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diff --git a/webrtc/modules/audio_coding/test/Tester.cc b/webrtc/modules/audio_coding/test/Tester.cc
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+/*
+ * 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/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/APITest.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/test/iSACTest.h"
+#include "webrtc/modules/audio_coding/test/opus_test.h"
+#include "webrtc/modules/audio_coding/test/PacketLossTest.h"
+#include "webrtc/modules/audio_coding/test/TestAllCodecs.h"
+#include "webrtc/modules/audio_coding/test/TestRedFec.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/test/TestVADDTX.h"
+#include "webrtc/modules/audio_coding/test/TwoWayCommunication.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/testsupport/fileutils.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();
+}
+
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestEncodeDecode) {
+#else
+TEST(AudioCodingModuleTest, TestEncodeDecode) {
+#endif
+ Trace::CreateTrace();
+ Trace::SetTraceFile((webrtc::test::OutputPath() +
+ "acm_encodedecode_trace.txt").c_str());
+ webrtc::EncodeDecodeTest(ACM_TEST_MODE).Perform();
+ Trace::ReturnTrace();
+}
+
+#if defined(WEBRTC_CODEC_RED)
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestRedFec) {
+#else
+TEST(AudioCodingModuleTest, TestRedFec) {
+#endif
+ Trace::CreateTrace();
+ Trace::SetTraceFile((webrtc::test::OutputPath() +
+ "acm_fec_trace.txt").c_str());
+ webrtc::TestRedFec().Perform();
+ Trace::ReturnTrace();
+}
+#endif
+
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestIsac) {
+#else
+TEST(AudioCodingModuleTest, TestIsac) {
+#endif
+ Trace::CreateTrace();
+ Trace::SetTraceFile((webrtc::test::OutputPath() +
+ "acm_isac_trace.txt").c_str());
+ webrtc::ISACTest(ACM_TEST_MODE).Perform();
+ Trace::ReturnTrace();
+}
+#endif
+
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
+ defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TwoWayCommunication) {
+#else
+TEST(AudioCodingModuleTest, TwoWayCommunication) {
+#endif
+ Trace::CreateTrace();
+ Trace::SetTraceFile((webrtc::test::OutputPath() +
+ "acm_twowaycom_trace.txt").c_str());
+ webrtc::TwoWayCommunication(ACM_TEST_MODE).Perform();
+ Trace::ReturnTrace();
+}
+#endif
+
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestStereo) {
+#else
+TEST(AudioCodingModuleTest, TestStereo) {
+#endif
+ Trace::CreateTrace();
+ Trace::SetTraceFile((webrtc::test::OutputPath() +
+ "acm_stereo_trace.txt").c_str());
+ webrtc::TestStereo(ACM_TEST_MODE).Perform();
+ Trace::ReturnTrace();
+}
+
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestWebRtcVadDtx) {
+#else
+TEST(AudioCodingModuleTest, TestWebRtcVadDtx) {
+#endif
+ 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