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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 "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/EncodeDecodeTest.h"
#include "modules/audio_coding/test/PacketLossTest.h"
#include "modules/audio_coding/test/TestAllCodecs.h"
#include "modules/audio_coding/test/TestRedFec.h"
#include "modules/audio_coding/test/TestStereo.h"
#include "modules/audio_coding/test/TestVADDTX.h"
#include "modules/audio_coding/test/opus_test.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
TEST(AudioCodingModuleTest, TestAllCodecs) {
webrtc::TestAllCodecs().Perform();
}
#if defined(WEBRTC_ANDROID)
TEST(AudioCodingModuleTest, DISABLED_TestEncodeDecode) {
#else
TEST(AudioCodingModuleTest, TestEncodeDecode) {
#endif
webrtc::EncodeDecodeTest().Perform();
}
TEST(AudioCodingModuleTest, TestRedFec) {
webrtc::TestRedFec().Perform();
}
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
TEST(AudioCodingModuleTest, DISABLED_TestStereo) {
#else
TEST(AudioCodingModuleTest, TestStereo) {
#endif
webrtc::TestStereo().Perform();
}
TEST(AudioCodingModuleTest, TestWebRtcVadDtx) {
webrtc::TestWebRtcVadDtx().Perform();
}
TEST(AudioCodingModuleTest, TestOpusDtx) {
webrtc::TestOpusDtx().Perform();
}
#if defined(WEBRTC_IOS)
TEST(AudioCodingModuleTest, DISABLED_TestOpus) {
#else
TEST(AudioCodingModuleTest, TestOpus) {
#endif
webrtc::OpusTest().Perform();
}
TEST(AudioCodingModuleTest, TestPacketLoss) {
webrtc::PacketLossTest(1, 10, 10, 1).Perform();
}
TEST(AudioCodingModuleTest, TestPacketLossBurst) {
webrtc::PacketLossTest(1, 10, 10, 2).Perform();
}
#if defined(WEBRTC_IOS)
TEST(AudioCodingModuleTest, DISABLED_TestPacketLossStereo) {
#else
TEST(AudioCodingModuleTest, TestPacketLossStereo) {
#endif
webrtc::PacketLossTest(2, 10, 10, 1).Perform();
}
#if defined(WEBRTC_IOS)
TEST(AudioCodingModuleTest, DISABLED_TestPacketLossStereoBurst) {
#else
TEST(AudioCodingModuleTest, TestPacketLossStereoBurst) {
#endif
webrtc::PacketLossTest(2, 10, 10, 2).Perform();
}
// 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) {
webrtc::APITest().Perform();
}
#endif