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 {
 
+const int kSampleRate = 44100;
+const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
+
 // The fake audio data is 0,1,..SCHAR_MAX-1,0,1,... This is to make it easy
 // to detect errors. This function verifies that the buffers contain such data.
 // E.g. if there are two buffers of size 3, buffer 1 would contain 0,1,2 and
 // buffer 2 would contain 3,4,5. Note that SCHAR_MAX is 127 so wrap-around
 // will happen.
 // |buffer| is the audio buffer to verify.
 bool VerifyBuffer(const int8_t* buffer, int buffer_number, int size) {
   int start_value = (buffer_number * size) % SCHAR_MAX;
   for (int i = 0; i < size; ++i) {
     if (buffer[i] != (i + start_value) % SCHAR_MAX) {
@@ skipping 36 matching lines @@
 ACTION_P2(VerifyInputBuffer, iteration, samples_per_10_ms) {
   const int8_t* buffer = static_cast<const int8_t*>(arg0);
   int bytes_per_10_ms = samples_per_10_ms * static_cast<int>(sizeof(int16_t));
   int start_value = (iteration * bytes_per_10_ms) % SCHAR_MAX;
   for (int i = 0; i < bytes_per_10_ms; ++i) {
     EXPECT_EQ(buffer[i], (i + start_value) % SCHAR_MAX);
   }
   return 0;
 }
 
-void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {
-  const int kSamplesPer10Ms = sample_rate * 10 / 1000;
+void RunFineBufferTest(int frame_size_in_samples) {
   const int kFrameSizeBytes =
       frame_size_in_samples * static_cast<int>(sizeof(int16_t));
   const int kNumberOfFrames = 5;
   // Ceiling of integer division: 1 + ((x - 1) / y)
   const int kNumberOfUpdateBufferCalls =
       1 + ((kNumberOfFrames * frame_size_in_samples - 1) / kSamplesPer10Ms);
 
   MockAudioDeviceBuffer audio_device_buffer;
   EXPECT_CALL(audio_device_buffer, RequestPlayoutData(_))
       .WillRepeatedly(Return(kSamplesPer10Ms));
@@ skipping 12 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, kSampleRate,
+                              kFrameSizeBytes);
+  std::unique_ptr<int8_t[]> out_buffer(new int8_t[kFrameSizeBytes]);
+  std::unique_ptr<int8_t[]> in_buffer(new int8_t[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]);
   for (int i = 0; i < kNumberOfFrames; ++i) {
-    fine_buffer.GetPlayoutData(out_buffer.get());
+    fine_buffer.GetPlayoutData(
+        rtc::ArrayView<int8_t>(out_buffer.get(), kFrameSizeBytes));
     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.DeliverRecordedData(
+        rtc::ArrayView<const int8_t>(in_buffer.get(), kFrameSizeBytes), 0, 0);
   }
 }
 
 TEST(FineBufferTest, BufferLessThan10ms) {
-  const int kSampleRate = 44100;
-  const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
   const int kFrameSizeSamples = kSamplesPer10Ms - 50;
-  RunFineBufferTest(kSampleRate, kFrameSizeSamples);
+  RunFineBufferTest(kFrameSizeSamples);
 }
 
 TEST(FineBufferTest, GreaterThan10ms) {
-  const int kSampleRate = 44100;
-  const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
   const int kFrameSizeSamples = kSamplesPer10Ms + 50;
-  RunFineBufferTest(kSampleRate, kFrameSizeSamples);
+  RunFineBufferTest(kFrameSizeSamples);
 }
 
 }  // namespace webrtc