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Moving src/webrtc into src/.

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In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
This commit is contained in:
Mirko Bonadei
2017-09-15 06:15:48 +02:00
committed by Commit Bot
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
Download Patch File Download Diff File Expand all files Collapse all files

198
modules/audio_coding/test/Tester.cc Normal file
View File

@ -0,0 +1,198 @@
/*
* 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 "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/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/modules/audio_coding/test/iSACTest.h"
#include "webrtc/modules/audio_coding/test/opus_test.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/gtest.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
// 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
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_stereo_trace.txt").c_str());
webrtc::TestStereo(ACM_TEST_MODE).Perform();
Trace::ReturnTrace();
}
// 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
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();
}
// Disabled on ios as flaky, see https://crbug.com/webrtc/7057
#if defined(WEBRTC_IOS)
TEST(AudioCodingModuleTest, DISABLED_TestOpus) {
#else
TEST(AudioCodingModuleTest, TestOpus) {
#endif
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();
}
// Disabled on ios as flake, see https://crbug.com/webrtc/7057
#if defined(WEBRTC_IOS)
TEST(AudioCodingModuleTest, DISABLED_TestPacketLossStereo) {
#else
TEST(AudioCodingModuleTest, TestPacketLossStereo) {
#endif
Trace::CreateTrace();
Trace::SetTraceFile((webrtc::test::OutputPath() +
"acm_packetloss_trace.txt").c_str());
webrtc::PacketLossTest(2, 10, 10, 1).Perform();
Trace::ReturnTrace();
}
// Disabled on ios as flake, see https://crbug.com/webrtc/7057
#if defined(WEBRTC_IOS)
TEST(AudioCodingModuleTest, DISABLED_TestPacketLossStereoBurst) {
#else
TEST(AudioCodingModuleTest, TestPacketLossStereoBurst) {
#endif
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