audio_coding: remove "main" directory

webrtc/modules/audio_coding/main/test/Tester.cc

-/*
- *  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/include/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/include/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();
-}
-
-#ifdef WEBRTC_CODEC_RED
-#define IF_RED(x) x
-#else
-#define IF_RED(x) DISABLED_##x
-#endif
-
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(IF_RED(TestRedFec))) {
-  Trace::CreateTrace();
-  Trace::SetTraceFile((webrtc::test::OutputPath() +
-      "acm_fec_trace.txt").c_str());
-  webrtc::TestRedFec().Perform();
-  Trace::ReturnTrace();
-}
-
-#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-#define IF_ISAC(x) x
-#else
-#define IF_ISAC(x) DISABLED_##x
-#endif
-
-TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(IF_ISAC(TestIsac))) {
-  Trace::CreateTrace();
-  Trace::SetTraceFile((webrtc::test::OutputPath() +
-      "acm_isac_trace.txt").c_str());
-  webrtc::ISACTest(ACM_TEST_MODE).Perform();
-  Trace::ReturnTrace();
-}
-
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
-    defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
-#define IF_ALL_CODECS(x) x
-#else
-#define IF_ALL_CODECS(x) DISABLED_##x
-#endif
-
-TEST(AudioCodingModuleTest,
-     DISABLED_ON_ANDROID(IF_ALL_CODECS(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(TestWebRtcVadDtx)) {
-  Trace::CreateTrace();
-  Trace::SetTraceFile((webrtc::test::OutputPath() +
-      "acm_vaddtx_trace.txt").c_str());
-  webrtc::TestWebRtcVadDtx().Perform();
-  Trace::ReturnTrace();
-}
-
-TEST(AudioCodingModuleTest, TestOpusDtx) {
-  Trace::CreateTrace();
-  Trace::SetTraceFile((webrtc::test::OutputPath() +
-      "acm_opusdtx_trace.txt").c_str());
-  webrtc::TestOpusDtx().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