Adding Opus frame length test

BUG=issue1015

Review URL: https://webrtc-codereview.appspot.com/1193005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3672 4adac7df-926f-26a2-2b94-8c16560cd09d

11 files changed, 396 insertions, 68 deletions

diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/source/audio_coding_module.gypi
index 56d595832e..1a8bcc17e0 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module.gypi
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module.gypi
@@ -134,6 +134,7 @@
              '../test/dual_stream_unittest.cc',
              '../test/EncodeDecodeTest.cc',
              '../test/iSACTest.cc',
+             '../test/opus_test.cc',
              '../test/PCMFile.cc',
              '../test/RTPFile.cc',
              '../test/SpatialAudio.cc',
diff --git a/webrtc/modules/audio_coding/main/test/APITest.cc b/webrtc/modules/audio_coding/main/test/APITest.cc
index 81e2668686..97376a2d05 100644
--- a/webrtc/modules/audio_coding/main/test/APITest.cc
+++ b/webrtc/modules/audio_coding/main/test/APITest.cc
@@ -8,25 +8,27 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/modules/audio_coding/main/test/APITest.h"
+
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
+
 #include <cctype>
 #include <iostream>
 #include <ostream>
 #include <string>
 
-#include "gtest/gtest.h"
-
-#include "APITest.h"
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "testsupport/fileutils.h"
-#include "tick_util.h"
-#include "trace.h"
-#include "utility.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
index 58ad6c8736..c4f9a47067 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -8,21 +8,22 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
 
-#include <sstream>
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
-#include <string>
 
-#include "gtest/gtest.h"
+#include <sstream>
+#include <string>
 
-#include "audio_coding_module.h"
-#include "common_types.h"
-#include "testsupport/fileutils.h"
-#include "trace.h"
-#include "utility.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc b/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
index 0d6a6b6681..983256564f 100644
--- a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
+++ b/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
@@ -8,16 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "TestVADDTX.h"
+#include "webrtc/modules/audio_coding/main/test/TestVADDTX.h"
 
 #include <iostream>
 
-#include "audio_coding_module_typedefs.h"
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "testsupport/fileutils.h"
-#include "trace.h"
-#include "utility.h"
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/system_wrappers/interface/trace.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/main/test/Tester.cc b/webrtc/modules/audio_coding/main/test/Tester.cc
index c6ac601dbf..3d8358d798 100644
--- a/webrtc/modules/audio_coding/main/test/Tester.cc
+++ b/webrtc/modules/audio_coding/main/test/Tester.cc
@@ -12,19 +12,19 @@
 #include <string>
 #include <vector>
 
-#include "gtest/gtest.h"
-
-#include "APITest.h"
-#include "audio_coding_module.h"
-#include "EncodeDecodeTest.h"
-#include "iSACTest.h"
-#include "TestAllCodecs.h"
-#include "TestFEC.h"
-#include "TestStereo.h"
-#include "testsupport/fileutils.h"
-#include "TestVADDTX.h"
-#include "trace.h"
-#include "TwoWayCommunication.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/main/test/APITest.h"
+#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/main/test/iSACTest.h"
+#include "webrtc/modules/audio_coding/main/test/opus_test.h"
+#include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h"
+#include "webrtc/modules/audio_coding/main/test/TestFEC.h"
+#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/main/test/TestVADDTX.h"
+#include "webrtc/modules/audio_coding/main/test/TwoWayCommunication.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
 
 using webrtc::AudioCodingModule;
 using webrtc::Trace;
@@ -128,6 +128,14 @@ TEST(AudioCodingModuleTest, TestAllCodecs) {
 }
 #endif
 
+TEST(AudioCodingModuleTest, TestOpus) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "acm_opus_trace.txt").c_str());
+  webrtc::OpusTest().Perform();
+  Trace::ReturnTrace();
+}
+
 TEST(AudioCodingModuleTest, RunAllTests) {
   std::vector<ACMTest*> tests;
   PopulateTests(&tests);
diff --git a/webrtc/modules/audio_coding/main/test/delay_test.cc b/webrtc/modules/audio_coding/main/test/delay_test.cc
index c1926e4c1f..ff63312bd2 100644
--- a/webrtc/modules/audio_coding/main/test/delay_test.cc
+++ b/webrtc/modules/audio_coding/main/test/delay_test.cc
@@ -16,16 +16,17 @@
 #include <iostream>
 
 #include "gflags/gflags.h"
-#include "gtest/gtest.h"
-#include "testsupport/fileutils.h"
+#include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/main/test/Channel.h"
 #include "webrtc/modules/audio_coding/main/test/PCMFile.h"
 #include "webrtc/modules/audio_coding/main/test/utility.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/test/testsupport/fileutils.h"
 
 DEFINE_string(codec, "isac", "Codec Name");
 DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.cc b/webrtc/modules/audio_coding/main/test/iSACTest.cc
index 566fdcc212..a40f2b72dd 100644
--- a/webrtc/modules/audio_coding/main/test/iSACTest.cc
+++ b/webrtc/modules/audio_coding/main/test/iSACTest.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/modules/audio_coding/main/test/iSACTest.h"
+
 #include <cctype>
 #include <stdio.h>
 #include <string.h>
@@ -21,12 +23,12 @@
 #include <time.h>
 #endif 
 
-#include "event_wrapper.h"
-#include "iSACTest.h"
-#include "utility.h"
-#include "trace.h"
-#include "testsupport/fileutils.h"
-#include "tick_util.h"
+#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/system_wrappers/interface/event_wrapper.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.cc b/webrtc/modules/audio_coding/main/test/opus_test.cc
new file mode 100644
index 0000000000..36aa355c71
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/test/opus_test.cc
@@ -0,0 +1,270 @@
+/*
+ *  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_coding/main/test/opus_test.h"
+
+#include <cassert>
+#include <string>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/main/source/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/source/acm_opus.h"
+#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+OpusTest::OpusTest()
+    : acm_receiver_(NULL),
+      channel_a2b_(NULL),
+      counter_(0),
+      payload_type_(255),
+      rtp_timestamp_(0) {
+}
+
+OpusTest::~OpusTest() {
+  if (acm_receiver_ != NULL) {
+    AudioCodingModule::Destroy(acm_receiver_);
+    acm_receiver_ = NULL;
+  }
+  if (channel_a2b_ != NULL) {
+    delete channel_a2b_;
+    channel_a2b_ = NULL;
+  }
+  if (opus_mono_encoder_ != NULL) {
+    WebRtcOpus_EncoderFree(opus_mono_encoder_);
+    opus_mono_encoder_ = NULL;
+  }
+  if (opus_stereo_encoder_ != NULL) {
+    WebRtcOpus_EncoderFree(opus_stereo_encoder_);
+    opus_stereo_encoder_ = NULL;
+  }
+}
+
+void OpusTest::Perform() {
+#ifndef WEBRTC_CODEC_OPUS
+  // Opus isn't defined, exit.
+  return;
+#else
+  uint16_t frequency_hz;
+  int audio_channels;
+  int16_t test_cntr = 0;
+
+  // Open both mono and stereo test files in 32 kHz.
+  const std::string file_name_stereo =
+      webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm");
+  const std::string file_name_mono =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  frequency_hz = 32000;
+  in_file_stereo_.Open(file_name_stereo, frequency_hz, "rb");
+  in_file_stereo_.ReadStereo(true);
+  in_file_mono_.Open(file_name_mono, frequency_hz, "rb");
+  in_file_mono_.ReadStereo(false);
+
+  // Create Opus encoders for mono and stereo.
+  ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1), -1);
+  ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2), -1);
+
+  // Create and initialize one ACM, to be used as receiver.
+  acm_receiver_ = AudioCodingModule::Create(0);
+  ASSERT_TRUE(acm_receiver_ != NULL);
+  EXPECT_EQ(0, acm_receiver_->InitializeReceiver());
+
+  // Register Opus stereo as receiving codec.
+  CodecInst opus_codec_param;
+  int codec_id = acm_receiver_->Codec("opus", 48000, 2);
+  EXPECT_EQ(0, acm_receiver_->Codec(codec_id, &opus_codec_param));
+  payload_type_ = opus_codec_param.pltype;
+  EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
+
+  // Create and connect the channel.
+  channel_a2b_ = new TestPackStereo;
+  channel_a2b_->RegisterReceiverACM(acm_receiver_);
+
+  //
+  // Test Stereo.
+  //
+
+  channel_a2b_->set_codec_mode(kStereo);
+  audio_channels = 2;
+  test_cntr++;
+  OpenOutFile(test_cntr);
+
+  // Run Opus with 2.5 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 120);
+
+  // Run Opus with 5 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 240);
+
+  // Run Opus with 10 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 480);
+
+  // Run Opus with 20 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 960);
+
+  // Run Opus with 40 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 1920);
+
+  // Run Opus with 60 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 2880);
+
+  out_file_.Close();
+
+  //
+  // Test Mono.
+  //
+  channel_a2b_->set_codec_mode(kMono);
+  audio_channels = 1;
+  test_cntr++;
+  OpenOutFile(test_cntr);
+
+  // Register Opus mono as receiving codec.
+  opus_codec_param.channels = 1;
+  EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
+
+  // Run Opus with 2.5 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 120);
+
+  // Run Opus with 5 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 240);
+
+  // Run Opus with 10 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 480);
+
+  // Run Opus with 20 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 960);
+
+  // Run Opus with 40 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 1920);
+
+  // Run Opus with 60 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 2880);
+
+  // Close the files.
+  in_file_stereo_.Close();
+  in_file_mono_.Close();
+  out_file_.Close();
+#endif
+}
+
+void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
+                   int frame_length, int percent_loss) {
+  AudioFrame audio_frame;
+  int32_t out_freq_hz_b = out_file_.SamplingFrequency();
+  int16_t audio[480 * 12 * 2];  // Can hold 120 ms stereo audio.
+  int written_samples = 0;
+  int read_samples = 0;
+  channel->reset_payload_size();
+
+  // Set encoder rate.
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, bitrate));
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, bitrate));
+
+  while (1) {
+    // Simulate packet loss by setting |packet_loss_| to "true" in
+    // |percent_loss| percent of the loops.
+    // TODO(tlegrand): Move handling of loss simulation to TestPackStereo.
+    if (percent_loss > 0) {
+      if (counter_ == floor((100 / percent_loss) + 0.5)) {
+        counter_ = 0;
+        channel->set_lost_packet(true);
+      } else {
+        channel->set_lost_packet(false);
+      }
+      counter_++;
+    }
+
+    // Get 10 msec of audio.
+    if (channels == 1) {
+      if (in_file_mono_.EndOfFile()) {
+        break;
+      }
+      in_file_mono_.Read10MsData(audio_frame);
+    } else {
+      if (in_file_stereo_.EndOfFile()) {
+        break;
+      }
+      in_file_stereo_.Read10MsData(audio_frame);
+    }
+
+    // Input audio is sampled at 32 kHz, but Opus operates at 48 kHz.
+    // Resampling is required.
+    EXPECT_EQ(480, resampler_.Resample10Msec(audio_frame.data_, 32000,
+                                             &audio[written_samples], 48000,
+                                             channels));
+    written_samples += 480 * channels;
+
+    // Sometimes we need to loop over the audio vector to produce the right
+    // number of packets.
+    int loop_encode = (written_samples - read_samples) /
+        (channels * frame_length);
+
+    if (loop_encode > 0) {
+      const int kMaxBytes = 1000;  // Maximum number of bytes for one packet.
+      int16_t bitstream_len_byte;
+      uint8_t bitstream[kMaxBytes];
+      for (int i = 0; i < loop_encode; i++) {
+        if (channels == 1) {
+          bitstream_len_byte = WebRtcOpus_Encode(
+              opus_mono_encoder_, &audio[read_samples],
+              frame_length, kMaxBytes, bitstream);
+          ASSERT_GT(bitstream_len_byte, -1);
+        } else {
+          bitstream_len_byte = WebRtcOpus_Encode(
+              opus_stereo_encoder_, &audio[read_samples],
+              frame_length, kMaxBytes, bitstream);
+          ASSERT_GT(bitstream_len_byte, -1);
+        }
+        channel->SendData(kAudioFrameSpeech, payload_type_, rtp_timestamp_,
+                          bitstream, bitstream_len_byte, NULL);
+        rtp_timestamp_ += frame_length;
+        read_samples += frame_length * channels;
+      }
+      if (read_samples == written_samples) {
+        read_samples = 0;
+        written_samples = 0;
+      }
+    }
+
+    // Run received side of ACM.
+    CHECK_ERROR(acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
+
+    // Write output speech to file.
+    out_file_.Write10MsData(
+        audio_frame.data_,
+        audio_frame.samples_per_channel_ * audio_frame.num_channels_);
+  }
+
+  if (in_file_mono_.EndOfFile()) {
+    in_file_mono_.Rewind();
+  }
+  if (in_file_stereo_.EndOfFile()) {
+    in_file_stereo_.Rewind();
+  }
+  // Reset in case we ended with a lost packet.
+  channel->set_lost_packet(false);
+}
+
+void OpusTest::OpenOutFile(int test_number) {
+  std::string file_name;
+  std::stringstream file_stream;
+  file_stream << webrtc::test::OutputPath() << "opustest_out_"
+      << test_number << ".pcm";
+  file_name = file_stream.str();
+  out_file_.Open(file_name, 32000, "wb");
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.h b/webrtc/modules/audio_coding/main/test/opus_test.h
new file mode 100644
index 0000000000..de4254eb32
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/test/opus_test.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
+
+#include <math.h>
+
+#include "webrtc/modules/audio_coding/main/source/acm_opus.h"
+#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
+#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/main/test/Channel.h"
+#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
+
+namespace webrtc {
+
+class OpusTest : public ACMTest {
+ public:
+  OpusTest();
+  ~OpusTest();
+
+  void Perform();
+ private:
+  void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
+           int percent_loss = 0);
+
+  void OpenOutFile(int test_number);
+
+  AudioCodingModule* acm_receiver_;
+  TestPackStereo* channel_a2b_;
+  PCMFile in_file_stereo_;
+  PCMFile in_file_mono_;
+  PCMFile out_file_;
+  int counter_;
+  uint8_t payload_type_;
+  int rtp_timestamp_;
+  ACMResampler resampler_;
+  WebRtcOpusEncInst* opus_mono_encoder_;
+  WebRtcOpusEncInst* opus_stereo_encoder_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
diff --git a/webrtc/modules/audio_coding/main/test/utility.cc b/webrtc/modules/audio_coding/main/test/utility.cc
index 0c614819a8..b727ccd0b6 100644
--- a/webrtc/modules/audio_coding/main/test/utility.cc
+++ b/webrtc/modules/audio_coding/main/test/utility.cc
@@ -14,9 +14,10 @@
 #include <stdio.h>
 #include <stdlib.h>
 
-#include "audio_coding_module.h"
-#include "common_types.h"
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
 
 #define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13
 
diff --git a/webrtc/modules/audio_coding/main/test/utility.h b/webrtc/modules/audio_coding/main/test/utility.h
index 887c735350..82935a537f 100644
--- a/webrtc/modules/audio_coding/main/test/utility.h
+++ b/webrtc/modules/audio_coding/main/test/utility.h
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef ACM_TEST_UTILITY_H
-#define ACM_TEST_UTILITY_H
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
 
-#include "audio_coding_module.h"
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
 
 namespace webrtc {
 
@@ -55,17 +55,6 @@ namespace webrtc {
     }while(0)
 
 
-
-#ifdef WIN32
-    /* Exclude rarely-used stuff from Windows headers */
-    //#define WIN32_LEAN_AND_MEAN 
-    /* OS-dependent case-insensitive string comparison */
-    #define STR_CASE_CMP(x,y) ::_stricmp(x,y)
-#else
-    /* OS-dependent case-insensitive string comparison */
-    #define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
-#endif
-
 #define DESTROY_ACM(acm)                                                                    \
     do {                                                                                    \
         if(acm != NULL) {                                                                   \
@@ -190,6 +179,6 @@ private:
     WebRtc_UWord32 _numFrameTypes[6];
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 
-#endif // ACM_TEST_UTILITY_H
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_