mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
5aba818e45
Since it isn't being run by the bots, it has bit rotted; when I try to run it manually, it fails with a long list of error messages: Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995 Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996 >>> Error Enabling VAD <<< Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995 Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996 >>> Error Enabling DTX <<< >>> Error Enabling VAD <<< Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995 Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996 >>> Error Enabling VAD <<< Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 995 Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 996 Error Calling API in file ../../modules/audio_coding/test/APITest.cc at line 985 ...and so on. Bug: webrtc:8396 Change-Id: Id8f1e01a751b4bb3527702b7b7a4986ce0abb378 Reviewed-on: https://webrtc-review.googlesource.com/81745 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23542}
130 lines
3.8 KiB
C++
130 lines
3.8 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 "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/TwoWayCommunication.h"
|
|
#include "modules/audio_coding/test/iSACTest.h"
|
|
#include "modules/audio_coding/test/opus_test.h"
|
|
#include "test/gtest.h"
|
|
#include "test/testsupport/fileutils.h"
|
|
|
|
// 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) {
|
|
webrtc::TestAllCodecs(ACM_TEST_MODE).Perform();
|
|
}
|
|
|
|
#if defined(WEBRTC_ANDROID)
|
|
TEST(AudioCodingModuleTest, DISABLED_TestEncodeDecode) {
|
|
#else
|
|
TEST(AudioCodingModuleTest, TestEncodeDecode) {
|
|
#endif
|
|
webrtc::EncodeDecodeTest(ACM_TEST_MODE).Perform();
|
|
}
|
|
|
|
#if defined(WEBRTC_CODEC_RED)
|
|
#if defined(WEBRTC_ANDROID)
|
|
TEST(AudioCodingModuleTest, DISABLED_TestRedFec) {
|
|
#else
|
|
TEST(AudioCodingModuleTest, TestRedFec) {
|
|
#endif
|
|
webrtc::TestRedFec().Perform();
|
|
}
|
|
#endif
|
|
|
|
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
|
|
#if defined(WEBRTC_ANDROID)
|
|
TEST(AudioCodingModuleTest, DISABLED_TestIsac) {
|
|
#else
|
|
TEST(AudioCodingModuleTest, TestIsac) {
|
|
#endif
|
|
webrtc::ISACTest(ACM_TEST_MODE).Perform();
|
|
}
|
|
#endif
|
|
|
|
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
|
|
defined(WEBRTC_CODEC_ILBC)
|
|
#if defined(WEBRTC_ANDROID)
|
|
TEST(AudioCodingModuleTest, DISABLED_TwoWayCommunication) {
|
|
#else
|
|
TEST(AudioCodingModuleTest, TwoWayCommunication) {
|
|
#endif
|
|
webrtc::TwoWayCommunication(ACM_TEST_MODE).Perform();
|
|
}
|
|
#endif
|
|
|
|
// Disabled on ios as flaky, see https://crbug.com/webrtc/7057
|
|
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
|
|
TEST(AudioCodingModuleTest, DISABLED_TestStereo) {
|
|
#else
|
|
TEST(AudioCodingModuleTest, TestStereo) {
|
|
#endif
|
|
webrtc::TestStereo(ACM_TEST_MODE).Perform();
|
|
}
|
|
|
|
// Disabled on ios as flaky, see https://crbug.com/webrtc/7057
|
|
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
|
|
TEST(AudioCodingModuleTest, DISABLED_TestWebRtcVadDtx) {
|
|
#else
|
|
TEST(AudioCodingModuleTest, TestWebRtcVadDtx) {
|
|
#endif
|
|
webrtc::TestWebRtcVadDtx().Perform();
|
|
}
|
|
|
|
TEST(AudioCodingModuleTest, TestOpusDtx) {
|
|
webrtc::TestOpusDtx().Perform();
|
|
}
|
|
|
|
// Disabled on ios as flaky, see https://crbug.com/webrtc/7057
|
|
#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();
|
|
}
|
|
|
|
// Disabled on ios as flake, see https://crbug.com/webrtc/7057
|
|
#if defined(WEBRTC_IOS)
|
|
TEST(AudioCodingModuleTest, DISABLED_TestPacketLossStereo) {
|
|
#else
|
|
TEST(AudioCodingModuleTest, TestPacketLossStereo) {
|
|
#endif
|
|
webrtc::PacketLossTest(2, 10, 10, 1).Perform();
|
|
}
|
|
|
|
// Disabled on ios as flake, see https://crbug.com/webrtc/7057
|
|
#if defined(WEBRTC_IOS)
|
|
TEST(AudioCodingModuleTest, DISABLED_TestPacketLossStereoBurst) {
|
|
#else
|
|
TEST(AudioCodingModuleTest, TestPacketLossStereoBurst) {
|
|
#endif
|
|
webrtc::PacketLossTest(2, 10, 10, 2).Perform();
|
|
}
|