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/agc/agc.h

@@ -24,13 +24,13 @@ class Agc {
   Agc();
   virtual ~Agc();
 
-  // |audio| must be mono; in a multi-channel stream, provide the first (usually
+  // `audio` must be mono; in a multi-channel stream, provide the first (usually
   // left) channel.
   virtual void Process(const int16_t* audio, size_t length, int sample_rate_hz);
 
   // Retrieves the difference between the target RMS level and the current
   // signal RMS level in dB. Returns true if an update is available and false
-  // otherwise, in which case |error| should be ignored and no action taken.
+  // otherwise, in which case `error` should be ignored and no action taken.
   virtual bool GetRmsErrorDb(int* error);
   virtual void Reset();
 

modules/audio_processing/agc/agc_manager_direct.cc

@@ -280,7 +280,7 @@ void MonoAgc::SetLevel(int new_level) {
 void MonoAgc::SetMaxLevel(int level) {
   RTC_DCHECK_GE(level, clipped_level_min_);
   max_level_ = level;
-  // Scale the |kSurplusCompressionGain| linearly across the restricted
+  // Scale the `kSurplusCompressionGain` linearly across the restricted
   // level range.
   max_compression_gain_ =
       kMaxCompressionGain + std::floor((1.f * kMaxMicLevel - max_level_) /
@@ -307,7 +307,7 @@ int MonoAgc::CheckVolumeAndReset() {
   int level = stream_analog_level_;
   // Reasons for taking action at startup:
   // 1) A person starting a call is expected to be heard.
-  // 2) Independent of interpretation of |level| == 0 we should raise it so the
+  // 2) Independent of interpretation of `level` == 0 we should raise it so the
   // AGC can do its job properly.
   if (level == 0 && !startup_) {
     RTC_DLOG(LS_INFO)

modules/audio_processing/agc/agc_manager_direct.h

@@ -112,7 +112,7 @@ class AgcManagerDirect final {
   FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest,
                            EnableClippingPredictorLowersVolume);
 
-  // Dependency injection for testing. Don't delete |agc| as the memory is owned
+  // Dependency injection for testing. Don't delete `agc` as the memory is owned
   // by the manager.
   AgcManagerDirect(
       Agc* agc,
@@ -196,7 +196,7 @@ class MonoAgc {
 
   // Set the maximum level the AGC is allowed to apply. Also updates the
   // maximum compression gain to compensate. The level must be at least
-  // |kClippedLevelMin|.
+  // `kClippedLevelMin`.
   void SetMaxLevel(int level);
 
   int CheckVolumeAndReset();

modules/audio_processing/agc/gain_control.h

@@ -20,12 +20,12 @@ namespace webrtc {
 // Recommended to be enabled on the client-side.
 class GainControl {
  public:
-  // When an analog mode is set, this must be called prior to |ProcessStream()|
+  // When an analog mode is set, this must be called prior to `ProcessStream()`
   // to pass the current analog level from the audio HAL. Must be within the
-  // range provided to |set_analog_level_limits()|.
+  // range provided to `set_analog_level_limits()`.
   virtual int set_stream_analog_level(int level) = 0;
 
-  // When an analog mode is set, this should be called after |ProcessStream()|
+  // When an analog mode is set, this should be called after `ProcessStream()`
   // to obtain the recommended new analog level for the audio HAL. It is the
   // users responsibility to apply this level.
   virtual int stream_analog_level() const = 0;
@@ -33,7 +33,7 @@ class GainControl {
   enum Mode {
     // Adaptive mode intended for use if an analog volume control is available
     // on the capture device. It will require the user to provide coupling
-    // between the OS mixer controls and AGC through the |stream_analog_level()|
+    // between the OS mixer controls and AGC through the `stream_analog_level()`
     // functions.
     //
     // It consists of an analog gain prescription for the audio device and a
@@ -61,7 +61,7 @@ class GainControl {
   virtual int set_mode(Mode mode) = 0;
   virtual Mode mode() const = 0;
 
-  // Sets the target peak |level| (or envelope) of the AGC in dBFs (decibels
+  // Sets the target peak `level` (or envelope) of the AGC in dBFs (decibels
   // from digital full-scale). The convention is to use positive values. For
   // instance, passing in a value of 3 corresponds to -3 dBFs, or a target
   // level 3 dB below full-scale. Limited to [0, 31].
@@ -71,7 +71,7 @@ class GainControl {
   virtual int set_target_level_dbfs(int level) = 0;
   virtual int target_level_dbfs() const = 0;
 
-  // Sets the maximum |gain| the digital compression stage may apply, in dB. A
+  // Sets the maximum `gain` the digital compression stage may apply, in dB. A
   // higher number corresponds to greater compression, while a value of 0 will
   // leave the signal uncompressed. Limited to [0, 90].
   virtual int set_compression_gain_db(int gain) = 0;
@@ -83,7 +83,7 @@ class GainControl {
   virtual int enable_limiter(bool enable) = 0;
   virtual bool is_limiter_enabled() const = 0;
 
-  // Sets the |minimum| and |maximum| analog levels of the audio capture device.
+  // Sets the `minimum` and `maximum` analog levels of the audio capture device.
   // Must be set if and only if an analog mode is used. Limited to [0, 65535].
   virtual int set_analog_level_limits(int minimum, int maximum) = 0;
   virtual int analog_level_minimum() const = 0;

modules/audio_processing/agc/legacy/analog_agc.cc

@@ -160,7 +160,7 @@ int WebRtcAgc_AddMic(void* state,
 
   /* apply slowly varying digital gain */
   if (stt->micVol > stt->maxAnalog) {
-    /* |maxLevel| is strictly >= |micVol|, so this condition should be
+    /* `maxLevel` is strictly >= `micVol`, so this condition should be
      * satisfied here, ensuring there is no divide-by-zero. */
     RTC_DCHECK_GT(stt->maxLevel, stt->maxAnalog);
 

modules/audio_processing/agc/legacy/digital_agc.cc

@@ -184,9 +184,9 @@ int32_t WebRtcAgc_CalculateGainTable(int32_t* gainTable,       // Q16
     numFIX -= (int32_t)logApprox * diffGain;       // Q14
 
     // Calculate ratio
-    // Shift |numFIX| as much as possible.
-    // Ensure we avoid wrap-around in |den| as well.
-    if (numFIX > (den >> 8) || -numFIX > (den >> 8)) {  // |den| is Q8.
+    // Shift `numFIX` as much as possible.
+    // Ensure we avoid wrap-around in `den` as well.
+    if (numFIX > (den >> 8) || -numFIX > (den >> 8)) {  // `den` is Q8.
       zeros = WebRtcSpl_NormW32(numFIX);
     } else {
       zeros = WebRtcSpl_NormW32(den) + 8;

modules/audio_processing/agc/loudness_histogram.cc

@@ -114,7 +114,7 @@ void LoudnessHistogram::RemoveOldestEntryAndUpdate() {
 
 void LoudnessHistogram::RemoveTransient() {
   // Don't expect to be here if high-activity region is longer than
-  // |kTransientWidthThreshold| or there has not been any transient.
+  // `kTransientWidthThreshold` or there has not been any transient.
   RTC_DCHECK_LE(len_high_activity_, kTransientWidthThreshold);
   int index =
       (buffer_index_ > 0) ? (buffer_index_ - 1) : len_circular_buffer_ - 1;

modules/audio_processing/agc/loudness_histogram.h

@@ -25,7 +25,7 @@ class LoudnessHistogram {
   static LoudnessHistogram* Create();
 
   // Create a sliding LoudnessHistogram, i.e. the histogram represents the last
-  // |window_size| samples.
+  // `window_size` samples.
   static LoudnessHistogram* Create(int window_size);
   ~LoudnessHistogram();
 
@@ -49,7 +49,7 @@ class LoudnessHistogram {
   LoudnessHistogram();
   explicit LoudnessHistogram(int window);
 
-  // Find the histogram bin associated with the given |rms|.
+  // Find the histogram bin associated with the given `rms`.
   int GetBinIndex(double rms);
 
   void RemoveOldestEntryAndUpdate();
@@ -63,10 +63,10 @@ class LoudnessHistogram {
   // Number of times the histogram is updated
   int num_updates_;
   // Audio content, this should be equal to the sum of the components of
-  // |bin_count_q10_|.
+  // `bin_count_q10_`.
   int64_t audio_content_q10_;
 
-  // LoudnessHistogram of input RMS in Q10 with |kHistSize_| bins. In each
+  // LoudnessHistogram of input RMS in Q10 with `kHistSize_` bins. In each
   // 'Update(),' we increment the associated histogram-bin with the given
   // probability. The increment is implemented in Q10 to avoid rounding errors.
   int64_t bin_count_q10_[kHistSize];