Revert "Remove WEBRTC_TRACE."
This reverts commit 2209b90449473e1df3e0797b6271c7624b41907d.
Reason for revert: breaks Chromium
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}
TBR=solenberg@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
Change-Id: Ie54fc05c1d7895c088cba410ed87a7c9a0701427
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5118
Reviewed-on: https://webrtc-review.googlesource.com/5980
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20115}diff --git a/modules/audio_coding/test/APITest.cc b/modules/audio_coding/test/APITest.cc
index 363eb6b..b29e84e 100644
--- a/modules/audio_coding/test/APITest.cc
+++ b/modules/audio_coding/test/APITest.cc
@@ -25,6 +25,7 @@
#include "rtc_base/platform_thread.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/event_wrapper.h"
+#include "system_wrappers/include/trace.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
#include "typedefs.h" // NOLINT(build/include)
@@ -258,8 +259,15 @@
// B
_outFreqHzB = _outFileB.SamplingFrequency();
+ //Trace::SetEncryptedTraceFile("ACMAPITestEncrypted.txt");
+
char print[11];
+ // Create a trace file.
+ Trace::CreateTrace();
+ Trace::SetTraceFile(
+ (webrtc::test::OutputPath() + "acm_api_trace.txt").c_str());
+
printf("\nRandom Test (y/n)?");
EXPECT_TRUE(fgets(print, 10, stdin) != NULL);
print[10] = '\0';
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index c765f68..2b6b4ac 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -19,6 +19,7 @@
#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/utility.h"
+#include "system_wrappers/include/trace.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
@@ -175,6 +176,9 @@
void Receiver::Teardown() {
delete[] _playoutBuffer;
_pcmFile.Close();
+ if (testMode > 1) {
+ Trace::ReturnTrace();
+ }
}
bool Receiver::IncomingPacket() {
@@ -250,6 +254,9 @@
EncodeDecodeTest::EncodeDecodeTest() {
_testMode = 2;
+ Trace::CreateTrace();
+ Trace::SetTraceFile(
+ (webrtc::test::OutputPath() + "acm_encdec_trace.txt").c_str());
}
EncodeDecodeTest::EncodeDecodeTest(int testMode) {
@@ -257,6 +264,11 @@
//testMode == 1 for testing all codecs/parameters
//testMode > 1 for specific user-input test (as it was used before)
_testMode = testMode;
+ if (_testMode != 0) {
+ Trace::CreateTrace();
+ Trace::SetTraceFile(
+ (webrtc::test::OutputPath() + "acm_encdec_trace.txt").c_str());
+ }
}
void EncodeDecodeTest::Perform() {
@@ -314,6 +326,11 @@
rtpFile.Close();
}
}
+
+ // End tracing.
+ if (_testMode == 1) {
+ Trace::ReturnTrace();
+ }
}
std::string EncodeDecodeTest::EncodeToFile(int fileType,
diff --git a/modules/audio_coding/test/Tester.cc b/modules/audio_coding/test/Tester.cc
index 73625f1..7d58b6d 100644
--- a/modules/audio_coding/test/Tester.cc
+++ b/modules/audio_coding/test/Tester.cc
@@ -23,15 +23,22 @@
#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)
@@ -39,7 +46,11 @@
#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)
@@ -48,7 +59,11 @@
#else
TEST(AudioCodingModuleTest, TestRedFec) {
#endif
+ Trace::CreateTrace();
+ Trace::SetTraceFile((webrtc::test::OutputPath() +
+ "acm_fec_trace.txt").c_str());
webrtc::TestRedFec().Perform();
+ Trace::ReturnTrace();
}
#endif
@@ -58,7 +73,11 @@
#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
@@ -69,7 +88,11 @@
#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
@@ -79,7 +102,11 @@
#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
@@ -88,11 +115,19 @@
#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
@@ -101,15 +136,27 @@
#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
@@ -118,7 +165,11 @@
#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
@@ -127,13 +178,21 @@
#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