Fredrik Solenberg 39cefdb3c5 Revert "Reland "Remove WEBRTC_TRACE.""
This reverts commit 68007e97ec9399125e4be9964af8b0338766cd91.

Reason for revert: More downstream breakages.

Original change's description:
> Reland "Remove WEBRTC_TRACE."
> 
> This is a reland of 2209b90449473e1df3e0797b6271c7624b41907d
> Original change's description:
> > Remove WEBRTC_TRACE.
> > 
> > Bug: webrtc:5118
> > Change-Id: I0af0f8845ee016fa61d7cecc526e2a672ec8732d
> > Reviewed-on: https://webrtc-review.googlesource.com/5382
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20114}
> 
> Bug: webrtc:5118
> Change-Id: I2d93fd40fcaa251c363bdcfb1c04b834a3a7f0e9
> Reviewed-on: https://webrtc-review.googlesource.com/6000
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20132}

TBR=solenberg@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org

Change-Id: I093ee8c5c997c0dd46b3a3ca0e6271e3ea083d82
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5118
Reviewed-on: https://webrtc-review.googlesource.com/6320
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20133}
2017-10-04 08:49:49 +00:00

199 lines
6.1 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/APITest.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 "system_wrappers/include/trace.h"
#include "test/gtest.h"
#include "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