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Unified Diff: webrtc/modules/audio_coding/main/test/Tester.cc

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years, 1 month ago
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Index: webrtc/modules/audio_coding/main/test/Tester.cc
diff --git a/webrtc/modules/audio_coding/main/test/Tester.cc b/webrtc/modules/audio_coding/main/test/Tester.cc
deleted file mode 100644
index 7302e5dcbe756e30cb11691fa23731698d8cd3fa..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/test/Tester.cc
+++ /dev/null
@@ -1,171 +0,0 @@
-/*
- * 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
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