diff --git a/webrtc/modules/audio_coding/main/test/APITest.cc b/webrtc/modules/audio_coding/main/test/APITest.cc index 15bac6adc..82940fa3e 100644 --- a/webrtc/modules/audio_coding/main/test/APITest.cc +++ b/webrtc/modules/audio_coding/main/test/APITest.cc @@ -56,8 +56,8 @@ void APITest::Wait(uint32_t waitLengthMs) { } 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 cdf9fdcae..1ee6abc30 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 @@ void Receiver::Run() { } } -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 @@ void EncodeDecodeTest::Perform() { 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::Perform() { 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 5aa359636..4fdd943cf 100644 --- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h +++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h @@ -23,8 +23,6 @@ namespace webrtc { #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 Receiver { 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 @@ class EncodeDecodeTest : public ACMTest { 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 fba7f0329..d6c6dc4e6 100644 --- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc +++ b/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc @@ -99,9 +99,9 @@ void TestPack::reset_payload_size() { 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 0231d84c6..10d82ae1c 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 TestPack : public AudioPacketizationCallback { 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 032579cf0..76b6d4bf3 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 af3cdd7dc..d7a62234d 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 b26334c32..f05896773 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 @@ void TestPackStereo::set_lost_packet(bool lost) { 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 88320a0e5..03f80411b 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 TestPackStereo : public AudioPacketizationCallback { 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 22e9696ff..d31e1d47a 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 e0aa6b813..f8c97e127 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 ActivityMonitor : public ACMVADCallback { 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 581b7bd31..a89c9cd40 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 @@ TEST(AudioCodingModuleTest, TestAllCodecs) { 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 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) { 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 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestFEC)) { 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 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestIsac)) { 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 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TwoWayCommunication)) { 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 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestStereo)) { 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 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestVADDTX)) { 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 @@ TEST(AudioCodingModuleTest, TestOpus) { 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 @@ TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestOpusNewACM)) { 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 fb3d6f487..81ef0c3ff 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 @@ namespace webrtc { #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 0d1e514da..9e0b72498 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 63bfe2be4..ba81507dd 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_string(input_file, "", "Input file, PCM16 32 kHz, optional."); 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 @@ struct TestSettings { 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 @@ class DelayTest { 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 @@ int main(int argc, char* argv[]) { 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 ba9bb6cb3..71657c9f4 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 @@ class DualStreamTest : public AudioPacketizationCallback { 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 @@ class DualStreamTest : public AudioPacketizationCallback { 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 @@ int32_t DualStreamTest::SendData(FrameType frameType, uint8_t payload_type, 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 @@ void DualStreamTest::ApiTest() { 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 f7fef4a80..c5da92e1e 100644 --- a/webrtc/modules/audio_coding/main/test/iSACTest.cc +++ b/webrtc/modules/audio_coding/main/test/iSACTest.cc @@ -86,11 +86,10 @@ int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm, 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 d9563dbd9..9fe6afffa 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 @@ struct ACMTestISACConfig { 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 87fed6ca8..192539d85 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 @@ double FrameRms(AudioFrame& frame) { } -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 @@ class InitialPlayoutDelayTest { 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 027aeb045..230e9f1a1 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 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, } // 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 08dce98a1..9ee2b9372 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 @@ class OpusTest : public ACMTest { 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 f01e6ffba..5636bdf80 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 @@ class TargetDelayTest { 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 d6441ac6b..084895446 100644 --- a/webrtc/modules/audio_coding/main/test/utility.cc +++ b/webrtc/modules/audio_coding/main/test/utility.cc @@ -330,14 +330,4 @@ int32_t VADCallback::InFrameType(int16_t frameType) { 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