[cleanup] Fix redundant webrtc name specifier
This CL was uploaded by git cl split.
R=hta@webrtc.org
Bug: webrtc:42232595
Change-Id: I0852d75dbab3768d3f508a7c37715eb0bd4cdc04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44566}
diff --git a/modules/pacing/pacing_controller_unittest.cc b/modules/pacing/pacing_controller_unittest.cc
index 2b755ff..a51e704 100644
--- a/modules/pacing/pacing_controller_unittest.cc
+++ b/modules/pacing/pacing_controller_unittest.cc
@@ -140,7 +140,7 @@
MOCK_METHOD(size_t, SendPadding, (size_t target_size));
MOCK_METHOD(void,
OnAbortedRetransmissions,
- (uint32_t, webrtc::ArrayView<const uint16_t>),
+ (uint32_t, ArrayView<const uint16_t>),
(override));
MOCK_METHOD(std::optional<uint32_t>,
GetRtxSsrcForMedia,
@@ -168,7 +168,7 @@
(override));
MOCK_METHOD(void,
OnAbortedRetransmissions,
- (uint32_t, webrtc::ArrayView<const uint16_t>),
+ (uint32_t, ArrayView<const uint16_t>),
(override));
MOCK_METHOD(std::optional<uint32_t>,
GetRtxSsrcForMedia,
@@ -304,7 +304,7 @@
type == RtpPacketMediaType::kRetransmission, false));
}
- void AdvanceTimeUntil(webrtc::Timestamp time) {
+ void AdvanceTimeUntil(Timestamp time) {
Timestamp now = clock_.CurrentTime();
clock_.AdvanceTime(std::max(TimeDelta::Zero(), time - now));
}
diff --git a/modules/rtp_rtcp/source/fec_test_helper.cc b/modules/rtp_rtcp/source/fec_test_helper.cc
index 3186a47..aa07e6b 100644
--- a/modules/rtp_rtcp/source/fec_test_helper.cc
+++ b/modules/rtp_rtcp/source/fec_test_helper.cc
@@ -84,9 +84,9 @@
// Only push one (fake) frame to the FEC.
data[1] &= 0x7f;
- webrtc::ByteWriter<uint16_t>::WriteBigEndian(&data[2], seq_num);
- webrtc::ByteWriter<uint32_t>::WriteBigEndian(&data[4], time_stamp);
- webrtc::ByteWriter<uint32_t>::WriteBigEndian(&data[8], ssrc_);
+ ByteWriter<uint16_t>::WriteBigEndian(&data[2], seq_num);
+ ByteWriter<uint32_t>::WriteBigEndian(&data[4], time_stamp);
+ ByteWriter<uint32_t>::WriteBigEndian(&data[8], ssrc_);
// Generate random values for payload.
for (size_t j = 12; j < media_packet->data.size(); ++j)
diff --git a/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc b/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc
index 312bdcb..03a3608 100644
--- a/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc
+++ b/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc
@@ -48,7 +48,7 @@
std::fill_n(data, 10, 5);
ArrayView<uint8_t> data_view(data);
EXPECT_CALL(original_frame, GetData()).WillRepeatedly(Return(data_view));
- webrtc::VideoFrameMetadata metadata;
+ VideoFrameMetadata metadata;
std::vector<uint32_t> csrcs{123, 321};
// Copy csrcs rather than moving so we can compare in an EXPECT_EQ later.
metadata.SetCsrcs(csrcs);
diff --git a/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index ea0a21a..90e75a3 100644
--- a/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -67,7 +67,7 @@
std::list<uint16_t> sequence_numbers_;
};
-class RtxLoopBackTransport : public webrtc::Transport {
+class RtxLoopBackTransport : public Transport {
public:
explicit RtxLoopBackTransport(uint32_t rtx_ssrc)
: count_(0),
diff --git a/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/modules/rtp_rtcp/source/receive_statistics_unittest.cc
index 918c74d..492eaae 100644
--- a/modules/rtp_rtcp/source/receive_statistics_unittest.cc
+++ b/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -631,7 +631,7 @@
// See jitter caluculation in https://www.rfc-editor.org/rfc/rfc3550 6.4.1.
const uint32_t expected_jitter = (kLateArrivalDeltaSamples) / 16;
EXPECT_EQ(expected_jitter, statistician->GetStats().jitter);
- EXPECT_EQ(webrtc::TimeDelta::Seconds(expected_jitter) / kCodecSampleRate,
+ EXPECT_EQ(TimeDelta::Seconds(expected_jitter) / kCodecSampleRate,
statistician->GetStats().interarrival_jitter);
}
diff --git a/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc
index 4448f34..384f1c0 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc
@@ -20,9 +20,9 @@
namespace webrtc {
namespace {
+using rtcp::App;
using ::testing::ElementsAreArray;
using ::testing::make_tuple;
-using ::webrtc::rtcp::App;
constexpr uint32_t kName = ((uint32_t)'n' << 24) | ((uint32_t)'a' << 16) |
((uint32_t)'m' << 8) | (uint32_t)'e';
diff --git a/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc
index b7b9c54..312f051 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc
@@ -21,9 +21,9 @@
namespace webrtc {
+using rtcp::LossNotification;
using ::testing::ElementsAreArray;
using ::testing::make_tuple;
-using ::webrtc::rtcp::LossNotification;
TEST(RtcpPacketLossNotificationTest, SetWithIllegalValuesFails) {
constexpr uint16_t kLastDecoded = 0x3c7b;
diff --git a/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc b/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc
index de7b942..a41700b 100644
--- a/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc
@@ -23,6 +23,7 @@
namespace webrtc {
namespace {
+using rtcp::Nack;
using ::testing::_;
using ::testing::ElementsAre;
using ::testing::ElementsAreArray;
@@ -30,7 +31,6 @@
using ::testing::make_tuple;
using ::testing::MockFunction;
using ::testing::UnorderedElementsAreArray;
-using ::webrtc::rtcp::Nack;
constexpr uint32_t kSenderSsrc = 0x12345678;
constexpr uint32_t kRemoteSsrc = 0x23456789;
diff --git a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index 4da2dce..31d218e 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -68,6 +68,7 @@
namespace {
using rtcp::ReceiveTimeInfo;
+using test::ExplicitKeyValueConfig;
using ::testing::_;
using ::testing::AllOf;
using ::testing::ElementsAre;
@@ -83,7 +84,6 @@
using ::testing::StrEq;
using ::testing::StrictMock;
using ::testing::UnorderedElementsAre;
-using ::webrtc::test::ExplicitKeyValueConfig;
class MockRtcpPacketTypeCounterObserver : public RtcpPacketTypeCounterObserver {
public:
diff --git a/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc b/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
index 3a10304..be485b8 100644
--- a/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
+++ b/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
@@ -59,6 +59,11 @@
namespace webrtc {
namespace {
+using rtcp::Bye;
+using rtcp::CompoundPacket;
+using rtcp::ReportBlock;
+using rtcp::SenderReport;
+using test::RtcpPacketParser;
using ::testing::_;
using ::testing::ElementsAre;
using ::testing::ElementsAreArray;
@@ -71,11 +76,6 @@
using ::testing::StrictMock;
using ::testing::UnorderedElementsAre;
using ::testing::WithArg;
-using ::webrtc::rtcp::Bye;
-using ::webrtc::rtcp::CompoundPacket;
-using ::webrtc::rtcp::ReportBlock;
-using ::webrtc::rtcp::SenderReport;
-using ::webrtc::test::RtcpPacketParser;
class MockReceiveStatisticsProvider : public ReceiveStatisticsProvider {
public:
@@ -432,7 +432,7 @@
RtcpTransceiverConfig config = DefaultTestConfig();
config.rtcp_transport = transport.AsStdFunction();
- config.rtcp_mode = webrtc::RtcpMode::kReducedSize;
+ config.rtcp_mode = RtcpMode::kReducedSize;
config.receive_statistics = &receive_statistics;
RtcpTransceiverImpl rtcp_transceiver(config);
@@ -445,7 +445,7 @@
RtcpTransceiverConfig config = DefaultTestConfig();
config.rtcp_transport = RtcpParserTransport(rtcp_parser);
- config.rtcp_mode = webrtc::RtcpMode::kReducedSize;
+ config.rtcp_mode = RtcpMode::kReducedSize;
config.receive_statistics = &receive_statistics;
// Set it to produce something (RRTR) in the "periodic" rtcp packets.
config.non_sender_rtt_measurement = true;
@@ -545,7 +545,7 @@
const uint32_t kSenderSsrc = 12345;
RtcpTransceiverConfig config = DefaultTestConfig();
config.send_remb_on_change = true;
- config.rtcp_mode = webrtc::RtcpMode::kReducedSize;
+ config.rtcp_mode = RtcpMode::kReducedSize;
config.feedback_ssrc = kSenderSsrc;
RtcpPacketParser rtcp_parser;
config.rtcp_transport = RtcpParserTransport(rtcp_parser);
@@ -717,12 +717,12 @@
rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc1, &observer1);
rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc2, &observer2);
- webrtc::rtcp::TargetBitrate target_bitrate;
+ rtcp::TargetBitrate target_bitrate;
target_bitrate.AddTargetBitrate(0, 0, /*target_bitrate_kbps=*/10);
target_bitrate.AddTargetBitrate(0, 1, /*target_bitrate_kbps=*/20);
target_bitrate.AddTargetBitrate(1, 0, /*target_bitrate_kbps=*/40);
target_bitrate.AddTargetBitrate(1, 1, /*target_bitrate_kbps=*/80);
- webrtc::rtcp::ExtendedReports xr;
+ rtcp::ExtendedReports xr;
xr.SetSenderSsrc(kRemoteSsrc1);
xr.SetTargetBitrate(target_bitrate);
auto raw_packet = xr.Build();
@@ -744,12 +744,12 @@
RtcpTransceiverImpl rtcp_transceiver(config);
rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, &observer);
- webrtc::rtcp::TargetBitrate target_bitrate;
+ rtcp::TargetBitrate target_bitrate;
target_bitrate.AddTargetBitrate(0, 0, /*target_bitrate_kbps=*/10);
- target_bitrate.AddTargetBitrate(0, webrtc::kMaxTemporalStreams, 20);
- target_bitrate.AddTargetBitrate(webrtc::kMaxSpatialLayers, 0, 40);
+ target_bitrate.AddTargetBitrate(0, kMaxTemporalStreams, 20);
+ target_bitrate.AddTargetBitrate(kMaxSpatialLayers, 0, 40);
- webrtc::rtcp::ExtendedReports xr;
+ rtcp::ExtendedReports xr;
xr.SetTargetBitrate(target_bitrate);
xr.SetSenderSsrc(kRemoteSsrc);
auto raw_packet = xr.Build();
@@ -790,7 +790,7 @@
CompoundPacket compound;
// Use Application-Defined rtcp packet as unknown.
- auto app = std::make_unique<webrtc::rtcp::App>();
+ auto app = std::make_unique<rtcp::App>();
compound.Append(std::move(app));
auto bye = std::make_unique<Bye>();
bye->SetSenderSsrc(kRemoteSsrc);
@@ -1180,7 +1180,7 @@
RtcpPacketParser rtcp_parser;
config.rtcp_transport = RtcpParserTransport(rtcp_parser);
- config.rtcp_mode = webrtc::RtcpMode::kCompound;
+ config.rtcp_mode = RtcpMode::kCompound;
RtcpTransceiverImpl rtcp_transceiver(config);
rtcp_transceiver.SendFullIntraRequest(kRemoteSsrcs, true);
@@ -1198,7 +1198,7 @@
RtcpPacketParser rtcp_parser;
config.rtcp_transport = RtcpParserTransport(rtcp_parser);
- config.rtcp_mode = webrtc::RtcpMode::kReducedSize;
+ config.rtcp_mode = RtcpMode::kReducedSize;
RtcpTransceiverImpl rtcp_transceiver(config);
rtcp_transceiver.SendFullIntraRequest(kRemoteSsrcs, true);
@@ -1393,10 +1393,10 @@
RtcpTransceiverImpl rtcp_transceiver(config);
Timestamp time = Timestamp::Micros(12345678);
- webrtc::rtcp::ReceiveTimeInfo rti;
+ rtcp::ReceiveTimeInfo rti;
rti.ssrc = kUnknownSsrc;
rti.last_rr = CompactNtp(config.clock->ConvertTimestampToNtpTime(time));
- webrtc::rtcp::ExtendedReports xr;
+ rtcp::ExtendedReports xr;
xr.AddDlrrItem(rti);
auto raw_packet = xr.Build();
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
index 83e5a1f..2cb50849 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
@@ -45,7 +45,7 @@
using ::testing::ElementsAreArray;
-class LoopbackTransportTest : public webrtc::Transport {
+class LoopbackTransportTest : public Transport {
public:
LoopbackTransportTest() {
receivers_extensions_.Register<AudioLevelExtension>(kAudioLevelExtensionId);
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index acb4c73..96662d1 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -669,7 +669,7 @@
Buffer encrypted_video_payload;
if (frame_encryptor_ != nullptr) {
const size_t max_ciphertext_size =
- frame_encryptor_->GetMaxCiphertextByteSize(webrtc::MediaType::VIDEO,
+ frame_encryptor_->GetMaxCiphertextByteSize(MediaType::VIDEO,
payload.size());
encrypted_video_payload.SetSize(max_ciphertext_size);
@@ -682,8 +682,8 @@
}
if (frame_encryptor_->Encrypt(
- webrtc::MediaType::VIDEO, first_packet->Ssrc(), additional_data,
- payload, encrypted_video_payload, &bytes_written) != 0) {
+ MediaType::VIDEO, first_packet->Ssrc(), additional_data, payload,
+ encrypted_video_payload, &bytes_written) != 0) {
return false;
}
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
index a67ebfd..79432a9 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -115,7 +115,7 @@
constexpr Timestamp kStartTime = Timestamp::Millis(123456789);
constexpr TimeDelta kDefaultExpectedRetransmissionTime = TimeDelta::Millis(125);
-class LoopbackTransportTest : public webrtc::Transport {
+class LoopbackTransportTest : public Transport {
public:
LoopbackTransportTest() {
receivers_extensions_.Register<TransmissionOffset>(
@@ -1577,8 +1577,7 @@
std::unique_ptr<EncodedImage> CreateDefaultEncodedImage() {
const uint8_t data[] = {1, 2, 3, 4};
auto encoded_image = std::make_unique<EncodedImage>();
- encoded_image->SetEncodedData(
- webrtc::EncodedImageBuffer::Create(data, sizeof(data)));
+ encoded_image->SetEncodedData(EncodedImageBuffer::Create(data, sizeof(data)));
return encoded_image;
}
diff --git a/modules/rtp_rtcp/test/testFec/test_fec.cc b/modules/rtp_rtcp/test/testFec/test_fec.cc
index bd6998d..2898a47 100644
--- a/modules/rtp_rtcp/test/testFec/test_fec.cc
+++ b/modules/rtp_rtcp/test/testFec/test_fec.cc
@@ -133,7 +133,7 @@
// reproduce past results.
const unsigned int random_seed = static_cast<unsigned int>(time(nullptr));
Random random(random_seed);
- std::string filename = webrtc::test::OutputPath() + "randomSeedLog.txt";
+ std::string filename = test::OutputPath() + "randomSeedLog.txt";
FILE* random_seed_file = fopen(filename.c_str(), "a");
fprintf(random_seed_file, "%u\n", random_seed);
fclose(random_seed_file);
