Don't make a top-level namespace called "voetest"
We shouldn't pollute the global namespace.
BUG=webrtc:7484
Review-Url: https://codereview.webrtc.org/2813373002
Cr-Original-Commit-Position: refs/heads/master@{#17797}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 492c09fe59064c5d49f9110ed9c638a5f8a016b4
diff --git a/voice_engine/test/auto_test/automated_mode.cc b/voice_engine/test/auto_test/automated_mode.cc
index 67a73ee..098c574 100644
--- a/voice_engine/test/auto_test/automated_mode.cc
+++ b/voice_engine/test/auto_test/automated_mode.cc
@@ -11,6 +11,9 @@
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
+namespace webrtc {
+namespace voetest {
+
void InitializeGoogleTest(int* argc, char** argv) {
// Initialize WebRTC testing framework so paths to resources can be resolved.
webrtc::test::SetExecutablePath(argv[0]);
@@ -20,3 +23,6 @@
int RunInAutomatedMode() {
return RUN_ALL_TESTS();
}
+
+} // namespace voetest
+} // namespace webrtc
diff --git a/voice_engine/test/auto_test/automated_mode.h b/voice_engine/test/auto_test/automated_mode.h
index 599f021..0d673a4 100644
--- a/voice_engine/test/auto_test/automated_mode.h
+++ b/voice_engine/test/auto_test/automated_mode.h
@@ -11,7 +11,13 @@
#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_
#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_
+namespace webrtc {
+namespace voetest {
+
void InitializeGoogleTest(int* argc, char** argv);
int RunInAutomatedMode();
+} // namespace voetest
+} // namespace webrtc
+
#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_
diff --git a/voice_engine/test/auto_test/fakes/conference_transport.cc b/voice_engine/test/auto_test/fakes/conference_transport.cc
index eab66e7..6fbf5f1 100644
--- a/voice_engine/test/auto_test/fakes/conference_transport.cc
+++ b/voice_engine/test/auto_test/fakes/conference_transport.cc
@@ -18,26 +18,28 @@
#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/voice_engine_impl.h"
-namespace {
- static const unsigned int kReflectorSsrc = 0x0000;
- static const unsigned int kLocalSsrc = 0x0001;
- static const unsigned int kFirstRemoteSsrc = 0x0002;
- static const webrtc::CodecInst kCodecInst =
- {120, "opus", 48000, 960, 2, 64000};
- static const int kAudioLevelHeaderId = 1;
-
- static unsigned int ParseRtcpSsrc(const void* data, size_t len) {
- const size_t ssrc_pos = 4;
- unsigned int ssrc = 0;
- if (len >= (ssrc_pos + sizeof(ssrc))) {
- ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
- }
- return ssrc;
- }
-} // namespace
-
+namespace webrtc {
namespace voetest {
+namespace {
+
+static const unsigned int kReflectorSsrc = 0x0000;
+static const unsigned int kLocalSsrc = 0x0001;
+static const unsigned int kFirstRemoteSsrc = 0x0002;
+static const webrtc::CodecInst kCodecInst = {120, "opus", 48000, 960, 2, 64000};
+static const int kAudioLevelHeaderId = 1;
+
+static unsigned int ParseRtcpSsrc(const void* data, size_t len) {
+ const size_t ssrc_pos = 4;
+ unsigned int ssrc = 0;
+ if (len >= (ssrc_pos + sizeof(ssrc))) {
+ ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
+ }
+ return ssrc;
+}
+
+} // namespace
+
ConferenceTransport::ConferenceTransport()
: packet_event_(webrtc::EventWrapper::Create()),
thread_(Run, this, "ConferenceTransport"),
@@ -297,4 +299,6 @@
EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats));
return true;
}
+
} // namespace voetest
+} // namespace webrtc
diff --git a/voice_engine/test/auto_test/fakes/conference_transport.h b/voice_engine/test/auto_test/fakes/conference_transport.h
index d8a5a49..37919e1 100644
--- a/voice_engine/test/auto_test/fakes/conference_transport.h
+++ b/voice_engine/test/auto_test/fakes/conference_transport.h
@@ -30,10 +30,11 @@
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h"
-static const size_t kMaxPacketSizeByte = 1500;
-
+namespace webrtc {
namespace voetest {
+static const size_t kMaxPacketSizeByte = 1500;
+
// This class is to simulate a conference call. There are two Voice Engines, one
// for local channels and the other for remote channels. There is a simulated
// reflector, which exchanges RTCP with local channels. For simplicity, it
@@ -158,6 +159,8 @@
const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
};
+
} // namespace voetest
+} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
diff --git a/voice_engine/test/auto_test/fakes/loudest_filter.cc b/voice_engine/test/auto_test/fakes/loudest_filter.cc
index 09ca127..4ee7386 100644
--- a/voice_engine/test/auto_test/fakes/loudest_filter.cc
+++ b/voice_engine/test/auto_test/fakes/loudest_filter.cc
@@ -12,6 +12,7 @@
#include "webrtc/base/checks.h"
+namespace webrtc {
namespace voetest {
void LoudestFilter::RemoveTimeoutStreams(int64_t time_ms) {
@@ -78,4 +79,4 @@
}
} // namespace voetest
-
+} // namespace webrtc
diff --git a/voice_engine/test/auto_test/fakes/loudest_filter.h b/voice_engine/test/auto_test/fakes/loudest_filter.h
index f862c81..525e8ee 100644
--- a/voice_engine/test/auto_test/fakes/loudest_filter.h
+++ b/voice_engine/test/auto_test/fakes/loudest_filter.h
@@ -15,6 +15,7 @@
#include "webrtc/base/timeutils.h"
#include "webrtc/common_types.h"
+namespace webrtc {
namespace voetest {
class LoudestFilter {
@@ -48,7 +49,7 @@
const int kInvalidAudioLevel = 128;
};
-
} // namespace voetest
+} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_
diff --git a/voice_engine/test/auto_test/voe_conference_test.cc b/voice_engine/test/auto_test/voe_conference_test.cc
index 9c4d204..ffffb48 100644
--- a/voice_engine/test/auto_test/voe_conference_test.cc
+++ b/voice_engine/test/auto_test/voe_conference_test.cc
@@ -17,7 +17,9 @@
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"
+namespace webrtc {
namespace {
+
const int kRttMs = 25;
bool IsNear(int ref, int comp, int error) {
@@ -174,3 +176,4 @@
}
} // namespace voetest
+} // namespace webrtc
diff --git a/voice_engine/test/auto_test/voe_standard_test.cc b/voice_engine/test/auto_test/voe_standard_test.cc
index 82aa441..4cb6d52 100644
--- a/voice_engine/test/auto_test/voe_standard_test.cc
+++ b/voice_engine/test/auto_test/voe_standard_test.cc
@@ -27,8 +27,7 @@
"If true, we'll run the automated tests we have in noninteractive "
"mode.");
-using namespace webrtc;
-
+namespace webrtc {
namespace voetest {
int dummy = 0; // Dummy used in different functions to avoid warnings
@@ -67,11 +66,8 @@
TEST_LOG(" AudioProcessing\n");
ANL();
}
-} // namespace voetest
int RunInManualMode() {
- using namespace voetest;
-
SubAPIManager api_manager;
api_manager.DisplayStatus();
@@ -97,24 +93,23 @@
}
}
-// ----------------------------------------------------------------------------
-// main
-// ----------------------------------------------------------------------------
+} // namespace voetest
+} // namespace webrtc
#if !defined(WEBRTC_IOS)
int main(int argc, char** argv) {
// This function and RunInAutomatedMode is defined in automated_mode.cc
// to avoid macro clashes with googletest (for instance ASSERT_TRUE).
- InitializeGoogleTest(&argc, argv);
+ webrtc::voetest::InitializeGoogleTest(&argc, argv);
// AllowCommandLineParsing allows us to ignore flags passed on to us by
// Chromium build bots without having to explicitly disable them.
google::AllowCommandLineReparsing();
google::ParseCommandLineFlags(&argc, &argv, true);
if (FLAGS_automated) {
- return RunInAutomatedMode();
+ return webrtc::voetest::RunInAutomatedMode();
}
- return RunInManualMode();
+ return webrtc::voetest::RunInManualMode();
}
#endif //#if !defined(WEBRTC_IOS)
diff --git a/voice_engine/test/auto_test/voe_standard_test.h b/voice_engine/test/auto_test/voe_standard_test.h
index 338eb36..1fd2706 100644
--- a/voice_engine/test/auto_test/voe_standard_test.h
+++ b/voice_engine/test/auto_test/voe_standard_test.h
@@ -18,12 +18,9 @@
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/voice_engine/test/auto_test/voe_test_common.h"
-#if defined(WEBRTC_ANDROID)
-extern char mobileLogMsg[640];
-#endif
-
DECLARE_bool(include_timing_dependent_tests);
+namespace webrtc {
namespace voetest {
class SubAPIManager {
@@ -53,5 +50,6 @@
};
} // namespace voetest
+} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_VOE_STANDARD_TEST_H
