Use backticks not vertical bars to denote variables in comments for /modules/audio_processing

Bug: webrtc:12338
Change-Id: I85bff694dd2ead83c939c4d1945eff82e1296001
No-Presubmit: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227161
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34690}

modules/audio_processing/test/audio_processing_simulator.cc

@@ -206,7 +206,7 @@ void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
   if (settings_.simulate_mic_gain) {
     if (settings_.aec_dump_input_filename) {
       // When the analog gain is simulated and an AEC dump is used as input, set
-      // the undo level to |aec_dump_mic_level_| to virtually restore the
+      // the undo level to `aec_dump_mic_level_` to virtually restore the
       // unmodified microphone signal level.
       fake_recording_device_.SetUndoMicLevel(aec_dump_mic_level_);
     }
@@ -261,7 +261,7 @@ void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
 
   // Store the mic level suggested by AGC.
   // Note that when the analog gain is simulated and an AEC dump is used as
-  // input, |analog_mic_level_| will not be used with set_stream_analog_level().
+  // input, `analog_mic_level_` will not be used with set_stream_analog_level().
   analog_mic_level_ = ap_->recommended_stream_analog_level();
   if (settings_.simulate_mic_gain) {
     fake_recording_device_.SetMicLevel(analog_mic_level_);

modules/audio_processing/test/audioproc_float_impl.h

@@ -19,11 +19,11 @@ namespace webrtc {
 namespace test {
 
 // This function implements the audio processing simulation utility. Pass
-// |input_aecdump| to provide the content of an AEC dump file as a string; if
-// |input_aecdump| is not passed, a WAV or AEC input dump file must be specified
-// via the |argv| argument. Pass |processed_capture_samples| to write in it the
-// samples processed on the capture side; if |processed_capture_samples| is not
-// passed, the output file can optionally be specified via the |argv| argument.
+// `input_aecdump` to provide the content of an AEC dump file as a string; if
+// `input_aecdump` is not passed, a WAV or AEC input dump file must be specified
+// via the `argv` argument. Pass `processed_capture_samples` to write in it the
+// samples processed on the capture side; if `processed_capture_samples` is not
+// passed, the output file can optionally be specified via the `argv` argument.
 // Any audio_processing object specified in the input is used for the
 // simulation. Note that when the audio_processing object is specified all
 // functionality that relies on using the internal builder is deactivated,
@@ -34,11 +34,11 @@ int AudioprocFloatImpl(rtc::scoped_refptr<AudioProcessing> audio_processing,
                        char* argv[]);
 
 // This function implements the audio processing simulation utility. Pass
-// |input_aecdump| to provide the content of an AEC dump file as a string; if
-// |input_aecdump| is not passed, a WAV or AEC input dump file must be specified
-// via the |argv| argument. Pass |processed_capture_samples| to write in it the
-// samples processed on the capture side; if |processed_capture_samples| is not
-// passed, the output file can optionally be specified via the |argv| argument.
+// `input_aecdump` to provide the content of an AEC dump file as a string; if
+// `input_aecdump` is not passed, a WAV or AEC input dump file must be specified
+// via the `argv` argument. Pass `processed_capture_samples` to write in it the
+// samples processed on the capture side; if `processed_capture_samples` is not
+// passed, the output file can optionally be specified via the `argv` argument.
 int AudioprocFloatImpl(std::unique_ptr<AudioProcessingBuilder> ap_builder,
                        int argc,
                        char* argv[],

modules/audio_processing/test/conversational_speech/simulator.cc

@@ -125,8 +125,8 @@ std::unique_ptr<std::map<std::string, std::vector<int16_t>>> PreloadAudioTracks(
   return audiotracks_map;
 }
 
-// Writes all the values in |source_samples| via |wav_writer|. If the number of
-// previously written samples in |wav_writer| is less than |interval_begin|, it
+// Writes all the values in `source_samples` via `wav_writer`. If the number of
+// previously written samples in `wav_writer` is less than `interval_begin`, it
 // adds zeros as left padding. The padding corresponds to intervals during which
 // a speaker is not active.
 void PadLeftWriteChunk(rtc::ArrayView<const int16_t> source_samples,
@@ -145,9 +145,9 @@ void PadLeftWriteChunk(rtc::ArrayView<const int16_t> source_samples,
   wav_writer->WriteSamples(source_samples.data(), source_samples.size());
 }
 
-// Appends zeros via |wav_writer|. The number of zeros is always non-negative
+// Appends zeros via `wav_writer`. The number of zeros is always non-negative
 // and equal to the difference between the previously written samples and
-// |pad_samples|.
+// `pad_samples`.
 void PadRightWrite(WavWriter* wav_writer, size_t pad_samples) {
   RTC_CHECK(wav_writer);
   RTC_CHECK_GE(pad_samples, wav_writer->num_samples());

modules/audio_processing/test/fake_recording_device.h

@@ -52,14 +52,14 @@ class FakeRecordingDevice final {
   void SetUndoMicLevel(const int level);
 
   // Simulates the analog gain.
-  // If |real_device_level| is a valid level, the unmodified mic signal is
-  // virtually restored. To skip the latter step set |real_device_level| to
+  // If `real_device_level` is a valid level, the unmodified mic signal is
+  // virtually restored. To skip the latter step set `real_device_level` to
   // an empty value.
   void SimulateAnalogGain(rtc::ArrayView<int16_t> buffer);
 
   // Simulates the analog gain.
-  // If |real_device_level| is a valid level, the unmodified mic signal is
-  // virtually restored. To skip the latter step set |real_device_level| to
+  // If `real_device_level` is a valid level, the unmodified mic signal is
+  // virtually restored. To skip the latter step set `real_device_level` to
   // an empty value.
   void SimulateAnalogGain(ChannelBuffer<float>* buffer);
 

modules/audio_processing/test/fake_recording_device_unittest.cc

@@ -75,7 +75,7 @@ void CheckIfMonotoneSamplesModules(const ChannelBuffer<float>* prev,
 }
 
 // Checks that the samples in each pair have the same sign unless the sample in
-// |dst| is zero (because of zero gain).
+// `dst` is zero (because of zero gain).
 void CheckSameSign(const ChannelBuffer<float>* src,
                    const ChannelBuffer<float>* dst) {
   RTC_DCHECK_EQ(src->num_channels(), dst->num_channels());

modules/audio_processing/test/performance_timer.h

@@ -31,7 +31,7 @@ class PerformanceTimer {
   double GetDurationStandardDeviation() const;
 
   // These methods are the same as those above, but they ignore the first
-  // |number_of_warmup_samples| measurements.
+  // `number_of_warmup_samples` measurements.
   double GetDurationAverage(size_t number_of_warmup_samples) const;
   double GetDurationStandardDeviation(size_t number_of_warmup_samples) const;
 

modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_boxplot.py

@@ -88,7 +88,7 @@ def FilterScoresByParams(data_frame, filter_params, score_name, config_dir):
         data_cell_scores = data_with_config[data_with_config.eval_score_name ==
                                             score_name]
 
-        # Exactly one of |params_to_plot| must match:
+        # Exactly one of `params_to_plot` must match:
         (matching_param, ) = [
             x for x in filter_params if '-' + x in config_json
         ]

modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py

@@ -133,7 +133,7 @@ def _FindOptimalParameter(configs_and_scores, score_weighting):
                                 {score1: value1, ...}}] into a numeric
                      value
   Returns:
-    the config that has the largest values of |score_weighting| applied
+    the config that has the largest values of `score_weighting` applied
     to its scores.
   """
 

modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores.py

@@ -397,7 +397,7 @@ class TotalHarmonicDistorsionScore(EvaluationScore):
         # TODO(alessiob): Fix or remove if not needed.
         # thd = np.sqrt(np.sum(b_terms[1:]**2)) / b_terms[0]
 
-        # TODO(alessiob): Check the range of |thd_plus_noise| and update the class
+        # TODO(alessiob): Check the range of `thd_plus_noise` and update the class
         # docstring above if accordingly.
         thd_plus_noise = distortion_and_noise / b_terms[0]
 

modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py

@@ -363,7 +363,7 @@ class HtmlExport(object):
     @classmethod
     def _SliceDataForScoreStatsTableCell(cls, scores, capture, render,
                                          echo_simulator):
-        """Slices |scores| to extract the data for a tab."""
+        """Slices `scores` to extract the data for a tab."""
         masks = []
 
         masks.append(scores.capture == capture)
@@ -378,7 +378,7 @@ class HtmlExport(object):
 
     @classmethod
     def _FindUniqueTuples(cls, data_frame, fields):
-        """Slices |data_frame| to a list of fields and finds unique tuples."""
+        """Slices `data_frame` to a list of fields and finds unique tuples."""
         return data_frame[fields].drop_duplicates().values.tolist()
 
     @classmethod

modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer.py

@@ -47,7 +47,7 @@ class ApmInputMixer(object):
 
     Hard-clipping may occur in the mix; a warning is raised when this happens.
 
-    If |echo_filepath| is None, nothing is done and |capture_input_filepath| is
+    If `echo_filepath` is None, nothing is done and `capture_input_filepath` is
     returned.
 
     Args:

modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py

@@ -174,7 +174,7 @@ class SignalProcessingUtils(object):
         """Detects hard clipping.
 
     Hard clipping is simply detected by counting samples that touch either the
-    lower or upper bound too many times in a row (according to |threshold|).
+    lower or upper bound too many times in a row (according to `threshold`).
     The presence of a single sequence of samples meeting such property is enough
     to label the signal as hard clipped.
 
@@ -295,16 +295,16 @@ class SignalProcessingUtils(object):
                    noise,
                    target_snr=0.0,
                    pad_noise=MixPadding.NO_PADDING):
-        """Mixes |signal| and |noise| with a target SNR.
+        """Mixes `signal` and `noise` with a target SNR.
 
-    Mix |signal| and |noise| with a desired SNR by scaling |noise|.
+    Mix `signal` and `noise` with a desired SNR by scaling `noise`.
     If the target SNR is +/- infinite, a copy of signal/noise is returned.
-    If |signal| is shorter than |noise|, the length of the mix equals that of
-    |signal|. Otherwise, the mix length depends on whether padding is applied.
-    When padding is not applied, that is |pad_noise| is set to NO_PADDING
-    (default), the mix length equals that of |noise| - i.e., |signal| is
-    truncated. Otherwise, |noise| is extended and the resulting mix has the same
-    length of |signal|.
+    If `signal` is shorter than `noise`, the length of the mix equals that of
+    `signal`. Otherwise, the mix length depends on whether padding is applied.
+    When padding is not applied, that is `pad_noise` is set to NO_PADDING
+    (default), the mix length equals that of `noise` - i.e., `signal` is
+    truncated. Otherwise, `noise` is extended and the resulting mix has the same
+    length of `signal`.
 
     Args:
       signal: AudioSegment instance (signal).
@@ -342,18 +342,18 @@ class SignalProcessingUtils(object):
         signal_duration = len(signal)
         noise_duration = len(noise)
         if signal_duration <= noise_duration:
-            # Ignore |pad_noise|, |noise| is truncated if longer that |signal|, the
-            # mix will have the same length of |signal|.
+            # Ignore `pad_noise`, `noise` is truncated if longer that `signal`, the
+            # mix will have the same length of `signal`.
             return signal.overlay(noise.apply_gain(gain_db))
         elif pad_noise == cls.MixPadding.NO_PADDING:
-            # |signal| is longer than |noise|, but no padding is applied to |noise|.
-            # Truncate |signal|.
+            # `signal` is longer than `noise`, but no padding is applied to `noise`.
+            # Truncate `signal`.
             return noise.overlay(signal, gain_during_overlay=gain_db)
         elif pad_noise == cls.MixPadding.ZERO_PADDING:
             # TODO(alessiob): Check that this works as expected.
             return signal.overlay(noise.apply_gain(gain_db))
         elif pad_noise == cls.MixPadding.LOOP:
-            # |signal| is longer than |noise|, extend |noise| by looping.
+            # `signal` is longer than `noise`, extend `noise` by looping.
             return signal.overlay(noise.apply_gain(gain_db), loop=True)
         else:
             raise exceptions.SignalProcessingException('invalid padding type')

modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation.py

@@ -264,7 +264,7 @@ class ApmModuleSimulator(object):
 
     The file name is parsed to extract input signal creator and params. If a
     creator is matched and the parameters are valid, a new signal is generated
-    and written in |input_signal_filepath|.
+    and written in `input_signal_filepath`.
 
     Args:
       input_signal_filepath: Path to the input signal audio file to write.

modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_unittest.py

@@ -116,7 +116,7 @@ class TestTestDataGenerators(unittest.TestCase):
             key = noisy_signal_filepaths.keys()[0]
             return noisy_signal_filepaths[key], reference_signal_filepaths[key]
 
-        # Test the |copy_with_identity| flag.
+        # Test the `copy_with_identity` flag.
         for copy_with_identity in [False, True]:
             # Instance the generator through the factory.
             factory = test_data_generation_factory.TestDataGeneratorFactory(
@@ -126,7 +126,7 @@ class TestTestDataGenerators(unittest.TestCase):
             factory.SetOutputDirectoryPrefix('datagen-')
             generator = factory.GetInstance(
                 test_data_generation.IdentityTestDataGenerator)
-            # Check |copy_with_identity| is set correctly.
+            # Check `copy_with_identity` is set correctly.
             self.assertEqual(copy_with_identity, generator.copy_with_identity)
 
             # Generate test data and extract the paths to the noise and the reference
@@ -137,7 +137,7 @@ class TestTestDataGenerators(unittest.TestCase):
             noisy_signal_filepath, reference_signal_filepath = (
                 GetNoiseReferenceFilePaths(generator))
 
-            # Check that a copy is made if and only if |copy_with_identity| is True.
+            # Check that a copy is made if and only if `copy_with_identity` is True.
             if copy_with_identity:
                 self.assertNotEqual(noisy_signal_filepath,
                                     input_signal_filepath)

modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc

@@ -63,7 +63,7 @@ int main(int argc, char* argv[]) {
   std::unique_ptr<Vad> vad = CreateVad(Vad::Aggressiveness::kVadNormal);
   std::array<int16_t, kMaxFrameLen> samples;
   char buff = 0;     // Buffer to write one bit per frame.
-  uint8_t next = 0;  // Points to the next bit to write in |buff|.
+  uint8_t next = 0;  // Points to the next bit to write in `buff`.
   while (true) {
     // Process frame.
     const auto read_samples =

modules/audio_processing/test/test_utils.h

@@ -78,7 +78,7 @@ class ChannelBufferWavReader final {
   explicit ChannelBufferWavReader(std::unique_ptr<WavReader> file);
   ~ChannelBufferWavReader();
 
-  // Reads data from the file according to the |buffer| format. Returns false if
+  // Reads data from the file according to the `buffer` format. Returns false if
   // a full buffer can't be read from the file.
   bool Read(ChannelBuffer<float>* buffer);
 
@@ -115,7 +115,7 @@ class ChannelBufferVectorWriter final {
       delete;
   ~ChannelBufferVectorWriter();
 
-  // Creates an interleaved copy of |buffer|, converts the samples to float S16
+  // Creates an interleaved copy of `buffer`, converts the samples to float S16
   // and appends the result to output_.
   void Write(const ChannelBuffer<float>& buffer);