Removing AudioCoding duplicate tests
Reverting to using one version of ACM in ACM tests.
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5924 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/test/APITest.cc b/webrtc/modules/audio_coding/main/test/APITest.cc
index 15bac6a..82940fa 100644
--- a/webrtc/modules/audio_coding/main/test/APITest.cc
+++ b/webrtc/modules/audio_coding/main/test/APITest.cc
@@ -56,8 +56,8 @@
}
APITest::APITest(const Config& config)
- : _acmA(config.Get<AudioCodingModuleFactory>().Create(1)),
- _acmB(config.Get<AudioCodingModuleFactory>().Create(2)),
+ : _acmA(AudioCodingModule::Create(1)),
+ _acmB(AudioCodingModule::Create(2)),
_channel_A2B(NULL),
_channel_B2A(NULL),
_writeToFile(true),
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
index cdf9fdc..1ee6abc 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -19,7 +19,6 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
-#include "webrtc/common.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
@@ -242,16 +241,14 @@
}
}
-EncodeDecodeTest::EncodeDecodeTest(const Config& config)
- : config_(config) {
+EncodeDecodeTest::EncodeDecodeTest() {
_testMode = 2;
Trace::CreateTrace();
Trace::SetTraceFile(
(webrtc::test::OutputPath() + "acm_encdec_trace.txt").c_str());
}
-EncodeDecodeTest::EncodeDecodeTest(int testMode, const Config& config)
- : config_(config) {
+EncodeDecodeTest::EncodeDecodeTest(int testMode) {
//testMode == 0 for autotest
//testMode == 1 for testing all codecs/parameters
//testMode > 1 for specific user-input test (as it was used before)
@@ -273,8 +270,7 @@
codePars[1] = 0;
codePars[2] = 0;
- scoped_ptr<AudioCodingModule> acm(
- config_.Get<AudioCodingModuleFactory>().Create(0));
+ scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
struct CodecInst sendCodecTmp;
numCodecs = acm->NumberOfCodecs();
@@ -329,8 +325,7 @@
void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
int testMode) {
- scoped_ptr<AudioCodingModule> acm(
- config_.Get<AudioCodingModuleFactory>().Create(1));
+ scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
RTPFile rtpFile;
std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
rtpFile.Open(fileName.c_str(), "wb+");
diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
index 5aa3596..4fdd943 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
@@ -23,8 +23,6 @@
#define MAX_INCOMING_PAYLOAD 8096
-class Config;
-
// TestPacketization callback which writes the encoded payloads to file
class TestPacketization : public AudioPacketizationCallback {
public:
@@ -92,8 +90,8 @@
class EncodeDecodeTest : public ACMTest {
public:
- explicit EncodeDecodeTest(const Config& config);
- EncodeDecodeTest(int testMode, const Config& config);
+ EncodeDecodeTest();
+ explicit EncodeDecodeTest(int testMode);
virtual void Perform();
uint16_t _playoutFreq;
@@ -102,8 +100,6 @@
private:
void EncodeToFile(int fileType, int codeId, int* codePars, int testMode);
- const Config& config_;
-
protected:
Sender _sender;
Receiver _receiver;
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
index fba7f03..d6c6dc4 100644
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
+++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
@@ -99,9 +99,9 @@
payload_size_ = 0;
}
-TestAllCodecs::TestAllCodecs(int test_mode, const Config& config)
- : acm_a_(config.Get<AudioCodingModuleFactory>().Create(0)),
- acm_b_(config.Get<AudioCodingModuleFactory>().Create(1)),
+TestAllCodecs::TestAllCodecs(int test_mode)
+ : acm_a_(AudioCodingModule::Create(0)),
+ acm_b_(AudioCodingModule::Create(1)),
channel_a_to_b_(NULL),
test_count_(0),
packet_size_samples_(0),
diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h b/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
index 0231d84..10d82ae 100644
--- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
+++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.h
@@ -11,7 +11,6 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
-#include "webrtc/common.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"
@@ -50,7 +49,7 @@
class TestAllCodecs : public ACMTest {
public:
- TestAllCodecs(int test_mode, const Config& config);
+ explicit TestAllCodecs(int test_mode);
~TestAllCodecs();
void Perform();
diff --git a/webrtc/modules/audio_coding/main/test/TestFEC.cc b/webrtc/modules/audio_coding/main/test/TestFEC.cc
index 032579c..76b6d4b 100644
--- a/webrtc/modules/audio_coding/main/test/TestFEC.cc
+++ b/webrtc/modules/audio_coding/main/test/TestFEC.cc
@@ -13,7 +13,6 @@
#include <assert.h>
#include <iostream>
-#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
@@ -23,12 +22,11 @@
namespace webrtc {
-TestFEC::TestFEC(const Config& config)
- : _acmA(config.Get<AudioCodingModuleFactory>().Create(0)),
- _acmB(config.Get<AudioCodingModuleFactory>().Create(1)),
+TestFEC::TestFEC()
+ : _acmA(AudioCodingModule::Create(0)),
+ _acmB(AudioCodingModule::Create(1)),
_channelA2B(NULL),
- _testCntr(0) {
-}
+ _testCntr(0) {}
TestFEC::~TestFEC() {
if (_channelA2B != NULL) {
diff --git a/webrtc/modules/audio_coding/main/test/TestFEC.h b/webrtc/modules/audio_coding/main/test/TestFEC.h
index af3cdd7..d7a6223 100644
--- a/webrtc/modules/audio_coding/main/test/TestFEC.h
+++ b/webrtc/modules/audio_coding/main/test/TestFEC.h
@@ -18,11 +18,9 @@
namespace webrtc {
-class Config;
-
class TestFEC : public ACMTest {
public:
- explicit TestFEC(const Config& config);
+ TestFEC();
~TestFEC();
void Perform();
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.cc b/webrtc/modules/audio_coding/main/test/TestStereo.cc
index b26334c..f058967 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.cc
+++ b/webrtc/modules/audio_coding/main/test/TestStereo.cc
@@ -15,7 +15,6 @@
#include <string>
#include "gtest/gtest.h"
-#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
@@ -108,9 +107,9 @@
lost_packet_ = lost;
}
-TestStereo::TestStereo(int test_mode, const Config& config)
- : acm_a_(config.Get<AudioCodingModuleFactory>().Create(0)),
- acm_b_(config.Get<AudioCodingModuleFactory>().Create(1)),
+TestStereo::TestStereo(int test_mode)
+ : acm_a_(AudioCodingModule::Create(0)),
+ acm_b_(AudioCodingModule::Create(1)),
channel_a2b_(NULL),
test_cntr_(0),
pack_size_samp_(0),
diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.h b/webrtc/modules/audio_coding/main/test/TestStereo.h
index 88320a0..03f8041 100644
--- a/webrtc/modules/audio_coding/main/test/TestStereo.h
+++ b/webrtc/modules/audio_coding/main/test/TestStereo.h
@@ -20,8 +20,6 @@
namespace webrtc {
-class Config;
-
enum StereoMonoMode {
kNotSet,
kMono,
@@ -62,7 +60,7 @@
class TestStereo : public ACMTest {
public:
- TestStereo(int test_mode, const Config& config);
+ explicit TestStereo(int test_mode);
~TestStereo();
void Perform();
diff --git a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc b/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
index 22e9696..d31e1d4 100644
--- a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
+++ b/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
@@ -12,7 +12,6 @@
#include <iostream>
-#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
@@ -23,11 +22,10 @@
namespace webrtc {
-TestVADDTX::TestVADDTX(const Config& config)
- : _acmA(config.Get<AudioCodingModuleFactory>().Create(0)),
- _acmB(config.Get<AudioCodingModuleFactory>().Create(1)),
- _channelA2B(NULL) {
-}
+TestVADDTX::TestVADDTX()
+ : _acmA(AudioCodingModule::Create(0)),
+ _acmB(AudioCodingModule::Create(1)),
+ _channelA2B(NULL) {}
TestVADDTX::~TestVADDTX() {
if (_channelA2B != NULL) {
diff --git a/webrtc/modules/audio_coding/main/test/TestVADDTX.h b/webrtc/modules/audio_coding/main/test/TestVADDTX.h
index e0aa6b8..f8c97e1 100644
--- a/webrtc/modules/audio_coding/main/test/TestVADDTX.h
+++ b/webrtc/modules/audio_coding/main/test/TestVADDTX.h
@@ -18,8 +18,6 @@
namespace webrtc {
-class Config;
-
typedef struct {
bool statusDTX;
bool statusVAD;
@@ -49,7 +47,7 @@
class TestVADDTX : public ACMTest {
public:
- explicit TestVADDTX(const Config& config);
+ TestVADDTX();
~TestVADDTX();
void Perform();
diff --git a/webrtc/modules/audio_coding/main/test/Tester.cc b/webrtc/modules/audio_coding/main/test/Tester.cc
index 581b7bd..a89c9cd 100644
--- a/webrtc/modules/audio_coding/main/test/Tester.cc
+++ b/webrtc/modules/audio_coding/main/test/Tester.cc
@@ -13,7 +13,6 @@
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common.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"
@@ -24,7 +23,6 @@
#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/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
@@ -39,21 +37,7 @@
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_allcodecs_trace.txt").c_str());
- webrtc::Config config;
-
- UseLegacyAcm(&config);
- webrtc::TestAllCodecs(ACM_TEST_MODE, config).Perform();
- Trace::ReturnTrace();
-}
-
-TEST(AudioCodingModuleTest, TestAllCodecsNewACM) {
- Trace::CreateTrace();
- Trace::SetTraceFile((webrtc::test::OutputPath() +
- "acm_allcodecs_trace_new.txt").c_str());
- webrtc::Config config;
-
- UseNewAcm(&config);
- webrtc::TestAllCodecs(ACM_TEST_MODE, config).Perform();
+ webrtc::TestAllCodecs(ACM_TEST_MODE).Perform();
Trace::ReturnTrace();
}
@@ -61,21 +45,7 @@
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_encodedecode_trace.txt").c_str());
- webrtc::Config config;
-
- UseLegacyAcm(&config);
- webrtc::EncodeDecodeTest(ACM_TEST_MODE, config).Perform();
- Trace::ReturnTrace();
-}
-
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecodeNewACM)) {
- Trace::CreateTrace();
- Trace::SetTraceFile((webrtc::test::OutputPath() +
- "acm_encodedecode_trace_new.txt").c_str());
- webrtc::Config config;
-
- UseNewAcm(&config);
- webrtc::EncodeDecodeTest(ACM_TEST_MODE, config).Perform();
+ webrtc::EncodeDecodeTest(ACM_TEST_MODE).Perform();
Trace::ReturnTrace();
}
@@ -83,21 +53,7 @@
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_fec_trace.txt").c_str());
- webrtc::Config config;
-
- UseLegacyAcm(&config);
- webrtc::TestFEC(config).Perform();
- Trace::ReturnTrace();
-}
-
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestFECNewACM)) {
- Trace::CreateTrace();
- Trace::SetTraceFile((webrtc::test::OutputPath() +
- "acm_fec_trace_new.txt").c_str());
- webrtc::Config config;
-
- UseNewAcm(&config);
- webrtc::TestFEC(config).Perform();
+ webrtc::TestFEC().Perform();
Trace::ReturnTrace();
}
@@ -105,21 +61,7 @@
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_isac_trace.txt").c_str());
- webrtc::Config config;
-
- UseLegacyAcm(&config);
- webrtc::ISACTest(ACM_TEST_MODE, config).Perform();
- Trace::ReturnTrace();
-}
-
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestIsacNewACM)) {
- Trace::CreateTrace();
- Trace::SetTraceFile((webrtc::test::OutputPath() +
- "acm_isac_trace_new.txt").c_str());
- webrtc::Config config;
-
- UseNewAcm(&config);
- webrtc::ISACTest(ACM_TEST_MODE, config).Perform();
+ webrtc::ISACTest(ACM_TEST_MODE).Perform();
Trace::ReturnTrace();
}
@@ -127,21 +69,7 @@
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_twowaycom_trace.txt").c_str());
- webrtc::Config config;
-
- UseLegacyAcm(&config);
- webrtc::TwoWayCommunication(ACM_TEST_MODE, config).Perform();
- Trace::ReturnTrace();
-}
-
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TwoWayCommunicationNewACM)) {
- Trace::CreateTrace();
- Trace::SetTraceFile((webrtc::test::OutputPath() +
- "acm_twowaycom_trace_new.txt").c_str());
- webrtc::Config config;
-
- UseNewAcm(&config);
- webrtc::TwoWayCommunication(ACM_TEST_MODE, config).Perform();
+ webrtc::TwoWayCommunication(ACM_TEST_MODE).Perform();
Trace::ReturnTrace();
}
@@ -149,21 +77,7 @@
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_stereo_trace.txt").c_str());
- webrtc::Config config;
-
- UseLegacyAcm(&config);
- webrtc::TestStereo(ACM_TEST_MODE, config).Perform();
- Trace::ReturnTrace();
-}
-
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestStereoNewACM)) {
- Trace::CreateTrace();
- Trace::SetTraceFile((webrtc::test::OutputPath() +
- "acm_stereo_trace_new.txt").c_str());
- webrtc::Config config;
-
- UseNewAcm(&config);
- webrtc::TestStereo(ACM_TEST_MODE, config).Perform();
+ webrtc::TestStereo(ACM_TEST_MODE).Perform();
Trace::ReturnTrace();
}
@@ -171,21 +85,7 @@
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_vaddtx_trace.txt").c_str());
- webrtc::Config config;
-
- UseLegacyAcm(&config);
- webrtc::TestVADDTX(config).Perform();
- Trace::ReturnTrace();
-}
-
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestVADDTXNewACM)) {
- Trace::CreateTrace();
- Trace::SetTraceFile((webrtc::test::OutputPath() +
- "acm_vaddtx_trace_new.txt").c_str());
- webrtc::Config config;
-
- UseNewAcm(&config);
- webrtc::TestVADDTX(config).Perform();
+ webrtc::TestVADDTX().Perform();
Trace::ReturnTrace();
}
@@ -193,21 +93,7 @@
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_opus_trace.txt").c_str());
- webrtc::Config config;
-
- UseLegacyAcm(&config);
- webrtc::OpusTest(config).Perform();
- Trace::ReturnTrace();
-}
-
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestOpusNewACM)) {
- Trace::CreateTrace();
- Trace::SetTraceFile((webrtc::test::OutputPath() +
- "acm_opus_trace_new.txt").c_str());
- webrtc::Config config;
-
- UseNewAcm(&config);
- webrtc::OpusTest(config).Perform();
+ webrtc::OpusTest().Perform();
Trace::ReturnTrace();
}
@@ -218,14 +104,7 @@
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_apitest_trace.txt").c_str());
- webrtc::Config config;
-
- UseLegacyAcm(&config);
- webrtc::APITest(config).Perform();
-
- UseNewAcm(&config);
- webrtc::APITest(config).Perform();
-
+ webrtc::APITest().Perform();
Trace::ReturnTrace();
}
#endif
diff --git a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc b/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
index fb3d6f4..81ef0c3 100644
--- a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
+++ b/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
@@ -20,7 +20,6 @@
#include "gtest/gtest.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
@@ -31,12 +30,12 @@
#define MAX_FILE_NAME_LENGTH_BYTE 500
-TwoWayCommunication::TwoWayCommunication(int testMode, const Config& config)
- : _acmA(config.Get<AudioCodingModuleFactory>().Create(1)),
- _acmB(config.Get<AudioCodingModuleFactory>().Create(2)),
- _acmRefA(config.Get<AudioCodingModuleFactory>().Create(3)),
- _acmRefB(config.Get<AudioCodingModuleFactory>().Create(4)),
- _testMode(testMode) { }
+TwoWayCommunication::TwoWayCommunication(int testMode)
+ : _acmA(AudioCodingModule::Create(1)),
+ _acmB(AudioCodingModule::Create(2)),
+ _acmRefA(AudioCodingModule::Create(3)),
+ _acmRefB(AudioCodingModule::Create(4)),
+ _testMode(testMode) {}
TwoWayCommunication::~TwoWayCommunication() {
delete _channel_A2B;
diff --git a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.h b/webrtc/modules/audio_coding/main/test/TwoWayCommunication.h
index 0d1e514..9e0b724 100644
--- a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.h
+++ b/webrtc/modules/audio_coding/main/test/TwoWayCommunication.h
@@ -20,11 +20,9 @@
namespace webrtc {
-class Config;
-
class TwoWayCommunication : public ACMTest {
public:
- TwoWayCommunication(int testMode, const Config& config);
+ explicit TwoWayCommunication(int testMode);
~TwoWayCommunication();
void Perform();
diff --git a/webrtc/modules/audio_coding/main/test/delay_test.cc b/webrtc/modules/audio_coding/main/test/delay_test.cc
index 63bfe2b..ba81507 100644
--- a/webrtc/modules/audio_coding/main/test/delay_test.cc
+++ b/webrtc/modules/audio_coding/main/test/delay_test.cc
@@ -35,7 +35,6 @@
DEFINE_int32(delay, 0, "Delay in millisecond.");
DEFINE_int32(init_delay, 0, "Initial delay in millisecond.");
DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
-DEFINE_bool(acm2, false, "Run the test with ACM2.");
DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
@@ -64,9 +63,9 @@
class DelayTest {
public:
- explicit DelayTest(const Config& config)
- : acm_a_(config.Get<AudioCodingModuleFactory>().Create(0)),
- acm_b_(config.Get<AudioCodingModuleFactory>().Create(1)),
+ DelayTest()
+ : acm_a_(AudioCodingModule::Create(0)),
+ acm_b_(AudioCodingModule::Create(1)),
channel_a2b_(new Channel),
test_cntr_(0),
encoding_sample_rate_hz_(8000) {}
@@ -245,7 +244,6 @@
int main(int argc, char* argv[]) {
google::ParseCommandLineFlags(&argc, &argv, true);
- webrtc::Config config;
webrtc::TestSettings test_setting;
strcpy(test_setting.codec.name, FLAGS_codec.c_str());
@@ -266,13 +264,7 @@
test_setting.acm.fec = FLAGS_fec;
test_setting.packet_loss = FLAGS_packet_loss;
- if (FLAGS_acm2) {
- webrtc::UseNewAcm(&config);
- } else {
- webrtc::UseLegacyAcm(&config);
- }
-
- webrtc::DelayTest delay_test(config);
+ webrtc::DelayTest delay_test;
delay_test.Initialize();
delay_test.Perform(&test_setting, 1, 240, "delay_test");
return 0;
diff --git a/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc b/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc
index ba9bb6c..71657c9 100644
--- a/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc
+++ b/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc
@@ -9,7 +9,6 @@
*/
#include "gtest/gtest.h"
-#include "webrtc/common.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
@@ -22,9 +21,10 @@
namespace webrtc {
-class DualStreamTest : public AudioPacketizationCallback {
- public:
- explicit DualStreamTest(const Config& config);
+class DualStreamTest : public AudioPacketizationCallback,
+ public ::testing::Test {
+ protected:
+ DualStreamTest();
~DualStreamTest();
void RunTest(int frame_size_primary_samples,
@@ -35,8 +35,6 @@
void ApiTest();
- protected:
-
int32_t SendData(FrameType frameType, uint8_t payload_type,
uint32_t timestamp, const uint8_t* payload_data,
uint16_t payload_size,
@@ -93,10 +91,10 @@
bool received_payload_[kMaxNumStreams];
};
-DualStreamTest::DualStreamTest(const Config& config)
- : acm_dual_stream_(config.Get<AudioCodingModuleFactory>().Create(0)),
- acm_ref_primary_(config.Get<AudioCodingModuleFactory>().Create(1)),
- acm_ref_secondary_(config.Get<AudioCodingModuleFactory>().Create(2)),
+DualStreamTest::DualStreamTest()
+ : acm_dual_stream_(AudioCodingModule::Create(0)),
+ acm_ref_primary_(AudioCodingModule::Create(1)),
+ acm_ref_secondary_(AudioCodingModule::Create(2)),
payload_ref_is_stored_(),
payload_dual_is_stored_(),
timestamp_ref_(),
@@ -388,17 +386,106 @@
return 0;
}
-void DualStreamTest::RunTest(int frame_size_primary_samples,
- int num_channels_primary,
- int sampling_rate,
- bool start_in_sync,
- int num_channels_input) {
- InitializeSender(
- frame_size_primary_samples, num_channels_primary, sampling_rate);
- Perform(start_in_sync, num_channels_input);
-};
+// Mono input, mono primary WB 20 ms frame.
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactSyncMonoInputMonoPrimaryWb20Ms)) {
+ InitializeSender(20, 1, 16000);
+ Perform(true, 1);
+}
-void DualStreamTest::ApiTest() {
+// Mono input, stereo primary WB 20 ms frame.
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactSyncMonoInput_StereoPrimaryWb20Ms)) {
+ InitializeSender(20, 2, 16000);
+ Perform(true, 1);
+}
+
+// Mono input, mono primary SWB 20 ms frame.
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactSyncMonoInputMonoPrimarySwb20Ms)) {
+ InitializeSender(20, 1, 32000);
+ Perform(true, 1);
+}
+
+// Mono input, stereo primary SWB 20 ms frame.
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactSyncMonoInputStereoPrimarySwb20Ms)) {
+ InitializeSender(20, 2, 32000);
+ Perform(true, 1);
+}
+
+// Mono input, mono primary WB 40 ms frame.
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactSyncMonoInputMonoPrimaryWb40Ms)) {
+ InitializeSender(40, 1, 16000);
+ Perform(true, 1);
+}
+
+// Mono input, stereo primary WB 40 ms frame
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactSyncMonoInputStereoPrimaryWb40Ms)) {
+ InitializeSender(40, 2, 16000);
+ Perform(true, 1);
+}
+
+// Stereo input, mono primary WB 20 ms frame.
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactSyncStereoInputMonoPrimaryWb20Ms)) {
+ InitializeSender(20, 1, 16000);
+ Perform(true, 2);
+}
+
+// Stereo input, stereo primary WB 20 ms frame.
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactSyncStereoInputStereoPrimaryWb20Ms)) {
+ InitializeSender(20, 2, 16000);
+ Perform(true, 2);
+}
+
+// Stereo input, mono primary SWB 20 ms frame.
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactSyncStereoInputMonoPrimarySwb20Ms)) {
+ InitializeSender(20, 1, 32000);
+ Perform(true, 2);
+}
+
+// Stereo input, stereo primary SWB 20 ms frame.
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactSyncStereoInputStereoPrimarySwb20Ms)) {
+ InitializeSender(20, 2, 32000);
+ Perform(true, 2);
+}
+
+// Stereo input, mono primary WB 40 ms frame.
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactSyncStereoInputMonoPrimaryWb40Ms)) {
+ InitializeSender(40, 1, 16000);
+ Perform(true, 2);
+}
+
+// Stereo input, stereo primary WB 40 ms frame.
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactSyncStereoInputStereoPrimaryWb40Ms)) {
+ InitializeSender(40, 2, 16000);
+ Perform(true, 2);
+}
+
+// Asynchronous test, ACM is fed with data then secondary coder is registered.
+// Mono input, mono primary WB 20 ms frame.
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactAsyncMonoInputMonoPrimaryWb20Ms)) {
+ InitializeSender(20, 1, 16000);
+ Perform(false, 1);
+}
+
+// Mono input, mono primary WB 20 ms frame.
+TEST_F(DualStreamTest,
+ DISABLED_ON_ANDROID(BitExactAsyncMonoInputMonoPrimaryWb40Ms)) {
+ InitializeSender(40, 1, 16000);
+ Perform(false, 1);
+}
+
+TEST_F(DualStreamTest, DISABLED_ON_ANDROID(Api)) {
PopulateCodecInstances(20, 1, 16000);
CodecInst my_codec;
ASSERT_EQ(0, acm_dual_stream_->InitializeSender());
@@ -449,171 +536,4 @@
EXPECT_EQ(VADVeryAggr, vad_mode);
}
-namespace {
-
-DualStreamTest* CreateLegacy() {
- Config config;
- UseLegacyAcm(&config);
- DualStreamTest* test = new DualStreamTest(config);
- return test;
-}
-
-DualStreamTest* CreateNew() {
- Config config;
- UseNewAcm(&config);
- DualStreamTest* test = new DualStreamTest(config);
- return test;
-}
-
-} // namespace
-
-// Mono input, mono primary WB 20 ms frame.
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactSyncMonoInputMonoPrimaryWb20Ms)) {
- scoped_ptr<DualStreamTest> test(CreateLegacy());
- test->RunTest(20, 1, 16000, true, 1);
-
- test.reset(CreateNew());
- test->RunTest(20, 1, 16000, true, 1);
-}
-
-// Mono input, stereo primary WB 20 ms frame.
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactSyncMonoInput_StereoPrimaryWb20Ms)) {
- scoped_ptr<DualStreamTest> test(CreateLegacy());
- test->RunTest(20, 2, 16000, true, 1);
-
- test.reset(CreateNew());
- test->RunTest(20, 2, 16000, true, 1);
-}
-
-// Mono input, mono primary SWB 20 ms frame.
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactSyncMonoInputMonoPrimarySwb20Ms)) {
- scoped_ptr<DualStreamTest> test(CreateLegacy());
- test->RunTest(20, 1, 32000, true, 1);
-
- test.reset(CreateNew());
- test->RunTest(20, 1, 32000, true, 1);
-}
-
-// Mono input, stereo primary SWB 20 ms frame.
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactSyncMonoInputStereoPrimarySwb20Ms)) {
- scoped_ptr<DualStreamTest> test(CreateLegacy());
- test->RunTest(20, 2, 32000, true, 1);
-
- test.reset(CreateNew());
- test->RunTest(20, 2, 32000, true, 1);
-}
-
-// Mono input, mono primary WB 40 ms frame.
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactSyncMonoInputMonoPrimaryWb40Ms)) {
- scoped_ptr<DualStreamTest> test(CreateNew());
- test->RunTest(40, 1, 16000, true, 1);
-
- test.reset(CreateNew());
- test->RunTest(40, 1, 16000, true, 1);
-}
-
-// Mono input, stereo primary WB 40 ms frame
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactSyncMonoInputStereoPrimaryWb40Ms)) {
- scoped_ptr<DualStreamTest> test(CreateNew());
- test->RunTest(40, 2, 16000, true, 1);
-
- test.reset(CreateNew());
- test->RunTest(40, 2, 16000, true, 1);
-}
-
-// Stereo input, mono primary WB 20 ms frame.
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactSyncStereoInputMonoPrimaryWb20Ms)) {
- scoped_ptr<DualStreamTest> test(CreateLegacy());
- test->RunTest(20, 1, 16000, true, 2);
-
- test.reset(CreateNew());
- test->RunTest(20, 1, 16000, true, 2);
-}
-
-// Stereo input, stereo primary WB 20 ms frame.
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactSyncStereoInputStereoPrimaryWb20Ms)) {
- scoped_ptr<DualStreamTest> test(CreateLegacy());
- test->RunTest(20, 2, 16000, true, 2);
-
- test.reset(CreateNew());
- test->RunTest(20, 2, 16000, true, 2);
-}
-
-// Stereo input, mono primary SWB 20 ms frame.
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactSyncStereoInputMonoPrimarySwb20Ms)) {
- scoped_ptr<DualStreamTest> test(CreateLegacy());
- test->RunTest(20, 1, 32000, true, 2);
-
- test.reset(CreateNew());
- test->RunTest(20, 1, 32000, true, 2);
-}
-
-// Stereo input, stereo primary SWB 20 ms frame.
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactSyncStereoInputStereoPrimarySwb20Ms)) {
- scoped_ptr<DualStreamTest> test(CreateLegacy());
- test->RunTest(20, 2, 32000, true, 2);
-
- test.reset(CreateNew());
- test->RunTest(20, 2, 32000, true, 2);
-}
-
-// Stereo input, mono primary WB 40 ms frame.
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactSyncStereoInputMonoPrimaryWb40Ms)) {
- scoped_ptr<DualStreamTest> test(CreateLegacy());
- test->RunTest(40, 1, 16000, true, 2);
-
- test.reset(CreateNew());
- test->RunTest(40, 1, 16000, true, 2);
-}
-
-// Stereo input, stereo primary WB 40 ms frame.
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactSyncStereoInputStereoPrimaryWb40Ms)) {
- scoped_ptr<DualStreamTest> test(CreateLegacy());
- test->RunTest(40, 2, 16000, true, 2);
-
- test.reset(CreateNew());
- test->RunTest(40, 2, 16000, true, 2);
-}
-
-// Asynchronous test, ACM is fed with data then secondary coder is registered.
-// Mono input, mono primary WB 20 ms frame.
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactAsyncMonoInputMonoPrimaryWb20Ms)) {
- scoped_ptr<DualStreamTest> test(CreateLegacy());
- test->RunTest(20, 1, 16000, false, 1);
-
- test.reset(CreateNew());
- test->RunTest(20, 1, 16000, false, 1);
-}
-
-// Mono input, mono primary WB 20 ms frame.
-TEST(DualStreamTest,
- DISABLED_ON_ANDROID(BitExactAsyncMonoInputMonoPrimaryWb40Ms)) {
- scoped_ptr<DualStreamTest> test(CreateLegacy());
- test->RunTest(40, 1, 16000, false, 1);
-
- test.reset(CreateNew());
- test->RunTest(40, 1, 16000, false, 1);
-}
-
-TEST(DualStreamTest, DISABLED_ON_ANDROID(ApiTest)) {
- scoped_ptr<DualStreamTest> test(CreateLegacy());
- test->ApiTest();
-
- test.reset(CreateNew());
- test->ApiTest();
-}
-
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.cc b/webrtc/modules/audio_coding/main/test/iSACTest.cc
index f7fef4a..c5da92e 100644
--- a/webrtc/modules/audio_coding/main/test/iSACTest.cc
+++ b/webrtc/modules/audio_coding/main/test/iSACTest.cc
@@ -86,11 +86,10 @@
return 0;
}
-ISACTest::ISACTest(int testMode, const Config& config)
- : _acmA(config.Get<AudioCodingModuleFactory>().Create(1)),
- _acmB(config.Get<AudioCodingModuleFactory>().Create(2)),
- _testMode(testMode) {
-}
+ISACTest::ISACTest(int testMode)
+ : _acmA(AudioCodingModule::Create(1)),
+ _acmB(AudioCodingModule::Create(2)),
+ _testMode(testMode) {}
ISACTest::~ISACTest() {}
diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.h b/webrtc/modules/audio_coding/main/test/iSACTest.h
index d9563db..9fe6aff 100644
--- a/webrtc/modules/audio_coding/main/test/iSACTest.h
+++ b/webrtc/modules/audio_coding/main/test/iSACTest.h
@@ -13,7 +13,6 @@
#include <string.h>
-#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
@@ -27,8 +26,6 @@
namespace webrtc {
-class Config;
-
struct ACMTestISACConfig {
int32_t currentRateBitPerSec;
int16_t currentFrameSizeMsec;
@@ -42,7 +39,7 @@
class ISACTest : public ACMTest {
public:
- ISACTest(int testMode, const Config& config);
+ explicit ISACTest(int testMode);
~ISACTest();
void Perform();
diff --git a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
index 87fed6c..192539d 100644
--- a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
+++ b/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
@@ -16,7 +16,6 @@
#include <iostream>
#include "gtest/gtest.h"
-#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
@@ -44,11 +43,11 @@
}
-class InitialPlayoutDelayTest {
- public:
- explicit InitialPlayoutDelayTest(const Config& config)
- : acm_a_(config.Get<AudioCodingModuleFactory>().Create(0)),
- acm_b_(config.Get<AudioCodingModuleFactory>().Create(1)),
+class InitialPlayoutDelayTest : public ::testing::Test {
+ protected:
+ InitialPlayoutDelayTest()
+ : acm_a_(AudioCodingModule::Create(0)),
+ acm_b_(AudioCodingModule::Create(1)),
channel_a2b_(NULL) {}
~InitialPlayoutDelayTest() {
@@ -162,72 +161,16 @@
Channel* channel_a2b_;
};
-namespace {
+TEST_F(InitialPlayoutDelayTest, NbMono) { NbMono(); }
-InitialPlayoutDelayTest* CreateLegacy() {
- Config config;
- UseLegacyAcm(&config);
- InitialPlayoutDelayTest* test = new InitialPlayoutDelayTest(config);
- test->SetUp();
- return test;
-}
+TEST_F(InitialPlayoutDelayTest, WbMono) { WbMono(); }
-InitialPlayoutDelayTest* CreateNew() {
- Config config;
- UseNewAcm(&config);
- InitialPlayoutDelayTest* test = new InitialPlayoutDelayTest(config);
- test->SetUp();
- return test;
-}
+TEST_F(InitialPlayoutDelayTest, SwbMono) { SwbMono(); }
-} // namespace
+TEST_F(InitialPlayoutDelayTest, NbStereo) { NbStereo(); }
-TEST(InitialPlayoutDelayTest, NbMono) {
- scoped_ptr<InitialPlayoutDelayTest> test(CreateLegacy());
- test->NbMono();
+TEST_F(InitialPlayoutDelayTest, WbStereo) { WbStereo(); }
- test.reset(CreateNew());
- test->NbMono();
-}
-
-TEST(InitialPlayoutDelayTest, WbMono) {
- scoped_ptr<InitialPlayoutDelayTest> test(CreateLegacy());
- test->WbMono();
-
- test.reset(CreateNew());
- test->WbMono();
-}
-
-TEST(InitialPlayoutDelayTest, SwbMono) {
- scoped_ptr<InitialPlayoutDelayTest> test(CreateLegacy());
- test->SwbMono();
-
- test.reset(CreateNew());
- test->SwbMono();
-}
-
-TEST(InitialPlayoutDelayTest, NbStereo) {
- scoped_ptr<InitialPlayoutDelayTest> test(CreateLegacy());
- test->NbStereo();
-
- test.reset(CreateNew());
- test->NbStereo();
-}
-
-TEST(InitialPlayoutDelayTest, WbStereo) {
- scoped_ptr<InitialPlayoutDelayTest> test(CreateLegacy());
- test->WbStereo();
-
- test.reset(CreateNew());
- test->WbStereo();
-}
-
-TEST(InitialPlayoutDelayTest, SwbStereo) {
- scoped_ptr<InitialPlayoutDelayTest> test(CreateLegacy());
- test->SwbStereo();
-
- test.reset(CreateNew());
- test->SwbStereo();
-}
+TEST_F(InitialPlayoutDelayTest, SwbStereo) { SwbStereo(); }
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.cc b/webrtc/modules/audio_coding/main/test/opus_test.cc
index 027aeb0..230e9f1 100644
--- a/webrtc/modules/audio_coding/main/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/main/test/opus_test.cc
@@ -15,13 +15,12 @@
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common.h" // Config.
#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/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
@@ -29,13 +28,12 @@
namespace webrtc {
-OpusTest::OpusTest(const Config& config)
- : acm_receiver_(config.Get<AudioCodingModuleFactory>().Create(0)),
+OpusTest::OpusTest()
+ : acm_receiver_(AudioCodingModule::Create(0)),
channel_a2b_(NULL),
counter_(0),
payload_type_(255),
- rtp_timestamp_(0) {
-}
+ rtp_timestamp_(0) {}
OpusTest::~OpusTest() {
if (channel_a2b_ != NULL) {
@@ -254,11 +252,12 @@
}
// If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
- EXPECT_EQ(480, resampler_.Resample10Msec(audio_frame.data_,
- audio_frame.sample_rate_hz_,
- &audio[written_samples],
- 48000,
- channels));
+ EXPECT_EQ(480,
+ resampler_.Resample10Msec(audio_frame.data_,
+ audio_frame.sample_rate_hz_,
+ 48000,
+ channels,
+ &audio[written_samples]));
written_samples += 480 * channels;
// Sometimes we need to loop over the audio vector to produce the right
diff --git a/webrtc/modules/audio_coding/main/test/opus_test.h b/webrtc/modules/audio_coding/main/test/opus_test.h
index 08dce98..9ee2b93 100644
--- a/webrtc/modules/audio_coding/main/test/opus_test.h
+++ b/webrtc/modules/audio_coding/main/test/opus_test.h
@@ -13,8 +13,8 @@
#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/acm2/acm_opus.h"
+#include "webrtc/modules/audio_coding/main/acm2/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"
@@ -23,11 +23,9 @@
namespace webrtc {
-class Config;
-
class OpusTest : public ACMTest {
public:
- explicit OpusTest(const Config& config);
+ OpusTest();
~OpusTest();
void Perform();
@@ -47,7 +45,7 @@
int counter_;
uint8_t payload_type_;
int rtp_timestamp_;
- acm1::ACMResampler resampler_;
+ acm2::ACMResampler resampler_;
WebRtcOpusEncInst* opus_mono_encoder_;
WebRtcOpusEncInst* opus_stereo_encoder_;
WebRtcOpusDecInst* opus_mono_decoder_;
diff --git a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
index f01e6ff..5636bdf 100644
--- a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
+++ b/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
@@ -9,7 +9,6 @@
*/
#include "gtest/gtest.h"
-#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
@@ -22,11 +21,9 @@
namespace webrtc {
-
-class TargetDelayTest {
- public:
- explicit TargetDelayTest(const Config& config)
- : acm_(config.Get<AudioCodingModuleFactory>().Create(0)) {}
+class TargetDelayTest : public ::testing::Test {
+ protected:
+ TargetDelayTest() : acm_(AudioCodingModule::Create(0)) {}
~TargetDelayTest() {}
@@ -202,65 +199,24 @@
uint8_t payload_[kPayloadLenBytes];
};
-
-namespace {
-
-TargetDelayTest* CreateLegacy() {
- Config config;
- UseLegacyAcm(&config);
- TargetDelayTest* test = new TargetDelayTest(config);
- test->SetUp();
- return test;
+TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(OutOfRangeInput)) {
+ OutOfRangeInput();
}
-TargetDelayTest* CreateNew() {
- Config config;
- UseNewAcm(&config);
- TargetDelayTest* test = new TargetDelayTest(config);
- test->SetUp();
- return test;
+TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(NoTargetDelayBufferSizeChanges)) {
+ NoTargetDelayBufferSizeChanges();
}
-} // namespace
-
-TEST(TargetDelayTest, DISABLED_ON_ANDROID(OutOfRangeInput)) {
- scoped_ptr<TargetDelayTest> test(CreateLegacy());
- test->OutOfRangeInput();
-
- test.reset(CreateNew());
- test->OutOfRangeInput();
+TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(WithTargetDelayBufferNotChanging)) {
+ WithTargetDelayBufferNotChanging();
}
-TEST(TargetDelayTest, DISABLED_ON_ANDROID(NoTargetDelayBufferSizeChanges)) {
- scoped_ptr<TargetDelayTest> test(CreateLegacy());
- test->NoTargetDelayBufferSizeChanges();
-
- test.reset(CreateNew());
- test->NoTargetDelayBufferSizeChanges();
+TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(RequiredDelayAtCorrectRange)) {
+ RequiredDelayAtCorrectRange();
}
-TEST(TargetDelayTest, DISABLED_ON_ANDROID(WithTargetDelayBufferNotChanging)) {
- scoped_ptr<TargetDelayTest> test(CreateLegacy());
- test->WithTargetDelayBufferNotChanging();
-
- test.reset(CreateNew());
- test->WithTargetDelayBufferNotChanging();
-}
-
-TEST(TargetDelayTest, DISABLED_ON_ANDROID(RequiredDelayAtCorrectRange)) {
- scoped_ptr<TargetDelayTest> test(CreateLegacy());
- test->RequiredDelayAtCorrectRange();
-
- test.reset(CreateNew());
- test->RequiredDelayAtCorrectRange();
-}
-
-TEST(TargetDelayTest, DISABLED_ON_ANDROID(TargetDelayBufferMinMax)) {
- scoped_ptr<TargetDelayTest> test(CreateLegacy());
- test->TargetDelayBufferMinMax();
-
- test.reset(CreateNew());
- test->TargetDelayBufferMinMax();
+TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(TargetDelayBufferMinMax)) {
+ TargetDelayBufferMinMax();
}
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/test/utility.cc b/webrtc/modules/audio_coding/main/test/utility.cc
index d6441ac..0848954 100644
--- a/webrtc/modules/audio_coding/main/test/utility.cc
+++ b/webrtc/modules/audio_coding/main/test/utility.cc
@@ -330,14 +330,4 @@
return 0;
}
-void UseLegacyAcm(webrtc::Config* config) {
- config->Set<webrtc::AudioCodingModuleFactory>(
- new webrtc::AudioCodingModuleFactory());
-}
-
-void UseNewAcm(webrtc::Config* config) {
- config->Set<webrtc::AudioCodingModuleFactory>(
- new webrtc::NewAudioCodingModuleFactory());
-}
-
} // namespace webrtc