Refactor the AudioDevice for iOS and improve the performance and stability

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/android/fine_audio_buffer.h"
+#include "webrtc/modules/audio_device/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::AtLeast;
 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.
+// This function replaces the real AudioDeviceBuffer::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;
 }
 
+// Writes a periodic ramp pattern to the supplied |buffer|. See UpdateBuffer()
+// for details.
+void UpdateInputBuffer(int8_t* buffer, int iteration, int size) {
+  int start_value = (iteration * size) % SCHAR_MAX;
+  for (int i = 0; i < size; ++i) {
+    buffer[i] = (i + start_value) % SCHAR_MAX;
+  }
+}
+
+// Action macro which verifies that the recorded 10ms chunk of audio data
+// (in |arg0|) contains the correct reference values even if they have been
+// supplied using a buffer size that is smaller or larger than 10ms.
+// See VerifyBuffer() for details.
+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;
-  const int kFrameSizeBytes = frame_size_in_samples *
-      static_cast<int>(sizeof(int16_t));
+  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();
     }
   }
+  {
+    InSequence s;
+    for (int j = 0; j < kNumberOfUpdateBufferCalls - 1; ++j) {
+      EXPECT_CALL(audio_device_buffer, SetRecordedBuffer(_, kSamplesPer10Ms))
+          .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);
 
   rtc::scoped_ptr<int8_t[]> out_buffer;
-  out_buffer.reset(
-      new int8_t[fine_buffer.RequiredBufferSizeBytes()]);
+  out_buffer.reset(new int8_t[fine_buffer.RequiredPlayoutBufferSizeBytes()]);
+  rtc::scoped_ptr<int8_t[]> in_buffer;
+  in_buffer.reset(new int8_t[kFrameSizeBytes]);
   for (int i = 0; i < kNumberOfFrames; ++i) {
-    fine_buffer.GetBufferData(out_buffer.get());
+    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);
   }
 }
 
 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