Refactor the AudioDevice for iOS and improve the performance and stability

webrtc/modules/audio_device/android/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/android/fine_audio_buffer.h"
-
-#include <limits.h>
-#include <memory>
-
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_device/mock_audio_device_buffer.h"
-
-using ::testing::_;
-using ::testing::InSequence;
-using ::testing::Return;
-
-namespace webrtc {
-
-// 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) {
-      return false;
-    }
-  }
-  return true;
-}
-
-// This function replaces GetPlayoutData when it's called (which is done
-// implicitly when calling GetBufferData). It writes the sequence
-// 0,1,..SCHAR_MAX-1,0,1,... to the buffer. Note that this is likely a buffer of
-// different size than the one VerifyBuffer verifies.
-// |iteration| is the number of calls made to UpdateBuffer prior to this call.
-// |samples_per_10_ms| is the number of samples that should be written to the
-// buffer (|arg0|).
-ACTION_P2(UpdateBuffer, iteration, samples_per_10_ms) {
-  int8_t* buffer = static_cast<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) {
-    buffer[i] = (i + start_value) % SCHAR_MAX;
-  }
-  return samples_per_10_ms;
-}
-
-void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {
-  const int kSamplesPer10Ms = sample_rate * 10 / 1000;
-  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));
-  {
-    InSequence s;
-    for (int i = 0; i < kNumberOfUpdateBufferCalls; ++i) {
-      EXPECT_CALL(audio_device_buffer, GetPlayoutData(_))
-          .WillOnce(UpdateBuffer(i, kSamplesPer10Ms))
-          .RetiresOnSaturation();
-    }
-  }
-  FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes,
-                              sample_rate);
-
-  rtc::scoped_ptr<int8_t[]> out_buffer;
-  out_buffer.reset(
-      new int8_t[fine_buffer.RequiredBufferSizeBytes()]);
-  for (int i = 0; i < kNumberOfFrames; ++i) {
-    fine_buffer.GetBufferData(out_buffer.get());
-    EXPECT_TRUE(VerifyBuffer(out_buffer.get(), i, kFrameSizeBytes));
-  }
-}
-
-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