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
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 56d5958..1a8bcc1 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 81e2668..97376a2 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 58ad6c8..c4f9a47 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 <sstream>
#include <string>
-#include "gtest/gtest.h"
-
-#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 0d6a6b6..9832565 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 c6ac601..3d8358d 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 @@
}
#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 c1926e4..ff63312 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 566fdcc..a40f2b7 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 0000000..36aa355
--- /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 0000000..de4254e
--- /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 0c61481..b727ccd 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 887c735..82935a5 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 @@
}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 @@
WebRtc_UWord32 _numFrameTypes[6];
};
-} // namespace webrtc
+} // namespace webrtc
-#endif // ACM_TEST_UTILITY_H
+#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_