Remove unnecessary using webrtc:: directives Bug: webrtc:42232595 Change-Id: Id81ef3afc4314dc00f7f2db81c8e73a4886e5fad Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/476380 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/main@{#47842}
diff --git a/api/candidate.cc b/api/candidate.cc index f32dfad..e3e0a28 100644 --- a/api/candidate.cc +++ b/api/candidate.cc
@@ -31,8 +31,6 @@ #include "rtc_base/string_encode.h" #include "rtc_base/strings/string_builder.h" -using webrtc::IceCandidateType; - namespace webrtc { namespace { constexpr char kLineTypeAttributes = 'a';
diff --git a/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/modules/audio_coding/neteq/tools/neteq_performance_test.cc index 1b4b259..0cceae3 100644 --- a/modules/audio_coding/neteq/tools/neteq_performance_test.cc +++ b/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -31,10 +31,6 @@ #include "system_wrappers/include/clock.h" #include "test/testsupport/file_utils.h" -using webrtc::NetEq; -using webrtc::test::AudioLoop; -using webrtc::test::RtpGenerator; - namespace webrtc { namespace test {
diff --git a/modules/audio_device/win/core_audio_utility_win.cc b/modules/audio_device/win/core_audio_utility_win.cc index 74a9c81..38a6161 100644 --- a/modules/audio_device/win/core_audio_utility_win.cc +++ b/modules/audio_device/win/core_audio_utility_win.cc
@@ -27,8 +27,6 @@ #include "rtc_base/win/windows_version.h" using Microsoft::WRL::ComPtr; -using webrtc::AudioDeviceName; -using webrtc::AudioParameters; namespace webrtc { namespace webrtc_win {
diff --git a/modules/audio_device/win/core_audio_utility_win_unittest.cc b/modules/audio_device/win/core_audio_utility_win_unittest.cc index b406194..58ee3b4 100644 --- a/modules/audio_device/win/core_audio_utility_win_unittest.cc +++ b/modules/audio_device/win/core_audio_utility_win_unittest.cc
@@ -16,7 +16,6 @@ #include "test/gtest.h" using Microsoft::WRL::ComPtr; -using webrtc::AudioDeviceName; namespace webrtc { namespace webrtc_win {
diff --git a/modules/desktop_capture/mac/screen_capturer_sck.mm b/modules/desktop_capture/mac/screen_capturer_sck.mm index 49c923e..6a31c6a 100644 --- a/modules/desktop_capture/mac/screen_capturer_sck.mm +++ b/modules/desktop_capture/mac/screen_capturer_sck.mm
@@ -25,8 +25,6 @@ #include "sck_picker_handle.h" #include "sdk/objc/helpers/scoped_cftyperef.h" -using webrtc::DesktopFrameIOSurface; - namespace webrtc { class ScreenCapturerSck; } // namespace webrtc
diff --git a/p2p/base/transport_description.cc b/p2p/base/transport_description.cc index 8d5ed4c..317bd7d 100644 --- a/p2p/base/transport_description.cc +++ b/p2p/base/transport_description.cc
@@ -25,10 +25,6 @@ #include "rtc_base/ssl_fingerprint.h" #include "rtc_base/strings/string_builder.h" -using webrtc::RTCError; -using webrtc::RTCErrorOr; -using webrtc::RTCErrorType; - namespace webrtc { namespace {
diff --git a/p2p/base/transport_description_unittest.cc b/p2p/base/transport_description_unittest.cc index adc63eb..21ba337 100644 --- a/p2p/base/transport_description_unittest.cc +++ b/p2p/base/transport_description_unittest.cc
@@ -16,8 +16,6 @@ #include "api/rtc_error.h" #include "test/gtest.h" -using webrtc::RTCErrorType; - namespace webrtc { TEST(IceParameters, SuccessfulParse) {
diff --git a/pc/BUILD.gn b/pc/BUILD.gn index 8e93782..867ec7f 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn
@@ -3598,7 +3598,7 @@ deps = [ ":sctp_utils", "../api:priority", - "../media:rtc_data_sctp_transport_internal", + "../api:sctp_transport_interface", "../rtc_base:byte_buffer", "../rtc_base:copy_on_write_buffer", "../test:test_support",
diff --git a/pc/dtmf_sender_unittest.cc b/pc/dtmf_sender_unittest.cc index 7c81ebc..e7a2653 100644 --- a/pc/dtmf_sender_unittest.cc +++ b/pc/dtmf_sender_unittest.cc
@@ -27,9 +27,7 @@ #include "test/time_controller/simulated_time_controller.h" #include "test/wait_until.h" -using webrtc::DtmfProviderInterface; -using webrtc::DtmfSender; -using webrtc::DtmfSenderObserverInterface; +namespace webrtc { // TODO(deadbeef): Even though this test now uses a fake clock, it has a // generous 3-second timeout for every test case. The timeout could be tuned @@ -400,3 +398,5 @@ webrtc::IsRtcOk()); VerifyExpectedState("BC", duration, inter_tone_gap); } + +} // namespace webrtc
diff --git a/pc/jsep_transport.cc b/pc/jsep_transport.cc index e63e2cc..7605a70 100644 --- a/pc/jsep_transport.cc +++ b/pc/jsep_transport.cc
@@ -51,8 +51,6 @@ #include "rtc_base/strings/string_builder.h" #include "rtc_base/trace_event.h" -using webrtc::SdpType; - namespace webrtc { JsepTransportDescription::JsepTransportDescription() {}
diff --git a/pc/local_audio_source.cc b/pc/local_audio_source.cc index 9e4f35f..87210b9 100644 --- a/pc/local_audio_source.cc +++ b/pc/local_audio_source.cc
@@ -12,11 +12,8 @@ #include "api/audio_options.h" #include "api/make_ref_counted.h" -#include "api/media_stream_interface.h" #include "api/scoped_refptr.h" -using webrtc::MediaSourceInterface; - namespace webrtc { scoped_refptr<LocalAudioSource> LocalAudioSource::Create(
diff --git a/pc/media_stream_unittest.cc b/pc/media_stream_unittest.cc index 8611ee4..8b02077 100644 --- a/pc/media_stream_unittest.cc +++ b/pc/media_stream_unittest.cc
@@ -27,7 +27,6 @@ static const char kAudioTrackId[] = "dummy_microphone_1"; using ::testing::Exactly; -using webrtc::scoped_refptr; namespace webrtc {
diff --git a/pc/sctp_utils_unittest.cc b/pc/sctp_utils_unittest.cc index df02229..2eb4848 100644 --- a/pc/sctp_utils_unittest.cc +++ b/pc/sctp_utils_unittest.cc
@@ -17,13 +17,11 @@ #include "absl/strings/string_view.h" #include "api/priority.h" -#include "media/sctp/sctp_transport_internal.h" +#include "api/sctp_transport_interface.h" #include "rtc_base/byte_buffer.h" #include "rtc_base/copy_on_write_buffer.h" #include "test/gtest.h" -using webrtc::StreamId; - class SctpUtilsTest : public ::testing::Test { public: void VerifyOpenMessageFormat(const webrtc::CopyOnWriteBuffer& packet,