Remove redundant webrtc:: prefixes in modules/audio_processing
Created by
tools_webrtc/remove_extra_namespace.py --namespace webrtc
and manual adjustments.
This CL was uploaded by git cl split.
R=eshr@webrtc.org
No-IWYU: Refactoring
Bug: webrtc:42232595
Change-Id: I1247a030b324697348ab77c7e6c1b92062409f01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/396183
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44895}
diff --git a/modules/audio_processing/aecm/aecm_core_mips.cc b/modules/audio_processing/aecm/aecm_core_mips.cc
index 819737d..2d008b0 100644
--- a/modules/audio_processing/aecm/aecm_core_mips.cc
+++ b/modules/audio_processing/aecm/aecm_core_mips.cc
@@ -1078,8 +1078,7 @@
// Far end signal through channel estimate in Q8
// How much can we shift right to preserve resolution
tmp32no1 = echoEst32[i] - aecm->echoFilt[i];
- aecm->echoFilt[i] +=
- webrtc::dchecked_cast<int32_t>((int64_t{tmp32no1} * 50) >> 8);
+ aecm->echoFilt[i] += dchecked_cast<int32_t>((int64_t{tmp32no1} * 50) >> 8);
zeros32 = WebRtcSpl_NormW32(aecm->echoFilt[i]) + 1;
zeros16 = WebRtcSpl_NormW16(supGain) + 1;
diff --git a/modules/audio_processing/agc/mock_agc.h b/modules/audio_processing/agc/mock_agc.h
index 9d76d7c..548358b 100644
--- a/modules/audio_processing/agc/mock_agc.h
+++ b/modules/audio_processing/agc/mock_agc.h
@@ -22,10 +22,7 @@
class MockAgc : public Agc {
public:
virtual ~MockAgc() {}
- MOCK_METHOD(void,
- Process,
- (webrtc::ArrayView<const int16_t> audio),
- (override));
+ MOCK_METHOD(void, Process, (ArrayView<const int16_t> audio), (override));
MOCK_METHOD(bool, GetRmsErrorDb, (int* error), (override));
MOCK_METHOD(void, Reset, (), (override));
MOCK_METHOD(int, set_target_level_dbfs, (int level), (override));
diff --git a/modules/audio_processing/agc2/biquad_filter_unittest.cc b/modules/audio_processing/agc2/biquad_filter_unittest.cc
index d5fa538..db29bfe 100644
--- a/modules/audio_processing/agc2/biquad_filter_unittest.cc
+++ b/modules/audio_processing/agc2/biquad_filter_unittest.cc
@@ -58,7 +58,7 @@
{{24.84286614f, -62.18094158f, 57.91488056f, -106.65685933f, 13.38760103f,
-36.60367134f, -94.44880104f, -3.59920354f}}}};
-// Fails for every pair from two equally sized webrtc::ArrayView<float> views
+// Fails for every pair from two equally sized ArrayView<float> views
// such that their relative error is above a given threshold. If the expected
// value of a pair is 0, `tolerance` is used to check the absolute error.
void ExpectNearRelative(ArrayView<const float> expected,
diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h
index a876aa8..6f11a70 100644
--- a/modules/audio_processing/audio_buffer.h
+++ b/modules/audio_processing/audio_buffer.h
@@ -36,7 +36,7 @@
public:
static const int kSplitBandSize = 160;
// TODO(tommi): Remove this (`AudioBuffer::kMaxSampleRate`) constant.
- static const int kMaxSampleRate = webrtc::kMaxSampleRateHz;
+ static const int kMaxSampleRate = kMaxSampleRateHz;
AudioBuffer(size_t input_rate,
size_t input_num_channels,
size_t buffer_rate,
diff --git a/modules/audio_processing/logging/apm_data_dumper.cc b/modules/audio_processing/logging/apm_data_dumper.cc
index 7c81e78..b26d90f 100644
--- a/modules/audio_processing/logging/apm_data_dumper.cc
+++ b/modules/audio_processing/logging/apm_data_dumper.cc
@@ -34,7 +34,7 @@
int reinit_index,
absl::string_view suffix) {
char buf[1024];
- webrtc::SimpleStringBuilder ss(buf);
+ SimpleStringBuilder ss(buf);
if (!output_dir.empty()) {
ss << output_dir;
if (output_dir.back() != kPathDelimiter) {
diff --git a/modules/audio_processing/logging/apm_data_dumper.h b/modules/audio_processing/logging/apm_data_dumper.h
index 4297997..24e5ad9 100644
--- a/modules/audio_processing/logging/apm_data_dumper.h
+++ b/modules/audio_processing/logging/apm_data_dumper.h
@@ -90,7 +90,7 @@
[[maybe_unused]] absl::string_view output_dir) {
#if WEBRTC_APM_DEBUG_DUMP == 1
RTC_CHECK_LT(output_dir.size(), kOutputDirMaxLength);
- webrtc::strcpyn(output_dir_, kOutputDirMaxLength, output_dir);
+ strcpyn(output_dir_, kOutputDirMaxLength, output_dir);
#endif
}
diff --git a/modules/audio_processing/ns/noise_suppressor_unittest.cc b/modules/audio_processing/ns/noise_suppressor_unittest.cc
index 119b939..1cb4563 100644
--- a/modules/audio_processing/ns/noise_suppressor_unittest.cc
+++ b/modules/audio_processing/ns/noise_suppressor_unittest.cc
@@ -72,7 +72,7 @@
SCOPED_TRACE(ProduceDebugText(rate, num_channels, level));
const size_t num_bands = rate / 16000;
- // const int frame_length = webrtc::CheckedDivExact(rate, 100);
+ // const int frame_length = CheckedDivExact(rate, 100);
AudioBuffer audio(rate, num_channels, rate, num_channels, rate,
num_channels);
NsConfig cfg;
diff --git a/modules/audio_processing/test/runtime_setting_util.h b/modules/audio_processing/test/runtime_setting_util.h
index 85ed5ec..54f826a 100644
--- a/modules/audio_processing/test/runtime_setting_util.h
+++ b/modules/audio_processing/test/runtime_setting_util.h
@@ -17,7 +17,7 @@
namespace webrtc {
void ReplayRuntimeSetting(AudioProcessing* apm,
- const webrtc::audioproc::RuntimeSetting& setting);
+ const audioproc::RuntimeSetting& setting);
}
#endif // MODULES_AUDIO_PROCESSING_TEST_RUNTIME_SETTING_UTIL_H_
diff --git a/modules/audio_processing/utility/pffft_wrapper.h b/modules/audio_processing/utility/pffft_wrapper.h
index b540a2f..590310e 100644
--- a/modules/audio_processing/utility/pffft_wrapper.h
+++ b/modules/audio_processing/utility/pffft_wrapper.h
@@ -68,7 +68,7 @@
// Creates a buffer of the right size.
std::unique_ptr<FloatBuffer> CreateBuffer() const;
- // TODO(https://crbug.com/webrtc/9577): Overload with webrtc::ArrayView args.
+ // TODO(https://crbug.com/webrtc/9577): Overload with ArrayView args.
// Computes the forward fast Fourier transform.
void ForwardTransform(const FloatBuffer& in, FloatBuffer* out, bool ordered);
// Computes the backward fast Fourier transform.
diff --git a/modules/audio_processing/vad/voice_activity_detector.h b/modules/audio_processing/vad/voice_activity_detector.h
index 401acca..5754e89 100644
--- a/modules/audio_processing/vad/voice_activity_detector.h
+++ b/modules/audio_processing/vad/voice_activity_detector.h
@@ -33,7 +33,7 @@
~VoiceActivityDetector();
// Processes each audio chunk and estimates the voice probability.
- // TODO(bugs.webrtc.org/7494): Switch to webrtc::ArrayView and remove
+ // TODO(bugs.webrtc.org/7494): Switch to ArrayView and remove
// `sample_rate_hz`.
void ProcessChunk(const int16_t* audio, size_t length, int sample_rate_hz);
