFineBufferTest(kSampleRate, kFrameSizeSamples);
 }
 
 }  // namespace webrtc