[cleanup] Fix redundant webrtc name specifier
This CL was uploaded by git cl split.
R=hta@webrtc.org
No-IWYU: LSC
Bug: webrtc:42232595
Change-Id: I79a0259f1ead29262eb143ad4073689856d4925e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390604
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44573}
diff --git a/modules/audio_processing/aec3/block_framer_unittest.cc b/modules/audio_processing/aec3/block_framer_unittest.cc
index c9b6058..d45b96e 100644
--- a/modules/audio_processing/aec3/block_framer_unittest.cc
+++ b/modules/audio_processing/aec3/block_framer_unittest.cc
@@ -124,9 +124,9 @@
num_sub_frame_bands,
std::vector<std::vector<float>>(
num_sub_frame_channels, std::vector<float>(sub_frame_length, 0.f)));
- std::vector<std::vector<webrtc::ArrayView<float>>> output_sub_frame_view(
+ std::vector<std::vector<ArrayView<float>>> output_sub_frame_view(
output_sub_frame.size(),
- std::vector<webrtc::ArrayView<float>>(num_sub_frame_channels));
+ std::vector<ArrayView<float>>(num_sub_frame_channels));
SetupSubFrameView(&output_sub_frame, &output_sub_frame_view);
BlockFramer framer(correct_num_bands, correct_num_channels);
EXPECT_DEATH(
@@ -147,9 +147,9 @@
correct_num_bands,
std::vector<std::vector<float>>(
correct_num_channels, std::vector<float>(kSubFrameLength, 0.f)));
- std::vector<std::vector<webrtc::ArrayView<float>>> output_sub_frame_view(
+ std::vector<std::vector<ArrayView<float>>> output_sub_frame_view(
output_sub_frame.size(),
- std::vector<webrtc::ArrayView<float>>(correct_num_channels));
+ std::vector<ArrayView<float>>(correct_num_channels));
SetupSubFrameView(&output_sub_frame, &output_sub_frame_view);
BlockFramer framer(correct_num_bands, correct_num_channels);
framer.InsertBlockAndExtractSubFrame(correct_block, &output_sub_frame_view);
@@ -174,9 +174,8 @@
correct_num_bands,
std::vector<std::vector<float>>(
num_channels, std::vector<float>(kSubFrameLength, 0.f)));
- std::vector<std::vector<webrtc::ArrayView<float>>> output_sub_frame_view(
- output_sub_frame.size(),
- std::vector<webrtc::ArrayView<float>>(num_channels));
+ std::vector<std::vector<ArrayView<float>>> output_sub_frame_view(
+ output_sub_frame.size(), std::vector<ArrayView<float>>(num_channels));
SetupSubFrameView(&output_sub_frame, &output_sub_frame_view);
BlockFramer framer(correct_num_bands, num_channels);
for (size_t k = 0; k < num_preceeding_api_calls; ++k) {
@@ -293,7 +292,7 @@
for (size_t num_channels : {1, 2, 8}) {
for (auto rate : {16000, 32000, 48000}) {
for (size_t num_calls = 0; num_calls < 4; ++num_calls) {
- webrtc::StringBuilder ss;
+ StringBuilder ss;
ss << "Sample rate: " << rate;
ss << ", Num channels: " << num_channels;
ss << ", Num preceeding InsertBlockAndExtractSubFrame calls: "
diff --git a/modules/audio_processing/aec3/block_processor_unittest.cc b/modules/audio_processing/aec3/block_processor_unittest.cc
index a56eaac..eece30c 100644
--- a/modules/audio_processing/aec3/block_processor_unittest.cc
+++ b/modules/audio_processing/aec3/block_processor_unittest.cc
@@ -132,9 +132,9 @@
Random random_generator(42U);
for (auto rate : {16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
- std::unique_ptr<testing::StrictMock<webrtc::test::MockRenderDelayBuffer>>
+ std::unique_ptr<testing::StrictMock<test::MockRenderDelayBuffer>>
render_delay_buffer_mock(
- new StrictMock<webrtc::test::MockRenderDelayBuffer>(rate, 1));
+ new StrictMock<test::MockRenderDelayBuffer>(rate, 1));
EXPECT_CALL(*render_delay_buffer_mock, Insert(_))
.Times(kNumBlocks)
.WillRepeatedly(Return(RenderDelayBuffer::BufferingEvent::kNone));
@@ -171,15 +171,14 @@
Random random_generator(42U);
for (auto rate : {16000, 32000, 48000}) {
SCOPED_TRACE(ProduceDebugText(rate));
- std::unique_ptr<testing::StrictMock<webrtc::test::MockRenderDelayBuffer>>
+ std::unique_ptr<testing::StrictMock<test::MockRenderDelayBuffer>>
render_delay_buffer_mock(
- new StrictMock<webrtc::test::MockRenderDelayBuffer>(rate, 1));
- std::unique_ptr<
- ::testing::StrictMock<webrtc::test::MockRenderDelayController>>
+ new StrictMock<test::MockRenderDelayBuffer>(rate, 1));
+ std::unique_ptr<::testing::StrictMock<test::MockRenderDelayController>>
render_delay_controller_mock(
- new StrictMock<webrtc::test::MockRenderDelayController>());
- std::unique_ptr<testing::StrictMock<webrtc::test::MockEchoRemover>>
- echo_remover_mock(new StrictMock<webrtc::test::MockEchoRemover>());
+ new StrictMock<test::MockRenderDelayController>());
+ std::unique_ptr<testing::StrictMock<test::MockEchoRemover>>
+ echo_remover_mock(new StrictMock<test::MockEchoRemover>());
EXPECT_CALL(*render_delay_buffer_mock, Insert(_))
.Times(kNumBlocks - 1)
@@ -291,10 +290,9 @@
std::unique_ptr<RenderDelayBuffer> delay_buffer(
RenderDelayBuffer::Create(config, kSampleRateHz, kNumRenderChannels));
- std::unique_ptr<testing::NiceMock<webrtc::test::MockEchoRemover>>
- echo_remover_mock(new NiceMock<webrtc::test::MockEchoRemover>());
- webrtc::test::MockEchoRemover* echo_remover_mock_pointer =
- echo_remover_mock.get();
+ std::unique_ptr<testing::NiceMock<test::MockEchoRemover>> echo_remover_mock(
+ new NiceMock<test::MockEchoRemover>());
+ test::MockEchoRemover* echo_remover_mock_pointer = echo_remover_mock.get();
std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create(
config, kSampleRateHz, kNumRenderChannels, kNumCaptureChannels,
diff --git a/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
index 486c64c..f0fa73d 100644
--- a/modules/audio_processing/aec3/echo_canceller3_unittest.cc
+++ b/modules/audio_processing/aec3/echo_canceller3_unittest.cc
@@ -316,9 +316,8 @@
constexpr size_t kNumFullBlocksPerFrame = 160 / kBlockSize;
constexpr size_t kExpectedNumBlocksToProcess =
(kNumFramesToProcess * 160) / kBlockSize;
- std::unique_ptr<testing::StrictMock<webrtc::test::MockBlockProcessor>>
- block_processor_mock(
- new StrictMock<webrtc::test::MockBlockProcessor>());
+ std::unique_ptr<testing::StrictMock<test::MockBlockProcessor>>
+ block_processor_mock(new StrictMock<test::MockBlockProcessor>());
EXPECT_CALL(*block_processor_mock, BufferRender(_))
.Times(kExpectedNumBlocksToProcess);
EXPECT_CALL(*block_processor_mock, UpdateEchoLeakageStatus(_)).Times(0);
@@ -393,9 +392,8 @@
EchoLeakageTestVariant leakage_report_variant) {
constexpr size_t kExpectedNumBlocksToProcess =
(kNumFramesToProcess * 160) / kBlockSize;
- std::unique_ptr<testing::StrictMock<webrtc::test::MockBlockProcessor>>
- block_processor_mock(
- new StrictMock<webrtc::test::MockBlockProcessor>());
+ std::unique_ptr<testing::StrictMock<test::MockBlockProcessor>>
+ block_processor_mock(new StrictMock<test::MockBlockProcessor>());
EXPECT_CALL(*block_processor_mock, BufferRender(_))
.Times(kExpectedNumBlocksToProcess);
EXPECT_CALL(*block_processor_mock, ProcessCapture(_, _, _, _))
@@ -482,9 +480,8 @@
const size_t kNumFullBlocksPerFrame = 160 / kBlockSize;
const size_t kExpectedNumBlocksToProcess =
(kNumFramesToProcess * 160) / kBlockSize;
- std::unique_ptr<testing::StrictMock<webrtc::test::MockBlockProcessor>>
- block_processor_mock(
- new StrictMock<webrtc::test::MockBlockProcessor>());
+ std::unique_ptr<testing::StrictMock<test::MockBlockProcessor>>
+ block_processor_mock(new StrictMock<test::MockBlockProcessor>());
EXPECT_CALL(*block_processor_mock, BufferRender(_))
.Times(kExpectedNumBlocksToProcess);
EXPECT_CALL(*block_processor_mock, UpdateEchoLeakageStatus(_)).Times(0);
diff --git a/modules/audio_processing/aec3/frame_blocker_unittest.cc b/modules/audio_processing/aec3/frame_blocker_unittest.cc
index 1350184..ee62969 100644
--- a/modules/audio_processing/aec3/frame_blocker_unittest.cc
+++ b/modules/audio_processing/aec3/frame_blocker_unittest.cc
@@ -199,9 +199,9 @@
num_sub_frame_bands,
std::vector<std::vector<float>>(
num_sub_frame_channels, std::vector<float>(sub_frame_length, 0.f)));
- std::vector<std::vector<webrtc::ArrayView<float>>> input_sub_frame_view(
+ std::vector<std::vector<ArrayView<float>>> input_sub_frame_view(
input_sub_frame.size(),
- std::vector<webrtc::ArrayView<float>>(num_sub_frame_channels));
+ std::vector<ArrayView<float>>(num_sub_frame_channels));
FillSubFrameView(0, 0, &input_sub_frame, &input_sub_frame_view);
FrameBlocker blocker(correct_num_bands, correct_num_channels);
EXPECT_DEATH(
@@ -222,9 +222,9 @@
correct_num_bands,
std::vector<std::vector<float>>(
correct_num_channels, std::vector<float>(kSubFrameLength, 0.f)));
- std::vector<std::vector<webrtc::ArrayView<float>>> input_sub_frame_view(
+ std::vector<std::vector<ArrayView<float>>> input_sub_frame_view(
input_sub_frame.size(),
- std::vector<webrtc::ArrayView<float>>(correct_num_channels));
+ std::vector<ArrayView<float>>(correct_num_channels));
FillSubFrameView(0, 0, &input_sub_frame, &input_sub_frame_view);
FrameBlocker blocker(correct_num_bands, correct_num_channels);
blocker.InsertSubFrameAndExtractBlock(input_sub_frame_view, &correct_block);
@@ -247,9 +247,8 @@
std::vector<std::vector<std::vector<float>>> input_sub_frame(
num_bands, std::vector<std::vector<float>>(
num_channels, std::vector<float>(kSubFrameLength, 0.f)));
- std::vector<std::vector<webrtc::ArrayView<float>>> input_sub_frame_view(
- input_sub_frame.size(),
- std::vector<webrtc::ArrayView<float>>(num_channels));
+ std::vector<std::vector<ArrayView<float>>> input_sub_frame_view(
+ input_sub_frame.size(), std::vector<ArrayView<float>>(num_channels));
FillSubFrameView(0, 0, &input_sub_frame, &input_sub_frame_view);
FrameBlocker blocker(num_bands, num_channels);
for (size_t k = 0; k < num_preceeding_api_calls; ++k) {
@@ -367,7 +366,7 @@
for (auto rate : {16000, 32000, 48000}) {
for (size_t num_channels : {1, 2, 4, 8}) {
for (size_t num_calls = 0; num_calls < 4; ++num_calls) {
- webrtc::StringBuilder ss;
+ StringBuilder ss;
ss << "Sample rate: " << rate;
ss << "Num channels: " << num_channels;
ss << ", Num preceeding InsertSubFrameAndExtractBlock calls: "
@@ -395,8 +394,7 @@
std::vector<std::vector<std::vector<float>>> sub_frame(
1, std::vector<std::vector<float>>(
1, std::vector<float>(kSubFrameLength, 0.f)));
- std::vector<std::vector<webrtc::ArrayView<float>>> sub_frame_view(
- sub_frame.size());
+ std::vector<std::vector<ArrayView<float>>> sub_frame_view(sub_frame.size());
FillSubFrameView(0, 0, &sub_frame, &sub_frame_view);
EXPECT_DEATH(
FrameBlocker(1, 1).InsertSubFrameAndExtractBlock(sub_frame_view, nullptr),
diff --git a/modules/audio_processing/aec_dump/aec_dump_impl.cc b/modules/audio_processing/aec_dump/aec_dump_impl.cc
index 0c19650..31826ff 100644
--- a/modules/audio_processing/aec_dump/aec_dump_impl.cc
+++ b/modules/audio_processing/aec_dump/aec_dump_impl.cc
@@ -23,8 +23,8 @@
namespace webrtc {
namespace {
-void CopyFromConfigToEvent(const webrtc::InternalAPMConfig& config,
- webrtc::audioproc::Config* pb_cfg) {
+void CopyFromConfigToEvent(const InternalAPMConfig& config,
+ audioproc::Config* pb_cfg) {
pb_cfg->set_aec_enabled(config.aec_enabled);
pb_cfg->set_aec_delay_agnostic_enabled(config.aec_delay_agnostic_enabled);
pb_cfg->set_aec_drift_compensation_enabled(
diff --git a/modules/audio_processing/aec_dump/aec_dump_unittest.cc b/modules/audio_processing/aec_dump/aec_dump_unittest.cc
index 2a8110c..f928ae8 100644
--- a/modules/audio_processing/aec_dump/aec_dump_unittest.cc
+++ b/modules/audio_processing/aec_dump/aec_dump_unittest.cc
@@ -21,14 +21,14 @@
TEST(AecDumper, APICallsDoNotCrash) {
// Note order of initialization: Task queue has to be initialized
// before AecDump.
- webrtc::TaskQueueForTest file_writer_queue("file_writer_queue");
+ TaskQueueForTest file_writer_queue("file_writer_queue");
const std::string filename =
- webrtc::test::TempFilename(webrtc::test::OutputPath(), "aec_dump");
+ test::TempFilename(test::OutputPath(), "aec_dump");
{
- std::unique_ptr<webrtc::AecDump> aec_dump =
- webrtc::AecDumpFactory::Create(filename, -1, file_writer_queue.Get());
+ std::unique_ptr<AecDump> aec_dump =
+ AecDumpFactory::Create(filename, -1, file_writer_queue.Get());
constexpr int kNumChannels = 1;
constexpr int kNumSamplesPerChannel = 160;
@@ -44,10 +44,10 @@
aec_dump->WriteCaptureStreamMessage();
- webrtc::InternalAPMConfig apm_config;
+ InternalAPMConfig apm_config;
aec_dump->WriteConfig(apm_config);
- webrtc::ProcessingConfig api_format;
+ ProcessingConfig api_format;
constexpr int64_t kTimeNowMs = 123456789ll;
aec_dump->WriteInitMessage(api_format, kTimeNowMs);
}
@@ -56,14 +56,14 @@
}
TEST(AecDumper, WriteToFile) {
- webrtc::TaskQueueForTest file_writer_queue("file_writer_queue");
+ TaskQueueForTest file_writer_queue("file_writer_queue");
const std::string filename =
- webrtc::test::TempFilename(webrtc::test::OutputPath(), "aec_dump");
+ test::TempFilename(test::OutputPath(), "aec_dump");
{
- std::unique_ptr<webrtc::AecDump> aec_dump =
- webrtc::AecDumpFactory::Create(filename, -1, file_writer_queue.Get());
+ std::unique_ptr<AecDump> aec_dump =
+ AecDumpFactory::Create(filename, -1, file_writer_queue.Get());
constexpr int kNumChannels = 1;
constexpr int kNumSamplesPerChannel = 160;
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc
index ff9bb18..672090b 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc
@@ -69,7 +69,7 @@
/*layer_name=*/"FC");
constexpr int kNumTests = 10000;
- ::webrtc::test::PerformanceTimer perf_timer(kNumTests);
+ test::PerformanceTimer perf_timer(kNumTests);
for (int k = 0; k < kNumTests; ++k) {
perf_timer.StartTimer();
fc.ComputeOutput(kFullyConnectedInputVector);
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
index 43b3e68..5afd438 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
@@ -141,7 +141,7 @@
input_sequence.size() / kInputLayerOutputSize;
constexpr int kNumTests = 100;
- ::webrtc::test::PerformanceTimer perf_timer(kNumTests);
+ test::PerformanceTimer perf_timer(kNumTests);
for (int k = 0; k < kNumTests; ++k) {
perf_timer.StartTimer();
for (int i = 0; i < input_sequence_length; ++i) {
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
index f33cd14..b9991ea 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
@@ -135,7 +135,7 @@
std::array<float, kFeatureVectorSize> feature_vector;
RnnVad rnn_vad(cpu_features);
constexpr int number_of_tests = 100;
- ::webrtc::test::PerformanceTimer perf_timer(number_of_tests);
+ test::PerformanceTimer perf_timer(number_of_tests);
for (int k = 0; k < number_of_tests; ++k) {
features_extractor.Reset();
rnn_vad.Reset();
diff --git a/modules/audio_processing/agc2/rnn_vad/test_utils.cc b/modules/audio_processing/agc2/rnn_vad/test_utils.cc
index a3951bf..aa78b70 100644
--- a/modules/audio_processing/agc2/rnn_vad/test_utils.cc
+++ b/modules/audio_processing/agc2/rnn_vad/test_utils.cc
@@ -68,7 +68,7 @@
} // namespace
-using webrtc::test::ResourcePath;
+using test::ResourcePath;
void ExpectEqualFloatArray(ArrayView<const float> expected,
ArrayView<const float> computed) {
diff --git a/modules/audio_processing/agc2/vad_wrapper_unittest.cc b/modules/audio_processing/agc2/vad_wrapper_unittest.cc
index a4c1dcba..4cac82d 100644
--- a/modules/audio_processing/agc2/vad_wrapper_unittest.cc
+++ b/modules/audio_processing/agc2/vad_wrapper_unittest.cc
@@ -42,10 +42,7 @@
public:
MOCK_METHOD(int, SampleRateHz, (), (const, override));
MOCK_METHOD(void, Reset, (), (override));
- MOCK_METHOD(float,
- Analyze,
- (webrtc::ArrayView<const float> frame),
- (override));
+ MOCK_METHOD(float, Analyze, (ArrayView<const float> frame), (override));
};
// Checks that the ctor and `Initialize()` read the sample rate of the wrapped
diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index 5d4b04e..e761f76 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -204,7 +204,7 @@
TEST(AudioProcessingImplTest, UpdateCapturePreGainRuntimeSetting) {
scoped_refptr<AudioProcessing> apm =
BuiltinAudioProcessingBuilder().Build(CreateEnvironment());
- webrtc::AudioProcessing::Config apm_config;
+ AudioProcessing::Config apm_config;
apm_config.pre_amplifier.enabled = true;
apm_config.pre_amplifier.fixed_gain_factor = 1.f;
apm->ApplyConfig(apm_config);
@@ -237,7 +237,7 @@
LevelAdjustmentUpdateCapturePreGainRuntimeSetting) {
scoped_refptr<AudioProcessing> apm =
BuiltinAudioProcessingBuilder().Build(CreateEnvironment());
- webrtc::AudioProcessing::Config apm_config;
+ AudioProcessing::Config apm_config;
apm_config.capture_level_adjustment.enabled = true;
apm_config.capture_level_adjustment.pre_gain_factor = 1.f;
apm->ApplyConfig(apm_config);
@@ -270,7 +270,7 @@
LevelAdjustmentUpdateCapturePostGainRuntimeSetting) {
scoped_refptr<AudioProcessing> apm =
BuiltinAudioProcessingBuilder().Build(CreateEnvironment());
- webrtc::AudioProcessing::Config apm_config;
+ AudioProcessing::Config apm_config;
apm_config.capture_level_adjustment.enabled = true;
apm_config.capture_level_adjustment.post_gain_factor = 1.f;
apm->ApplyConfig(apm_config);
@@ -393,7 +393,7 @@
.SetEchoControlFactory(std::move(echo_control_factory))
.Build(CreateEnvironment());
// Disable AGC.
- webrtc::AudioProcessing::Config apm_config;
+ AudioProcessing::Config apm_config;
apm_config.gain_controller1.enabled = false;
apm_config.gain_controller2.enabled = false;
apm_config.pre_amplifier.enabled = true;
@@ -436,7 +436,7 @@
.SetEchoControlFactory(std::move(echo_control_factory))
.Build(CreateEnvironment());
// Disable AGC.
- webrtc::AudioProcessing::Config apm_config;
+ AudioProcessing::Config apm_config;
apm_config.gain_controller1.enabled = false;
apm_config.gain_controller2.enabled = false;
apm_config.capture_level_adjustment.enabled = true;
@@ -478,7 +478,7 @@
BuiltinAudioProcessingBuilder()
.SetEchoControlFactory(std::move(echo_control_factory))
.Build(CreateEnvironment());
- webrtc::AudioProcessing::Config apm_config;
+ AudioProcessing::Config apm_config;
// Enable AGC1.
apm_config.gain_controller1.enabled = true;
apm_config.gain_controller1.analog_gain_controller.enabled = true;
@@ -534,7 +534,7 @@
.SetEchoControlFactory(std::move(echo_control_factory))
.Build(CreateEnvironment());
// Disable AGC.
- webrtc::AudioProcessing::Config apm_config;
+ AudioProcessing::Config apm_config;
apm_config.gain_controller1.enabled = false;
apm_config.gain_controller2.enabled = false;
apm->ApplyConfig(apm_config);
@@ -591,7 +591,7 @@
.SetEchoDetector(test_echo_detector)
.SetRenderPreProcessing(std::move(test_render_pre_processor))
.Build(CreateEnvironment());
- webrtc::AudioProcessing::Config apm_config;
+ AudioProcessing::Config apm_config;
apm_config.pre_amplifier.enabled = true;
apm->ApplyConfig(apm_config);
@@ -643,7 +643,7 @@
// is never modified.
TEST_P(StartupInputVolumeParameterizedTest,
WithNoInputVolumeControllerStartupVolumeNotModified) {
- webrtc::AudioProcessing::Config config;
+ AudioProcessing::Config config;
config.gain_controller1.enabled = false;
config.gain_controller2.enabled = false;
auto apm = BuiltinAudioProcessingBuilder(config).Build(CreateEnvironment());
@@ -662,7 +662,7 @@
// volume always matches the applied one.
TEST(AudioProcessingImplTest,
WithNoInputVolumeControllerAppliedAndRecommendedVolumesMatch) {
- webrtc::AudioProcessing::Config config;
+ AudioProcessing::Config config;
config.gain_controller1.enabled = false;
config.gain_controller2.enabled = false;
auto apm = BuiltinAudioProcessingBuilder(config).Build(CreateEnvironment());
diff --git a/modules/audio_processing/audio_processing_performance_unittest.cc b/modules/audio_processing/audio_processing_performance_unittest.cc
index d12302c..820d403 100644
--- a/modules/audio_processing/audio_processing_performance_unittest.cc
+++ b/modules/audio_processing/audio_processing_performance_unittest.cc
@@ -33,10 +33,10 @@
namespace webrtc {
namespace {
-using ::webrtc::test::GetGlobalMetricsLogger;
-using ::webrtc::test::ImprovementDirection;
-using ::webrtc::test::Metric;
-using ::webrtc::test::Unit;
+using test::GetGlobalMetricsLogger;
+using test::ImprovementDirection;
+using test::Metric;
+using test::Unit;
class CallSimulator;
@@ -206,7 +206,7 @@
simulation_config_(simulation_config),
apm_(apm),
frame_data_(kMaxFrameSize),
- clock_(webrtc::Clock::GetRealTimeClock()),
+ clock_(Clock::GetRealTimeClock()),
num_durations_to_store_(num_durations_to_store),
api_call_durations_(num_durations_to_store_ - kNumInitializationFrames),
samples_count_(0),
@@ -355,7 +355,7 @@
const SimulationConfig* const simulation_config_ = nullptr;
AudioProcessing* apm_ = nullptr;
AudioFrameData frame_data_;
- webrtc::Clock* clock_;
+ Clock* clock_;
const size_t num_durations_to_store_;
SamplesStatsCounter api_call_durations_;
size_t samples_count_ = 0;
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 8e32d34..373b574 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -2369,7 +2369,7 @@
void RunApmRateAndChannelTest(ArrayView<const int> sample_rates_hz,
ArrayView<const int> render_channel_counts,
ArrayView<const int> capture_channel_counts) {
- webrtc::AudioProcessing::Config apm_config;
+ AudioProcessing::Config apm_config;
apm_config.pipeline.multi_channel_render = true;
apm_config.pipeline.multi_channel_capture = true;
apm_config.echo_canceller.enabled = true;
diff --git a/modules/audio_processing/test/aec_dump_based_simulator.cc b/modules/audio_processing/test/aec_dump_based_simulator.cc
index 896c4b8..6a249ee 100644
--- a/modules/audio_processing/test/aec_dump_based_simulator.cc
+++ b/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -45,7 +45,7 @@
// Verify output bitexactness for the fixed interface.
// TODO(peah): Check whether it would make sense to add a threshold
// to use for checking the bitexactness in a soft manner.
-bool VerifyFixedBitExactness(const webrtc::audioproc::Stream& msg,
+bool VerifyFixedBitExactness(const audioproc::Stream& msg,
const Int16Frame& frame) {
if (sizeof(frame.data[0]) * frame.data.size() != msg.output_data().size()) {
return false;
@@ -61,7 +61,7 @@
}
// Verify output bitexactness for the float interface.
-bool VerifyFloatBitExactness(const webrtc::audioproc::Stream& msg,
+bool VerifyFloatBitExactness(const audioproc::Stream& msg,
const StreamConfig& out_config,
const ChannelBuffer<float>& out_buf) {
if (static_cast<size_t>(msg.output_channel_size()) !=
@@ -86,7 +86,7 @@
bool ReadNextMessage(bool use_dump_file,
FILE* dump_input_file,
std::stringstream& input,
- webrtc::audioproc::Event& event_msg) {
+ audioproc::Event& event_msg) {
if (use_dump_file) {
return ReadMessageFromFile(dump_input_file, &event_msg);
}
@@ -105,7 +105,7 @@
AecDumpBasedSimulator::~AecDumpBasedSimulator() = default;
void AecDumpBasedSimulator::PrepareProcessStreamCall(
- const webrtc::audioproc::Stream& msg) {
+ const audioproc::Stream& msg) {
if (msg.has_input_data()) {
// Fixed interface processing.
// Verify interface invariance.
@@ -196,7 +196,7 @@
}
void AecDumpBasedSimulator::VerifyProcessStreamBitExactness(
- const webrtc::audioproc::Stream& msg) {
+ const audioproc::Stream& msg) {
if (bitexact_output_) {
if (interface_used_ == InterfaceType::kFixedInterface) {
bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_);
@@ -207,7 +207,7 @@
}
void AecDumpBasedSimulator::PrepareReverseProcessStreamCall(
- const webrtc::audioproc::ReverseStream& msg) {
+ const audioproc::ReverseStream& msg) {
if (msg.has_data()) {
// Fixed interface processing.
// Verify interface invariance.
@@ -267,7 +267,7 @@
input << settings_.aec_dump_input_string.value();
}
- webrtc::audioproc::Event event_msg;
+ audioproc::Event event_msg;
int capture_frames_since_init = 0;
int init_index = 0;
while (ReadNextMessage(use_dump_file, dump_input_file_, input, event_msg)) {
@@ -300,12 +300,12 @@
input << settings_.aec_dump_input_string.value();
}
- webrtc::audioproc::Event event_msg;
+ audioproc::Event event_msg;
int num_capture_frames = 0;
int num_render_frames = 0;
int init_index = 0;
while (ReadNextMessage(use_dump_file, dump_input_file_, input, event_msg)) {
- if (event_msg.type() == webrtc::audioproc::Event::INIT) {
+ if (event_msg.type() == audioproc::Event::INIT) {
++init_index;
constexpr float kNumFramesPerSecond = 100.f;
float capture_time_seconds = num_capture_frames / kNumFramesPerSecond;
@@ -318,9 +318,9 @@
<< num_capture_frames << " frames) " << std::endl;
std::cout << " Render: " << render_time_seconds << " s ("
<< num_render_frames << " frames) " << std::endl;
- } else if (event_msg.type() == webrtc::audioproc::Event::STREAM) {
+ } else if (event_msg.type() == audioproc::Event::STREAM) {
++num_capture_frames;
- } else if (event_msg.type() == webrtc::audioproc::Event::REVERSE_STREAM) {
+ } else if (event_msg.type() == audioproc::Event::REVERSE_STREAM) {
++num_render_frames;
}
}
@@ -330,40 +330,38 @@
}
}
-void AecDumpBasedSimulator::HandleEvent(
- const webrtc::audioproc::Event& event_msg,
- int& capture_frames_since_init,
- int& init_index) {
+void AecDumpBasedSimulator::HandleEvent(const audioproc::Event& event_msg,
+ int& capture_frames_since_init,
+ int& init_index) {
switch (event_msg.type()) {
- case webrtc::audioproc::Event::INIT:
+ case audioproc::Event::INIT:
RTC_CHECK(event_msg.has_init());
++init_index;
capture_frames_since_init = 0;
HandleMessage(event_msg.init(), init_index);
break;
- case webrtc::audioproc::Event::STREAM:
+ case audioproc::Event::STREAM:
RTC_CHECK(event_msg.has_stream());
++capture_frames_since_init;
HandleMessage(event_msg.stream());
break;
- case webrtc::audioproc::Event::REVERSE_STREAM:
+ case audioproc::Event::REVERSE_STREAM:
RTC_CHECK(event_msg.has_reverse_stream());
HandleMessage(event_msg.reverse_stream());
break;
- case webrtc::audioproc::Event::CONFIG:
+ case audioproc::Event::CONFIG:
RTC_CHECK(event_msg.has_config());
HandleMessage(event_msg.config());
break;
- case webrtc::audioproc::Event::RUNTIME_SETTING:
+ case audioproc::Event::RUNTIME_SETTING:
HandleMessage(event_msg.runtime_setting());
break;
- case webrtc::audioproc::Event::UNKNOWN_EVENT:
+ case audioproc::Event::UNKNOWN_EVENT:
RTC_CHECK_NOTREACHED();
}
}
-void AecDumpBasedSimulator::HandleMessage(
- const webrtc::audioproc::Config& msg) {
+void AecDumpBasedSimulator::HandleMessage(const audioproc::Config& msg) {
if (settings_.use_verbose_logging) {
std::cout << "Config at frame:" << std::endl;
std::cout << " Forward: " << get_num_process_stream_calls() << std::endl;
@@ -403,7 +401,7 @@
}
if (msg.has_aecm_routing_mode() &&
- static_cast<webrtc::EchoControlMobileImpl::RoutingMode>(
+ static_cast<EchoControlMobileImpl::RoutingMode>(
msg.aecm_routing_mode()) != EchoControlMobileImpl::kSpeakerphone) {
RTC_LOG(LS_ERROR) << "Ignoring deprecated setting: AECM routing mode: "
<< msg.aecm_routing_mode();
@@ -421,8 +419,7 @@
if (msg.has_agc_mode() || settings_.agc_mode) {
int mode = settings_.agc_mode ? *settings_.agc_mode : msg.agc_mode();
apm_config.gain_controller1.mode =
- static_cast<webrtc::AudioProcessing::Config::GainController1::Mode>(
- mode);
+ static_cast<AudioProcessing::Config::GainController1::Mode>(mode);
if (settings_.use_verbose_logging) {
std::cout << " agc_mode: " << mode << std::endl;
}
@@ -524,7 +521,7 @@
}
}
-void AecDumpBasedSimulator::HandleMessage(const webrtc::audioproc::Init& msg,
+void AecDumpBasedSimulator::HandleMessage(const audioproc::Init& msg,
int init_index) {
RTC_CHECK(msg.has_sample_rate());
RTC_CHECK(msg.has_num_input_channels());
@@ -586,8 +583,7 @@
msg.num_reverse_channels(), num_reverse_output_channels);
}
-void AecDumpBasedSimulator::HandleMessage(
- const webrtc::audioproc::Stream& msg) {
+void AecDumpBasedSimulator::HandleMessage(const audioproc::Stream& msg) {
if (call_order_output_file_) {
*call_order_output_file_ << "c";
}
@@ -596,8 +592,7 @@
VerifyProcessStreamBitExactness(msg);
}
-void AecDumpBasedSimulator::HandleMessage(
- const webrtc::audioproc::ReverseStream& msg) {
+void AecDumpBasedSimulator::HandleMessage(const audioproc::ReverseStream& msg) {
if (call_order_output_file_) {
*call_order_output_file_ << "r";
}
@@ -606,7 +601,7 @@
}
void AecDumpBasedSimulator::HandleMessage(
- const webrtc::audioproc::RuntimeSetting& msg) {
+ const audioproc::RuntimeSetting& msg) {
RTC_CHECK(ap_.get());
if (msg.has_capture_pre_gain()) {
// Handle capture pre-gain runtime setting only if not overridden.
diff --git a/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc
index a273083..914edb7 100644
--- a/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/modules/audio_processing/test/audio_processing_simulator.cc
@@ -503,7 +503,7 @@
}
if (settings_.agc_mode) {
apm_config.gain_controller1.mode =
- static_cast<webrtc::AudioProcessing::Config::GainController1::Mode>(
+ static_cast<AudioProcessing::Config::GainController1::Mode>(
*settings_.agc_mode);
}
if (settings_.use_agc_limiter) {
diff --git a/modules/audio_processing/test/performance_timer.cc b/modules/audio_processing/test/performance_timer.cc
index 1a82258..cc12e00 100644
--- a/modules/audio_processing/test/performance_timer.cc
+++ b/modules/audio_processing/test/performance_timer.cc
@@ -20,7 +20,7 @@
namespace test {
PerformanceTimer::PerformanceTimer(int num_frames_to_process)
- : clock_(webrtc::Clock::GetRealTimeClock()) {
+ : clock_(Clock::GetRealTimeClock()) {
timestamps_us_.reserve(num_frames_to_process);
}
diff --git a/modules/audio_processing/test/runtime_setting_util.cc b/modules/audio_processing/test/runtime_setting_util.cc
index 4899d2d..2240768 100644
--- a/modules/audio_processing/test/runtime_setting_util.cc
+++ b/modules/audio_processing/test/runtime_setting_util.cc
@@ -15,7 +15,7 @@
namespace webrtc {
void ReplayRuntimeSetting(AudioProcessing* apm,
- const webrtc::audioproc::RuntimeSetting& setting) {
+ const audioproc::RuntimeSetting& setting) {
RTC_CHECK(apm);
// TODO(bugs.webrtc.org/9138): Add ability to handle different types
// of settings. Currently CapturePreGain, CaptureFixedPostGain and
