session-android/jni/webrtc/modules/audio_coding/main/test/Tester.cc
Moxie Marlinspike d83a3d71bc Support for Signal calls.
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

144 lines
4.8 KiB
C++

/*
* 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.
*/
#include <stdio.h>
#include <string>
#include <vector>
#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/PacketLossTest.h"
#include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h"
#include "webrtc/modules/audio_coding/main/test/TestRedFec.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"
#include "webrtc/test/testsupport/gtest_disable.h"
using webrtc::Trace;
// This parameter is used to describe how to run the tests. It is normally
// set to 0, and all tests are run in quite mode.
#define ACM_TEST_MODE 0
TEST(AudioCodingModuleTest, TestAllCodecs) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_allcodecs_trace.txt").c_str());
webrtc::TestAllCodecs(ACM_TEST_MODE).Perform();
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_encodedecode_trace.txt").c_str());
webrtc::EncodeDecodeTest(ACM_TEST_MODE).Perform();
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestRedFec)) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_fec_trace.txt").c_str());
webrtc::TestRedFec().Perform();
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestIsac)) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_isac_trace.txt").c_str());
webrtc::ISACTest(ACM_TEST_MODE).Perform();
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TwoWayCommunication)) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_twowaycom_trace.txt").c_str());
webrtc::TwoWayCommunication(ACM_TEST_MODE).Perform();
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestStereo)) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_stereo_trace.txt").c_str());
webrtc::TestStereo(ACM_TEST_MODE).Perform();
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestVADDTX)) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_vaddtx_trace.txt").c_str());
webrtc::TestVADDTX().Perform();
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, TestOpus) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_opus_trace.txt").c_str());
webrtc::OpusTest().Perform();
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, TestPacketLoss) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_packetloss_trace.txt").c_str());
webrtc::PacketLossTest(1, 10, 10, 1).Perform();
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, TestPacketLossBurst) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_packetloss_burst_trace.txt").c_str());
webrtc::PacketLossTest(1, 10, 10, 2).Perform();
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, TestPacketLossStereo) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_packetloss_trace.txt").c_str());
webrtc::PacketLossTest(2, 10, 10, 1).Perform();
Trace::ReturnTrace();
}
TEST(AudioCodingModuleTest, TestPacketLossStereoBurst) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_packetloss_burst_trace.txt").c_str());
webrtc::PacketLossTest(2, 10, 10, 2).Perform();
Trace::ReturnTrace();
}
// The full API test is too long to run automatically on bots, but can be used
// for offline testing. User interaction is needed.
#ifdef ACM_TEST_FULL_API
TEST(AudioCodingModuleTest, TestAPI) {
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_apitest_trace.txt").c_str());
webrtc::APITest().Perform();
Trace::ReturnTrace();
}
#endif