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}
diff --git a/modules/audio_processing/aec_dump/aec_dump_factory.h b/modules/audio_processing/aec_dump/aec_dump_factory.h
index 429a8a5..c902a58 100644
--- a/modules/audio_processing/aec_dump/aec_dump_factory.h
+++ b/modules/audio_processing/aec_dump/aec_dump_factory.h
@@ -26,10 +26,10 @@
class RTC_EXPORT AecDumpFactory {
public:
- // The |worker_queue| may not be null and must outlive the created
+ // The `worker_queue` may not be null and must outlive the created
// AecDump instance. |max_log_size_bytes == -1| means the log size
- // will be unlimited. |handle| may not be null. The AecDump takes
- // responsibility for |handle| and closes it in the destructor. A
+ // will be unlimited. `handle` may not be null. The AecDump takes
+ // responsibility for `handle` and closes it in the destructor. A
// non-null return value indicates that the file has been
// sucessfully opened.
static std::unique_ptr<AecDump> Create(webrtc::FileWrapper file,
diff --git a/modules/audio_processing/aecm/aecm_core.cc b/modules/audio_processing/aecm/aecm_core.cc
index 78c0133..fbc3239 100644
--- a/modules/audio_processing/aecm/aecm_core.cc
+++ b/modules/audio_processing/aecm/aecm_core.cc
@@ -124,7 +124,7 @@
-1140, -998, -856, -713, -571, -428, -285, -142};
-// Moves the pointer to the next entry and inserts |far_spectrum| and
+// Moves the pointer to the next entry and inserts `far_spectrum` and
// corresponding Q-domain in its buffer.
//
// Inputs:
@@ -574,7 +574,7 @@
// Obtain an output frame.
WebRtc_ReadBuffer(aecm->outFrameBuf, (void**)&out_ptr, out, FRAME_LEN);
if (out_ptr != out) {
- // ReadBuffer() hasn't copied to |out| in this case.
+ // ReadBuffer() hasn't copied to `out` in this case.
memcpy(out, out_ptr, FRAME_LEN * sizeof(int16_t));
}
@@ -616,22 +616,22 @@
// ExtractFractionPart(a, zeros)
//
-// returns the fraction part of |a|, with |zeros| number of leading zeros, as an
-// int16_t scaled to Q8. There is no sanity check of |a| in the sense that the
+// returns the fraction part of `a`, with `zeros` number of leading zeros, as an
+// int16_t scaled to Q8. There is no sanity check of `a` in the sense that the
// number of zeros match.
static int16_t ExtractFractionPart(uint32_t a, int zeros) {
return (int16_t)(((a << zeros) & 0x7FFFFFFF) >> 23);
}
-// Calculates and returns the log of |energy| in Q8. The input |energy| is
-// supposed to be in Q(|q_domain|).
+// Calculates and returns the log of `energy` in Q8. The input `energy` is
+// supposed to be in Q(`q_domain`).
static int16_t LogOfEnergyInQ8(uint32_t energy, int q_domain) {
static const int16_t kLogLowValue = PART_LEN_SHIFT << 7;
int16_t log_energy_q8 = kLogLowValue;
if (energy > 0) {
int zeros = WebRtcSpl_NormU32(energy);
int16_t frac = ExtractFractionPart(energy, zeros);
- // log2 of |energy| in Q8.
+ // log2 of `energy` in Q8.
log_energy_q8 += ((31 - zeros) << 8) + frac - (q_domain << 8);
}
return log_energy_q8;
diff --git a/modules/audio_processing/aecm/aecm_core.h b/modules/audio_processing/aecm/aecm_core.h
index aaa74e1..d6d0d8d 100644
--- a/modules/audio_processing/aecm/aecm_core.h
+++ b/modules/audio_processing/aecm/aecm_core.h
@@ -58,7 +58,7 @@
void* delay_estimator;
uint16_t currentDelay;
// Far end history variables
- // TODO(bjornv): Replace |far_history| with ring_buffer.
+ // TODO(bjornv): Replace `far_history` with ring_buffer.
uint16_t far_history[PART_LEN1 * MAX_DELAY];
int far_history_pos;
int far_q_domains[MAX_DELAY];
@@ -271,7 +271,7 @@
////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_UpdateFarHistory()
//
-// Moves the pointer to the next entry and inserts |far_spectrum| and
+// Moves the pointer to the next entry and inserts `far_spectrum` and
// corresponding Q-domain in its buffer.
//
// Inputs:
diff --git a/modules/audio_processing/aecm/aecm_core_c.cc b/modules/audio_processing/aecm/aecm_core_c.cc
index 7b6ca59..d363dd2 100644
--- a/modules/audio_processing/aecm/aecm_core_c.cc
+++ b/modules/audio_processing/aecm/aecm_core_c.cc
@@ -98,7 +98,7 @@
// Track the minimum.
if (aecm->noiseEst[i] < (1 << minTrackShift)) {
// For small values, decrease noiseEst[i] every
- // |kNoiseEstIncCount| block. The regular approach below can not
+ // `kNoiseEstIncCount` block. The regular approach below can not
// go further down due to truncation.
aecm->noiseEstTooHighCtr[i]++;
if (aecm->noiseEstTooHighCtr[i] >= kNoiseEstIncCount) {
@@ -125,7 +125,7 @@
aecm->noiseEst[i] >>= 11;
} else {
// Make incremental increases based on size every
- // |kNoiseEstIncCount| block
+ // `kNoiseEstIncCount` block
aecm->noiseEstTooLowCtr[i]++;
if (aecm->noiseEstTooLowCtr[i] >= kNoiseEstIncCount) {
aecm->noiseEst[i] += (aecm->noiseEst[i] >> 9) + 1;
@@ -181,7 +181,7 @@
// FFT of signal
for (i = 0; i < PART_LEN; i++) {
// Window time domain signal and insert into real part of
- // transformation array |fft|
+ // transformation array `fft`
int16_t scaled_time_signal = time_signal[i] * (1 << time_signal_scaling);
fft[i] = (int16_t)((scaled_time_signal * WebRtcAecm_kSqrtHanning[i]) >> 14);
scaled_time_signal = time_signal[i + PART_LEN] * (1 << time_signal_scaling);
@@ -204,8 +204,8 @@
const int16_t* nearendClean) {
int i, j, outCFFT;
int32_t tmp32no1;
- // Reuse |efw| for the inverse FFT output after transferring
- // the contents to |fft|.
+ // Reuse `efw` for the inverse FFT output after transferring
+ // the contents to `fft`.
int16_t* ifft_out = (int16_t*)efw;
// Synthesis
@@ -312,7 +312,7 @@
} else {
// Approximation for magnitude of complex fft output
// magn = sqrt(real^2 + imag^2)
- // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
+ // magn ~= alpha * max(`imag`,`real`) + beta * min(`imag`,`real`)
//
// The parameters alpha and beta are stored in Q15
@@ -541,7 +541,7 @@
}
zeros16 = WebRtcSpl_NormW16(aecm->nearFilt[i]);
- RTC_DCHECK_GE(zeros16, 0); // |zeros16| is a norm, hence non-negative.
+ RTC_DCHECK_GE(zeros16, 0); // `zeros16` is a norm, hence non-negative.
dfa_clean_q_domain_diff = aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld;
if (zeros16 < dfa_clean_q_domain_diff && aecm->nearFilt[i]) {
tmp16no1 = aecm->nearFilt[i] * (1 << zeros16);
diff --git a/modules/audio_processing/aecm/aecm_core_mips.cc b/modules/audio_processing/aecm/aecm_core_mips.cc
index f2f43e1..828aa6d2f 100644
--- a/modules/audio_processing/aecm/aecm_core_mips.cc
+++ b/modules/audio_processing/aecm/aecm_core_mips.cc
@@ -822,7 +822,7 @@
} else {
// Approximation for magnitude of complex fft output
// magn = sqrt(real^2 + imag^2)
- // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
+ // magn ~= alpha * max(`imag`,`real`) + beta * min(`imag`,`real`)
//
// The parameters alpha and beta are stored in Q15
tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
@@ -1106,7 +1106,7 @@
}
zeros16 = WebRtcSpl_NormW16(aecm->nearFilt[i]);
- RTC_DCHECK_GE(zeros16, 0); // |zeros16| is a norm, hence non-negative.
+ RTC_DCHECK_GE(zeros16, 0); // `zeros16` is a norm, hence non-negative.
dfa_clean_q_domain_diff = aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld;
if (zeros16 < dfa_clean_q_domain_diff && aecm->nearFilt[i]) {
tmp16no1 = aecm->nearFilt[i] << zeros16;
@@ -1411,7 +1411,7 @@
// Track the minimum.
if (tnoise < (1 << minTrackShift)) {
// For small values, decrease noiseEst[i] every
- // |kNoiseEstIncCount| block. The regular approach below can not
+ // `kNoiseEstIncCount` block. The regular approach below can not
// go further down due to truncation.
aecm->noiseEstTooHighCtr[i]++;
if (aecm->noiseEstTooHighCtr[i] >= kNoiseEstIncCount) {
@@ -1442,7 +1442,7 @@
: "hi", "lo");
} else {
// Make incremental increases based on size every
- // |kNoiseEstIncCount| block
+ // `kNoiseEstIncCount` block
aecm->noiseEstTooLowCtr[i]++;
if (aecm->noiseEstTooLowCtr[i] >= kNoiseEstIncCount) {
__asm __volatile(
@@ -1484,7 +1484,7 @@
// Track the minimum.
if (tnoise1 < (1 << minTrackShift)) {
// For small values, decrease noiseEst[i] every
- // |kNoiseEstIncCount| block. The regular approach below can not
+ // `kNoiseEstIncCount` block. The regular approach below can not
// go further down due to truncation.
aecm->noiseEstTooHighCtr[i + 1]++;
if (aecm->noiseEstTooHighCtr[i + 1] >= kNoiseEstIncCount) {
@@ -1515,7 +1515,7 @@
: "hi", "lo");
} else {
// Make incremental increases based on size every
- // |kNoiseEstIncCount| block
+ // `kNoiseEstIncCount` block
aecm->noiseEstTooLowCtr[i + 1]++;
if (aecm->noiseEstTooLowCtr[i + 1] >= kNoiseEstIncCount) {
__asm __volatile(
diff --git a/modules/audio_processing/agc/agc.h b/modules/audio_processing/agc/agc.h
index b9bd5ea..2693d94 100644
--- a/modules/audio_processing/agc/agc.h
+++ b/modules/audio_processing/agc/agc.h
@@ -24,13 +24,13 @@
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();
diff --git a/modules/audio_processing/agc/agc_manager_direct.cc b/modules/audio_processing/agc/agc_manager_direct.cc
index e2a5b99..0cd67ca 100644
--- a/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/modules/audio_processing/agc/agc_manager_direct.cc
@@ -280,7 +280,7 @@
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 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)
diff --git a/modules/audio_processing/agc/agc_manager_direct.h b/modules/audio_processing/agc/agc_manager_direct.h
index d80a255..27569d8 100644
--- a/modules/audio_processing/agc/agc_manager_direct.h
+++ b/modules/audio_processing/agc/agc_manager_direct.h
@@ -112,7 +112,7 @@
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 @@
// 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();
diff --git a/modules/audio_processing/agc/gain_control.h b/modules/audio_processing/agc/gain_control.h
index f8c706b..389b211 100644
--- a/modules/audio_processing/agc/gain_control.h
+++ b/modules/audio_processing/agc/gain_control.h
@@ -20,12 +20,12 @@
// 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 @@
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 @@
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 @@
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 @@
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;
diff --git a/modules/audio_processing/agc/legacy/analog_agc.cc b/modules/audio_processing/agc/legacy/analog_agc.cc
index b53e3f9..e40a3f1 100644
--- a/modules/audio_processing/agc/legacy/analog_agc.cc
+++ b/modules/audio_processing/agc/legacy/analog_agc.cc
@@ -160,7 +160,7 @@
/* 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);
diff --git a/modules/audio_processing/agc/legacy/digital_agc.cc b/modules/audio_processing/agc/legacy/digital_agc.cc
index e0c0766..4cd86ac 100644
--- a/modules/audio_processing/agc/legacy/digital_agc.cc
+++ b/modules/audio_processing/agc/legacy/digital_agc.cc
@@ -184,9 +184,9 @@
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;
diff --git a/modules/audio_processing/agc/loudness_histogram.cc b/modules/audio_processing/agc/loudness_histogram.cc
index 4775ff7..b0a1f53 100644
--- a/modules/audio_processing/agc/loudness_histogram.cc
+++ b/modules/audio_processing/agc/loudness_histogram.cc
@@ -114,7 +114,7 @@
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;
diff --git a/modules/audio_processing/agc/loudness_histogram.h b/modules/audio_processing/agc/loudness_histogram.h
index badd443..51b3871 100644
--- a/modules/audio_processing/agc/loudness_histogram.h
+++ b/modules/audio_processing/agc/loudness_histogram.h
@@ -25,7 +25,7 @@
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 @@
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 @@
// 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];
diff --git a/modules/audio_processing/agc2/biquad_filter.cc b/modules/audio_processing/agc2/biquad_filter.cc
index da8557c..ccb7807 100644
--- a/modules/audio_processing/agc2/biquad_filter.cc
+++ b/modules/audio_processing/agc2/biquad_filter.cc
@@ -15,7 +15,7 @@
namespace webrtc {
// Transposed direct form I implementation of a bi-quad filter applied to an
-// input signal |x| to produce an output signal |y|.
+// input signal `x` to produce an output signal `y`.
void BiQuadFilter::Process(rtc::ArrayView<const float> x,
rtc::ArrayView<float> y) {
for (size_t k = 0; k < x.size(); ++k) {
diff --git a/modules/audio_processing/agc2/biquad_filter_unittest.cc b/modules/audio_processing/agc2/biquad_filter_unittest.cc
index cd9a272..55ca1a5 100644
--- a/modules/audio_processing/agc2/biquad_filter_unittest.cc
+++ b/modules/audio_processing/agc2/biquad_filter_unittest.cc
@@ -64,7 +64,7 @@
rtc::ArrayView<const float> computed,
const float tolerance) {
// The relative error is undefined when the expected value is 0.
- // When that happens, check the absolute error instead. |safe_den| is used
+ // When that happens, check the absolute error instead. `safe_den` is used
// below to implement such logic.
auto safe_den = [](float x) { return (x == 0.f) ? 1.f : std::fabs(x); };
ASSERT_EQ(expected.size(), computed.size());
diff --git a/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc b/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc
index bc92613..221b499 100644
--- a/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc
+++ b/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc
@@ -105,7 +105,7 @@
const auto interval = q.top();
q.pop();
- // Split |interval| and enqueue.
+ // Split `interval` and enqueue.
double x_split = (interval.x0 + interval.x1) / 2.0;
q.emplace(interval.x0, x_split,
LimiterUnderApproximationNegativeError(limiter, interval.x0,
@@ -135,7 +135,7 @@
void PrecomputeKneeApproxParams(const LimiterDbGainCurve* limiter,
test::InterpolatedParameters* parameters) {
static_assert(kInterpolatedGainCurveKneePoints > 2, "");
- // Get |kInterpolatedGainCurveKneePoints| - 1 equally spaced points.
+ // Get `kInterpolatedGainCurveKneePoints` - 1 equally spaced points.
const std::vector<double> points = test::LinSpace(
limiter->knee_start_linear(), limiter->limiter_start_linear(),
kInterpolatedGainCurveKneePoints - 1);
diff --git a/modules/audio_processing/agc2/compute_interpolated_gain_curve.h b/modules/audio_processing/agc2/compute_interpolated_gain_curve.h
index 5f52441..08b676f 100644
--- a/modules/audio_processing/agc2/compute_interpolated_gain_curve.h
+++ b/modules/audio_processing/agc2/compute_interpolated_gain_curve.h
@@ -29,8 +29,8 @@
// Knee and beyond-knee regions approximation parameters.
// The gain curve is approximated as a piece-wise linear function.
-// |approx_params_x_| are the boundaries between adjacent linear pieces,
-// |approx_params_m_| and |approx_params_q_| are the slope and the y-intercept
+// `approx_params_x_` are the boundaries between adjacent linear pieces,
+// `approx_params_m_` and `approx_params_q_` are the slope and the y-intercept
// values of each piece.
struct InterpolatedParameters {
std::array<float, kInterpolatedGainCurveTotalPoints>
diff --git a/modules/audio_processing/agc2/fixed_digital_level_estimator.cc b/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
index 3e9bb2e..eb8a64a 100644
--- a/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
+++ b/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
@@ -26,7 +26,7 @@
constexpr float kAttackFilterConstant = 0.f;
// This is computed from kDecayMs by
// 10 ** (-1/20 * subframe_duration / kDecayMs).
-// |subframe_duration| is |kFrameDurationMs / kSubFramesInFrame|.
+// `subframe_duration` is |kFrameDurationMs / kSubFramesInFrame|.
// kDecayMs is defined in agc2_testing_common.h
constexpr float kDecayFilterConstant = 0.9998848773724686f;
diff --git a/modules/audio_processing/agc2/interpolated_gain_curve.cc b/modules/audio_processing/agc2/interpolated_gain_curve.cc
index 3dd5010..ac7fbec 100644
--- a/modules/audio_processing/agc2/interpolated_gain_curve.cc
+++ b/modules/audio_processing/agc2/interpolated_gain_curve.cc
@@ -151,11 +151,11 @@
}
// Looks up a gain to apply given a non-negative input level.
-// The cost of this operation depends on the region in which |input_level|
+// The cost of this operation depends on the region in which `input_level`
// falls.
// For the identity and the saturation regions the cost is O(1).
// For the other regions, namely knee and limiter, the cost is
-// O(2 + log2(|LightkInterpolatedGainCurveTotalPoints|), plus O(1) for the
+// O(2 + log2(`LightkInterpolatedGainCurveTotalPoints`), plus O(1) for the
// linear interpolation (one product and one sum).
float InterpolatedGainCurve::LookUpGainToApply(float input_level) const {
UpdateStats(input_level);
diff --git a/modules/audio_processing/agc2/limiter.h b/modules/audio_processing/agc2/limiter.h
index df7b540..f8eec3d 100644
--- a/modules/audio_processing/agc2/limiter.h
+++ b/modules/audio_processing/agc2/limiter.h
@@ -31,7 +31,7 @@
Limiter& operator=(const Limiter& limiter) = delete;
~Limiter();
- // Applies limiter and hard-clipping to |signal|.
+ // Applies limiter and hard-clipping to `signal`.
void Process(AudioFrameView<float> signal);
InterpolatedGainCurve::Stats GetGainCurveStats() const;
diff --git a/modules/audio_processing/agc2/limiter_db_gain_curve.cc b/modules/audio_processing/agc2/limiter_db_gain_curve.cc
index d55ed5d..d47c0b2 100644
--- a/modules/audio_processing/agc2/limiter_db_gain_curve.cc
+++ b/modules/audio_processing/agc2/limiter_db_gain_curve.cc
@@ -105,7 +105,7 @@
input_level_linear;
}
-// Computes the first derivative of GetGainLinear() in |x|.
+// Computes the first derivative of GetGainLinear() in `x`.
double LimiterDbGainCurve::GetGainFirstDerivativeLinear(double x) const {
// Beyond-knee region only.
RTC_CHECK_GE(x, limiter_start_linear_ - 1e-7 * kMaxAbsFloatS16Value);
diff --git a/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc b/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc
index 431c01f..3ddeec8 100644
--- a/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc
+++ b/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc
@@ -40,7 +40,7 @@
// [ y_{m-1} ]
// x and y are sub-array of equal length; x is never moved, whereas y slides.
// The cross-correlation between y_0 and x corresponds to the auto-correlation
-// for the maximum pitch period. Hence, the first value in |auto_corr| has an
+// for the maximum pitch period. Hence, the first value in `auto_corr` has an
// inverted lag equal to 0 that corresponds to a lag equal to the maximum
// pitch period.
void AutoCorrelationCalculator::ComputeOnPitchBuffer(
diff --git a/modules/audio_processing/agc2/rnn_vad/auto_correlation.h b/modules/audio_processing/agc2/rnn_vad/auto_correlation.h
index d58558c..1ae5054 100644
--- a/modules/audio_processing/agc2/rnn_vad/auto_correlation.h
+++ b/modules/audio_processing/agc2/rnn_vad/auto_correlation.h
@@ -31,7 +31,7 @@
~AutoCorrelationCalculator();
// Computes the auto-correlation coefficients for a target pitch interval.
- // |auto_corr| indexes are inverted lags.
+ // `auto_corr` indexes are inverted lags.
void ComputeOnPitchBuffer(
rtc::ArrayView<const float, kBufSize12kHz> pitch_buf,
rtc::ArrayView<float, kNumLags12kHz> auto_corr);
diff --git a/modules/audio_processing/agc2/rnn_vad/common.h b/modules/audio_processing/agc2/rnn_vad/common.h
index be5a2d5..c099373 100644
--- a/modules/audio_processing/agc2/rnn_vad/common.h
+++ b/modules/audio_processing/agc2/rnn_vad/common.h
@@ -52,8 +52,8 @@
constexpr int kInitialMinPitch12kHz = kInitialMinPitch24kHz / 2;
constexpr int kMaxPitch12kHz = kMaxPitch24kHz / 2;
static_assert(kMaxPitch12kHz > kInitialMinPitch12kHz, "");
-// The inverted lags for the pitch interval [|kInitialMinPitch12kHz|,
-// |kMaxPitch12kHz|] are in the range [0, |kNumLags12kHz|].
+// The inverted lags for the pitch interval [`kInitialMinPitch12kHz`,
+// `kMaxPitch12kHz`] are in the range [0, `kNumLags12kHz`].
constexpr int kNumLags12kHz = kMaxPitch12kHz - kInitialMinPitch12kHz;
// 48 kHz constants.
diff --git a/modules/audio_processing/agc2/rnn_vad/features_extraction.cc b/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
index f86eba7..5c276c8 100644
--- a/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
+++ b/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
@@ -55,10 +55,10 @@
if (use_high_pass_filter_) {
std::array<float, kFrameSize10ms24kHz> samples_filtered;
hpf_.Process(samples, samples_filtered);
- // Feed buffer with the pre-processed version of |samples|.
+ // Feed buffer with the pre-processed version of `samples`.
pitch_buf_24kHz_.Push(samples_filtered);
} else {
- // Feed buffer with |samples|.
+ // Feed buffer with `samples`.
pitch_buf_24kHz_.Push(samples);
}
// Extract the LP residual.
diff --git a/modules/audio_processing/agc2/rnn_vad/features_extraction.h b/modules/audio_processing/agc2/rnn_vad/features_extraction.h
index f4cea7a..d47a85b 100644
--- a/modules/audio_processing/agc2/rnn_vad/features_extraction.h
+++ b/modules/audio_processing/agc2/rnn_vad/features_extraction.h
@@ -33,7 +33,7 @@
void Reset();
// Analyzes the samples, computes the feature vector and returns true if
// silence is detected (false if not). When silence is detected,
- // |feature_vector| is partially written and therefore must not be used to
+ // `feature_vector` is partially written and therefore must not be used to
// feed the VAD RNN.
bool CheckSilenceComputeFeatures(
rtc::ArrayView<const float, kFrameSize10ms24kHz> samples,
diff --git a/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc b/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
index 98da39e..96f956a 100644
--- a/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
@@ -29,7 +29,7 @@
}
// Number of 10 ms frames required to fill a pitch buffer having size
-// |kBufSize24kHz|.
+// `kBufSize24kHz`.
constexpr int kNumTestDataFrames = ceil(kBufSize24kHz, kFrameSize10ms24kHz);
// Number of samples for the test data.
constexpr int kNumTestDataSize = kNumTestDataFrames * kFrameSize10ms24kHz;
@@ -47,8 +47,8 @@
}
}
-// Feeds |features_extractor| with |samples| splitting it in 10 ms frames.
-// For every frame, the output is written into |feature_vector|. Returns true
+// Feeds `features_extractor` with `samples` splitting it in 10 ms frames.
+// For every frame, the output is written into `feature_vector`. Returns true
// if silence is detected in the last frame.
bool FeedTestData(FeaturesExtractor& features_extractor,
rtc::ArrayView<const float> samples,
diff --git a/modules/audio_processing/agc2/rnn_vad/lp_residual.cc b/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
index c553aa2..484bfba 100644
--- a/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
+++ b/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
@@ -22,9 +22,9 @@
namespace rnn_vad {
namespace {
-// Computes auto-correlation coefficients for |x| and writes them in
-// |auto_corr|. The lag values are in {0, ..., max_lag - 1}, where max_lag
-// equals the size of |auto_corr|.
+// Computes auto-correlation coefficients for `x` and writes them in
+// `auto_corr`. The lag values are in {0, ..., max_lag - 1}, where max_lag
+// equals the size of `auto_corr`.
void ComputeAutoCorrelation(
rtc::ArrayView<const float> x,
rtc::ArrayView<float, kNumLpcCoefficients> auto_corr) {
diff --git a/modules/audio_processing/agc2/rnn_vad/lp_residual.h b/modules/audio_processing/agc2/rnn_vad/lp_residual.h
index 380d9f6..d04c536 100644
--- a/modules/audio_processing/agc2/rnn_vad/lp_residual.h
+++ b/modules/audio_processing/agc2/rnn_vad/lp_residual.h
@@ -21,14 +21,14 @@
// Linear predictive coding (LPC) inverse filter length.
constexpr int kNumLpcCoefficients = 5;
-// Given a frame |x|, computes a post-processed version of LPC coefficients
+// Given a frame `x`, computes a post-processed version of LPC coefficients
// tailored for pitch estimation.
void ComputeAndPostProcessLpcCoefficients(
rtc::ArrayView<const float> x,
rtc::ArrayView<float, kNumLpcCoefficients> lpc_coeffs);
-// Computes the LP residual for the input frame |x| and the LPC coefficients
-// |lpc_coeffs|. |y| and |x| can point to the same array for in-place
+// Computes the LP residual for the input frame `x` and the LPC coefficients
+// `lpc_coeffs`. `y` and `x` can point to the same array for in-place
// computation.
void ComputeLpResidual(
rtc::ArrayView<const float, kNumLpcCoefficients> lpc_coeffs,
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search.cc b/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
index 77a1188..419620f 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
@@ -44,7 +44,7 @@
CandidatePitchPeriods pitch_periods = ComputePitchPeriod12kHz(
pitch_buffer_12kHz_view, auto_correlation_12kHz_view, cpu_features_);
// The refinement is done using the pitch buffer that contains 24 kHz samples.
- // Therefore, adapt the inverted lags in |pitch_candidates_inv_lags| from 12
+ // Therefore, adapt the inverted lags in `pitch_candidates_inv_lags` from 12
// to 24 kHz.
pitch_periods.best *= 2;
pitch_periods.second_best *= 2;
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
index 0b8a77e..4000e33 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
@@ -54,18 +54,18 @@
float next_auto_correlation) {
if ((next_auto_correlation - prev_auto_correlation) >
0.7f * (curr_auto_correlation - prev_auto_correlation)) {
- return 1; // |next_auto_correlation| is the largest auto-correlation
+ return 1; // `next_auto_correlation` is the largest auto-correlation
// coefficient.
} else if ((prev_auto_correlation - next_auto_correlation) >
0.7f * (curr_auto_correlation - next_auto_correlation)) {
- return -1; // |prev_auto_correlation| is the largest auto-correlation
+ return -1; // `prev_auto_correlation` is the largest auto-correlation
// coefficient.
}
return 0;
}
-// Refines a pitch period |lag| encoded as lag with pseudo-interpolation. The
-// output sample rate is twice as that of |lag|.
+// Refines a pitch period `lag` encoded as lag with pseudo-interpolation. The
+// output sample rate is twice as that of `lag`.
int PitchPseudoInterpolationLagPitchBuf(
int lag,
rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
@@ -217,8 +217,8 @@
auto_correlation[best_inverted_lag + 1],
auto_correlation[best_inverted_lag],
auto_correlation[best_inverted_lag - 1]);
- // TODO(bugs.webrtc.org/9076): When retraining, check if |offset| below should
- // be subtracted since |inverted_lag| is an inverted lag but offset is a lag.
+ // TODO(bugs.webrtc.org/9076): When retraining, check if `offset` below should
+ // be subtracted since `inverted_lag` is an inverted lag but offset is a lag.
return 2 * best_inverted_lag + offset;
}
@@ -359,7 +359,7 @@
}
}
}
- // Update |squared_energy_y| for the next inverted lag.
+ // Update `squared_energy_y` for the next inverted lag.
const float y_old = pitch_buffer[inverted_lag];
const float y_new = pitch_buffer[inverted_lag + kFrameSize20ms12kHz];
denominator -= y_old * y_old;
@@ -458,8 +458,8 @@
initial_pitch.period, /*multiplier=*/1, period_divisor);
RTC_DCHECK_GE(alternative_pitch.period, kMinPitch24kHz);
// When looking at |alternative_pitch.period|, we also look at one of its
- // sub-harmonics. |kSubHarmonicMultipliers| is used to know where to look.
- // |period_divisor| == 2 is a special case since |dual_alternative_period|
+ // sub-harmonics. `kSubHarmonicMultipliers` is used to know where to look.
+ // `period_divisor` == 2 is a special case since `dual_alternative_period`
// might be greater than the maximum pitch period.
int dual_alternative_period = GetAlternativePitchPeriod(
initial_pitch.period, kSubHarmonicMultipliers[period_divisor - 2],
@@ -473,7 +473,7 @@
"coincide.";
// Compute an auto-correlation score for the primary pitch candidate
// |alternative_pitch.period| by also looking at its possible sub-harmonic
- // |dual_alternative_period|.
+ // `dual_alternative_period`.
const float xy_primary_period = ComputeAutoCorrelation(
kMaxPitch24kHz - alternative_pitch.period, pitch_buffer, vector_math);
// TODO(webrtc:10480): Copy `xy_primary_period` if the secondary period is
diff --git a/modules/audio_processing/agc2/rnn_vad/ring_buffer.h b/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
index f0270af..a6f7fdd 100644
--- a/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
+++ b/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
@@ -35,7 +35,7 @@
~RingBuffer() = default;
// Set the ring buffer values to zero.
void Reset() { buffer_.fill(0); }
- // Replace the least recently pushed array in the buffer with |new_values|.
+ // Replace the least recently pushed array in the buffer with `new_values`.
void Push(rtc::ArrayView<const T, S> new_values) {
std::memcpy(buffer_.data() + S * tail_, new_values.data(), S * sizeof(T));
tail_ += 1;
@@ -43,7 +43,7 @@
tail_ = 0;
}
// Return an array view onto the array with a given delay. A view on the last
- // and least recently push array is returned when |delay| is 0 and N - 1
+ // and least recently push array is returned when `delay` is 0 and N - 1
// respectively.
rtc::ArrayView<const T, S> GetArrayView(int delay) const {
RTC_DCHECK_LE(0, delay);
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc b/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
index ecbb198..91501fb 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
@@ -32,7 +32,7 @@
// TODO(bugs.chromium.org/10480): Hard-code optimized layout and remove this
// function to improve setup time.
-// Casts and scales |weights| and re-arranges the layout.
+// Casts and scales `weights` and re-arranges the layout.
std::vector<float> PreprocessWeights(rtc::ArrayView<const int8_t> weights,
int output_size) {
if (output_size == 1) {
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc b/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc
index 482016e..ef37410 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc
@@ -24,7 +24,7 @@
std::vector<float> PreprocessGruTensor(rtc::ArrayView<const int8_t> tensor_src,
int output_size) {
// Transpose, cast and scale.
- // |n| is the size of the first dimension of the 3-dim tensor |weights|.
+ // `n` is the size of the first dimension of the 3-dim tensor `weights`.
const int n = rtc::CheckedDivExact(rtc::dchecked_cast<int>(tensor_src.size()),
output_size * kNumGruGates);
const int stride_src = kNumGruGates * output_size;
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 989b235..f33cd14 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
@@ -49,7 +49,7 @@
// constant below to true in order to write new expected output binary files.
constexpr bool kWriteComputedOutputToFile = false;
-// Avoids that one forgets to set |kWriteComputedOutputToFile| back to false
+// Avoids that one forgets to set `kWriteComputedOutputToFile` back to false
// when the expected output files are re-exported.
TEST(RnnVadTest, CheckWriteComputedOutputIsFalse) {
ASSERT_FALSE(kWriteComputedOutputToFile)
diff --git a/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc b/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc
index f577571..af00583 100644
--- a/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc
@@ -50,7 +50,7 @@
for (int i = 0; i < N; ++i)
chunk[i] = static_cast<T>(i + 1);
seq_buf.Push(chunk);
- // With the next Push(), |last| will be moved left by N positions.
+ // With the next Push(), `last` will be moved left by N positions.
const T last = chunk[N - 1];
for (int i = 0; i < N; ++i)
chunk[i] = static_cast<T>(last + i + 1);
diff --git a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
index 91c0086..a10b0f7 100644
--- a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
+++ b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
@@ -23,7 +23,7 @@
// Weights for each FFT coefficient for each Opus band (Nyquist frequency
// excluded). The size of each band is specified in
-// |kOpusScaleNumBins24kHz20ms|.
+// `kOpusScaleNumBins24kHz20ms`.
constexpr std::array<float, kFrameSize20ms24kHz / 2> kOpusBandWeights24kHz20ms =
{{
0.f, 0.25f, 0.5f, 0.75f, // Band 0
diff --git a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
index aa7b1c6..f4b293a 100644
--- a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
+++ b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
@@ -50,8 +50,8 @@
~SpectralCorrelator();
// Computes the band-wise spectral auto-correlations.
- // |x| must:
- // - have size equal to |kFrameSize20ms24kHz|;
+ // `x` must:
+ // - have size equal to `kFrameSize20ms24kHz`;
// - be encoded as vectors of interleaved real-complex FFT coefficients
// where x[1] = y[1] = 0 (the Nyquist frequency coefficient is omitted).
void ComputeAutoCorrelation(
@@ -59,8 +59,8 @@
rtc::ArrayView<float, kOpusBands24kHz> auto_corr) const;
// Computes the band-wise spectral cross-correlations.
- // |x| and |y| must:
- // - have size equal to |kFrameSize20ms24kHz|;
+ // `x` and `y` must:
+ // - have size equal to `kFrameSize20ms24kHz`;
// - be encoded as vectors of interleaved real-complex FFT coefficients where
// x[1] = y[1] = 0 (the Nyquist frequency coefficient is omitted).
void ComputeCrossCorrelation(
@@ -82,12 +82,12 @@
// TODO(bugs.webrtc.org/10480): Move to anonymous namespace in
// spectral_features.cc. Creates a DCT table for arrays having size equal to
-// |kNumBands|. Declared here for unit testing.
+// `kNumBands`. Declared here for unit testing.
std::array<float, kNumBands * kNumBands> ComputeDctTable();
// TODO(bugs.webrtc.org/10480): Move to anonymous namespace in
-// spectral_features.cc. Computes DCT for |in| given a pre-computed DCT table.
-// In-place computation is not allowed and |out| can be smaller than |in| in
+// spectral_features.cc. Computes DCT for `in` given a pre-computed DCT table.
+// In-place computation is not allowed and `out` can be smaller than `in` in
// order to only compute the first DCT coefficients. Declared here for unit
// testing.
void ComputeDct(rtc::ArrayView<const float> in,
diff --git a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc
index 11a44a5..ece4eb5 100644
--- a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc
@@ -28,7 +28,7 @@
namespace rnn_vad {
namespace {
-// Generates the values for the array named |kOpusBandWeights24kHz20ms| in the
+// Generates the values for the array named `kOpusBandWeights24kHz20ms` in the
// anonymous namespace of the .cc file, which is the array of FFT coefficient
// weights for the Opus scale triangular filters.
std::vector<float> ComputeTriangularFiltersWeights() {
@@ -66,7 +66,7 @@
// Checks that the computed triangular filters weights for the Opus scale are
// monotonic withing each Opus band. This test should only be enabled when
-// ComputeTriangularFiltersWeights() is changed and |kOpusBandWeights24kHz20ms|
+// ComputeTriangularFiltersWeights() is changed and `kOpusBandWeights24kHz20ms`
// is updated accordingly.
TEST(RnnVadTest, DISABLED_TestOpusScaleWeights) {
auto weights = ComputeTriangularFiltersWeights();
diff --git a/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h b/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h
index dd3b62a..d186479 100644
--- a/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h
+++ b/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h
@@ -46,9 +46,9 @@
buf_.fill(0);
}
// Pushes the results from the comparison between the most recent item and
- // those that are still in the ring buffer. The first element in |values| must
+ // those that are still in the ring buffer. The first element in `values` must
// correspond to the comparison between the most recent item and the second
- // most recent one in the ring buffer, whereas the last element in |values|
+ // most recent one in the ring buffer, whereas the last element in `values`
// must correspond to the comparison between the most recent item and the
// oldest one in the ring buffer.
void Push(rtc::ArrayView<T, S - 1> values) {
@@ -64,7 +64,7 @@
}
}
// Reads the value that corresponds to comparison of two items in the ring
- // buffer having delay |delay1| and |delay2|. The two arguments must not be
+ // buffer having delay `delay1` and `delay2`. The two arguments must not be
// equal and both must be in {0, ..., S - 1}.
T GetValue(int delay1, int delay2) const {
int row = S - 1 - delay1;
diff --git a/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc b/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
index 6f61c87..1509ca5 100644
--- a/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
@@ -58,17 +58,17 @@
SCOPED_TRACE(t);
const int t_removed = ring_buf.GetArrayView(kRingBufSize - 1)[0];
ring_buf.Push({&t, 1});
- // The head of the ring buffer is |t|.
+ // The head of the ring buffer is `t`.
ASSERT_EQ(t, ring_buf.GetArrayView(0)[0]);
- // Create the comparisons between |t| and the older elements in the ring
+ // Create the comparisons between `t` and the older elements in the ring
// buffer.
std::array<PairType, kRingBufSize - 1> new_comparions;
for (int i = 0; i < kRingBufSize - 1; ++i) {
- // Start comparing |t| to the second newest element in the ring buffer.
+ // Start comparing `t` to the second newest element in the ring buffer.
const int delay = i + 1;
const auto t_prev = ring_buf.GetArrayView(delay)[0];
ASSERT_EQ(std::max(0, t - delay), t_prev);
- // Compare the last element |t| with |t_prev|.
+ // Compare the last element `t` with `t_prev`.
new_comparions[i].first = t_prev;
new_comparions[i].second = t;
}
diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h
index 3eecf0d..ab0af44 100644
--- a/modules/audio_processing/audio_buffer.h
+++ b/modules/audio_processing/audio_buffer.h
@@ -71,8 +71,8 @@
// Usage:
// channels()[channel][sample].
// Where:
- // 0 <= channel < |buffer_num_channels_|
- // 0 <= sample < |buffer_num_frames_|
+ // 0 <= channel < `buffer_num_channels_`
+ // 0 <= sample < `buffer_num_frames_`
float* const* channels() { return data_->channels(); }
const float* const* channels_const() const { return data_->channels(); }
@@ -80,9 +80,9 @@
// Usage:
// split_bands(channel)[band][sample].
// Where:
- // 0 <= channel < |buffer_num_channels_|
- // 0 <= band < |num_bands_|
- // 0 <= sample < |num_split_frames_|
+ // 0 <= channel < `buffer_num_channels_`
+ // 0 <= band < `num_bands_`
+ // 0 <= sample < `num_split_frames_`
const float* const* split_bands_const(size_t channel) const {
return split_data_.get() ? split_data_->bands(channel)
: data_->bands(channel);
@@ -96,9 +96,9 @@
// Usage:
// split_channels(band)[channel][sample].
// Where:
- // 0 <= band < |num_bands_|
- // 0 <= channel < |buffer_num_channels_|
- // 0 <= sample < |num_split_frames_|
+ // 0 <= band < `num_bands_`
+ // 0 <= channel < `buffer_num_channels_`
+ // 0 <= sample < `num_split_frames_`
const float* const* split_channels_const(Band band) const {
if (split_data_.get()) {
return split_data_->channels(band);
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 4a19855..5acf693 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -1325,7 +1325,7 @@
capture_.key_pressed);
}
- // Experimental APM sub-module that analyzes |capture_buffer|.
+ // Experimental APM sub-module that analyzes `capture_buffer`.
if (submodules_.capture_analyzer) {
submodules_.capture_analyzer->Analyze(capture_buffer);
}
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index 686e417..2c22536 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -169,7 +169,7 @@
const ApmSubmoduleCreationOverrides& overrides);
// Class providing thread-safe message pipe functionality for
- // |runtime_settings_|.
+ // `runtime_settings_`.
class RuntimeSettingEnqueuer {
public:
explicit RuntimeSettingEnqueuer(
@@ -320,8 +320,8 @@
// Collects configuration settings from public and private
// submodules to be saved as an audioproc::Config message on the
- // AecDump if it is attached. If not |forced|, only writes the current
- // config if it is different from the last saved one; if |forced|,
+ // AecDump if it is attached. If not `forced`, only writes the current
+ // config if it is different from the last saved one; if `forced`,
// writes the config regardless of the last saved.
void WriteAecDumpConfigMessage(bool forced)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 4d30a34..faeca79 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -321,10 +321,10 @@
// Reads a 10 ms chunk of int16 interleaved audio from the given (assumed
// stereo) file, converts to deinterleaved float (optionally downmixing) and
-// returns the result in |cb|. Returns false if the file ended (or on error) and
+// returns the result in `cb`. Returns false if the file ended (or on error) and
// true otherwise.
//
-// |int_data| and |float_data| are just temporary space that must be
+// `int_data` and `float_data` are just temporary space that must be
// sufficiently large to hold the 10 ms chunk.
bool ReadChunk(FILE* file,
int16_t* int_data,
@@ -596,7 +596,7 @@
int system_delay_ms,
int delay_min,
int delay_max) {
- // The |revframe_| and |frame_| should include the proper frame information,
+ // The `revframe_` and `frame_` should include the proper frame information,
// hence can be used for extracting information.
Int16FrameData tmp_frame;
std::queue<Int16FrameData*> frame_queue;
@@ -606,7 +606,7 @@
SetFrameTo(&tmp_frame, 0);
EXPECT_EQ(apm_->kNoError, apm_->Initialize());
- // Initialize the |frame_queue| with empty frames.
+ // Initialize the `frame_queue` with empty frames.
int frame_delay = delay_ms / 10;
while (frame_delay < 0) {
Int16FrameData* frame = new Int16FrameData();
@@ -1884,7 +1884,7 @@
if (!absl::GetFlag(FLAGS_write_apm_ref_data)) {
const int kIntNear = 1;
// When running the test on a N7 we get a {2, 6} difference of
- // |has_voice_count| and |max_output_average| is up to 18 higher.
+ // `has_voice_count` and `max_output_average` is up to 18 higher.
// All numbers being consistently higher on N7 compare to ref_data.
// TODO(bjornv): If we start getting more of these offsets on Android we
// should consider a different approach. Either using one slack for all,
@@ -2058,7 +2058,7 @@
static void TearDownTestSuite() { ClearTempFiles(); }
// Runs a process pass on files with the given parameters and dumps the output
- // to a file specified with |output_file_prefix|. Both forward and reverse
+ // to a file specified with `output_file_prefix`. Both forward and reverse
// output streams are dumped.
static void ProcessFormat(int input_rate,
int output_rate,
@@ -2277,7 +2277,7 @@
out_ptr = cmp_data.get();
}
- // Update the |sq_error| and |variance| accumulators with the highest
+ // Update the `sq_error` and `variance` accumulators with the highest
// SNR of reference vs output.
UpdateBestSNR(ref_data.get(), out_ptr, ref_length, expected_delay,
&variance, &sq_error);
diff --git a/modules/audio_processing/echo_control_mobile_impl.h b/modules/audio_processing/echo_control_mobile_impl.h
index 23f3c06..f7f2626 100644
--- a/modules/audio_processing/echo_control_mobile_impl.h
+++ b/modules/audio_processing/echo_control_mobile_impl.h
@@ -42,7 +42,7 @@
kLoudSpeakerphone
};
- // Sets echo control appropriate for the audio routing |mode| on the device.
+ // Sets echo control appropriate for the audio routing `mode` on the device.
// It can and should be updated during a call if the audio routing changes.
int set_routing_mode(RoutingMode mode);
RoutingMode routing_mode() const;
diff --git a/modules/audio_processing/gain_controller2_unittest.cc b/modules/audio_processing/gain_controller2_unittest.cc
index 85c08bb..b1ab00e 100644
--- a/modules/audio_processing/gain_controller2_unittest.cc
+++ b/modules/audio_processing/gain_controller2_unittest.cc
@@ -27,7 +27,7 @@
namespace {
void SetAudioBufferSamples(float value, AudioBuffer* ab) {
- // Sets all the samples in |ab| to |value|.
+ // Sets all the samples in `ab` to `value`.
for (size_t k = 0; k < ab->num_channels(); ++k) {
std::fill(ab->channels()[k], ab->channels()[k] + ab->num_frames(), value);
}
diff --git a/modules/audio_processing/include/audio_frame_proxies.h b/modules/audio_processing/include/audio_frame_proxies.h
index 2d0f5b5..5dd111c 100644
--- a/modules/audio_processing/include/audio_frame_proxies.h
+++ b/modules/audio_processing/include/audio_frame_proxies.h
@@ -16,21 +16,21 @@
class AudioFrame;
class AudioProcessing;
-// Processes a 10 ms |frame| of the primary audio stream using the provided
+// Processes a 10 ms `frame` of the primary audio stream using the provided
// AudioProcessing object. On the client-side, this is the near-end (or
-// captured) audio. The |sample_rate_hz_|, |num_channels_|, and
-// |samples_per_channel_| members of |frame| must be valid. If changed from the
+// captured) audio. The `sample_rate_hz_`, `num_channels_`, and
+// `samples_per_channel_` members of `frame` must be valid. If changed from the
// previous call to this function, it will trigger an initialization of the
// provided AudioProcessing object.
// The function returns any error codes passed from the AudioProcessing
// ProcessStream method.
int ProcessAudioFrame(AudioProcessing* ap, AudioFrame* frame);
-// Processes a 10 ms |frame| of the reverse direction audio stream using the
+// Processes a 10 ms `frame` of the reverse direction audio stream using the
// provided AudioProcessing object. The frame may be modified. On the
// client-side, this is the far-end (or to be rendered) audio. The
-// |sample_rate_hz_|, |num_channels_|, and |samples_per_channel_| members of
-// |frame| must be valid. If changed from the previous call to this function, it
+// `sample_rate_hz_`, `num_channels_`, and `samples_per_channel_` members of
+// `frame` must be valid. If changed from the previous call to this function, it
// will trigger an initialization of the provided AudioProcessing object.
// The function returns any error codes passed from the AudioProcessing
// ProcessReverseStream method.
diff --git a/modules/audio_processing/include/audio_frame_view.h b/modules/audio_processing/include/audio_frame_view.h
index ab5779a..9786cd9 100644
--- a/modules/audio_processing/include/audio_frame_view.h
+++ b/modules/audio_processing/include/audio_frame_view.h
@@ -19,8 +19,8 @@
template <class T>
class AudioFrameView {
public:
- // |num_channels| and |channel_size| describe the T**
- // |audio_samples|. |audio_samples| is assumed to point to a
+ // `num_channels` and `channel_size` describe the T**
+ // `audio_samples`. `audio_samples` is assumed to point to a
// two-dimensional |num_channels * channel_size| array of floats.
AudioFrameView(T* const* audio_samples,
size_t num_channels,
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 64b1b5d..047776b 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -53,7 +53,7 @@
class CustomProcessing;
// Use to enable experimental gain control (AGC). At startup the experimental
-// AGC moves the microphone volume up to |startup_min_volume| if the current
+// AGC moves the microphone volume up to `startup_min_volume` if the current
// microphone volume is set too low. The value is clamped to its operating range
// [12, 255]. Here, 255 maps to 100%.
//
@@ -99,8 +99,8 @@
//
// APM operates on two audio streams on a frame-by-frame basis. Frames of the
// primary stream, on which all processing is applied, are passed to
-// |ProcessStream()|. Frames of the reverse direction stream are passed to
-// |ProcessReverseStream()|. On the client-side, this will typically be the
+// `ProcessStream()`. Frames of the reverse direction stream are passed to
+// `ProcessReverseStream()`. On the client-side, this will typically be the
// near-end (capture) and far-end (render) streams, respectively. APM should be
// placed in the signal chain as close to the audio hardware abstraction layer
// (HAL) as possible.
@@ -264,7 +264,7 @@
bool enabled = false;
} transient_suppression;
- // Enables reporting of |voice_detected| in webrtc::AudioProcessingStats.
+ // Enables reporting of `voice_detected` in webrtc::AudioProcessingStats.
struct VoiceDetection {
bool enabled = false;
} voice_detection;
@@ -377,7 +377,7 @@
// Enables the next generation AGC functionality. This feature replaces the
// standard methods of gain control in the previous AGC. Enabling this
// submodule enables an adaptive digital AGC followed by a limiter. By
- // setting |fixed_gain_db|, the limiter can be turned into a compressor that
+ // setting `fixed_gain_db`, the limiter can be turned into a compressor that
// first applies a fixed gain. The adaptive digital AGC can be turned off by
// setting |adaptive_digital_mode=false|.
struct RTC_EXPORT GainController2 {
@@ -425,7 +425,7 @@
bool enabled = true;
} residual_echo_detector;
- // Enables reporting of |output_rms_dbfs| in webrtc::AudioProcessingStats.
+ // Enables reporting of `output_rms_dbfs` in webrtc::AudioProcessingStats.
struct LevelEstimation {
bool enabled = false;
} level_estimation;
@@ -501,7 +501,7 @@
}
// Creates a runtime setting to notify play-out (aka render) volume changes.
- // |volume| is the unnormalized volume, the maximum of which
+ // `volume` is the unnormalized volume, the maximum of which
static RuntimeSetting CreatePlayoutVolumeChange(int volume) {
return {Type::kPlayoutVolumeChange, volume};
}
@@ -562,13 +562,13 @@
//
// It is also not necessary to call if the audio parameters (sample
// rate and number of channels) have changed. Passing updated parameters
- // directly to |ProcessStream()| and |ProcessReverseStream()| is permissible.
+ // directly to `ProcessStream()` and `ProcessReverseStream()` is permissible.
// If the parameters are known at init-time though, they may be provided.
// TODO(webrtc:5298): Change to return void.
virtual int Initialize() = 0;
// The int16 interfaces require:
- // - only |NativeRate|s be used
+ // - only `NativeRate`s be used
// - that the input, output and reverse rates must match
// - that |processing_config.output_stream()| matches
// |processing_config.input_stream()|.
@@ -616,7 +616,7 @@
virtual bool PostRuntimeSetting(RuntimeSetting setting) = 0;
// Accepts and produces a 10 ms frame interleaved 16 bit integer audio as
- // specified in |input_config| and |output_config|. |src| and |dest| may use
+ // specified in `input_config` and `output_config`. `src` and `dest` may use
// the same memory, if desired.
virtual int ProcessStream(const int16_t* const src,
const StreamConfig& input_config,
@@ -624,35 +624,35 @@
int16_t* const dest) = 0;
// Accepts deinterleaved float audio with the range [-1, 1]. Each element of
- // |src| points to a channel buffer, arranged according to |input_stream|. At
- // output, the channels will be arranged according to |output_stream| in
- // |dest|.
+ // `src` points to a channel buffer, arranged according to `input_stream`. At
+ // output, the channels will be arranged according to `output_stream` in
+ // `dest`.
//
- // The output must have one channel or as many channels as the input. |src|
- // and |dest| may use the same memory, if desired.
+ // The output must have one channel or as many channels as the input. `src`
+ // and `dest` may use the same memory, if desired.
virtual int ProcessStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) = 0;
// Accepts and produces a 10 ms frame of interleaved 16 bit integer audio for
- // the reverse direction audio stream as specified in |input_config| and
- // |output_config|. |src| and |dest| may use the same memory, if desired.
+ // the reverse direction audio stream as specified in `input_config` and
+ // `output_config`. `src` and `dest` may use the same memory, if desired.
virtual int ProcessReverseStream(const int16_t* const src,
const StreamConfig& input_config,
const StreamConfig& output_config,
int16_t* const dest) = 0;
// Accepts deinterleaved float audio with the range [-1, 1]. Each element of
- // |data| points to a channel buffer, arranged according to |reverse_config|.
+ // `data` points to a channel buffer, arranged according to `reverse_config`.
virtual int ProcessReverseStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) = 0;
// Accepts deinterleaved float audio with the range [-1, 1]. Each element
- // of |data| points to a channel buffer, arranged according to
- // |reverse_config|.
+ // of `data` points to a channel buffer, arranged according to
+ // `reverse_config`.
virtual int AnalyzeReverseStream(const float* const* data,
const StreamConfig& reverse_config) = 0;
@@ -675,7 +675,7 @@
// This must be called if and only if echo processing is enabled.
//
- // Sets the |delay| in ms between ProcessReverseStream() receiving a far-end
+ // Sets the `delay` in ms between ProcessReverseStream() receiving a far-end
// frame and ProcessStream() receiving a near-end frame containing the
// corresponding echo. On the client-side this can be expressed as
// delay = (t_render - t_analyze) + (t_process - t_capture)
@@ -695,10 +695,10 @@
// Creates and attaches an webrtc::AecDump for recording debugging
// information.
- // The |worker_queue| may not be null and must outlive the created
+ // The `worker_queue` may not be null and must outlive the created
// AecDump instance. |max_log_size_bytes == -1| means the log size
- // will be unlimited. |handle| may not be null. The AecDump takes
- // responsibility for |handle| and closes it in the destructor. A
+ // will be unlimited. `handle` may not be null. The AecDump takes
+ // responsibility for `handle` and closes it in the destructor. A
// return value of true indicates that the file has been
// sucessfully opened, while a value of false indicates that
// opening the file failed.
@@ -726,7 +726,7 @@
// Get audio processing statistics.
virtual AudioProcessingStats GetStatistics() = 0;
- // TODO(webrtc:5298) Deprecated variant. The |has_remote_tracks| argument
+ // TODO(webrtc:5298) Deprecated variant. The `has_remote_tracks` argument
// should be set if there are active remote tracks (this would usually be true
// during a call). If there are no remote tracks some of the stats will not be
// set by AudioProcessing, because they only make sense if there is at least
diff --git a/modules/audio_processing/include/audio_processing_statistics.h b/modules/audio_processing/include/audio_processing_statistics.h
index 87babee..c81d7eb 100644
--- a/modules/audio_processing/include/audio_processing_statistics.h
+++ b/modules/audio_processing/include/audio_processing_statistics.h
@@ -50,9 +50,9 @@
// The delay metrics consists of the delay median and standard deviation. It
// also consists of the fraction of delay estimates that can make the echo
// cancellation perform poorly. The values are aggregated until the first
- // call to |GetStatistics()| and afterwards aggregated and updated every
+ // call to `GetStatistics()` and afterwards aggregated and updated every
// second. Note that if there are several clients pulling metrics from
- // |GetStatistics()| during a session the first call from any of them will
+ // `GetStatistics()` during a session the first call from any of them will
// change to one second aggregation window for all.
absl::optional<int32_t> delay_median_ms;
absl::optional<int32_t> delay_standard_deviation_ms;
@@ -64,7 +64,7 @@
// The instantaneous delay estimate produced in the AEC. The unit is in
// milliseconds and the value is the instantaneous value at the time of the
- // call to |GetStatistics()|.
+ // call to `GetStatistics()`.
absl::optional<int32_t> delay_ms;
};
diff --git a/modules/audio_processing/level_estimator.h b/modules/audio_processing/level_estimator.h
index 1d8a071..d2bcfa1 100644
--- a/modules/audio_processing/level_estimator.h
+++ b/modules/audio_processing/level_estimator.h
@@ -35,7 +35,7 @@
// The computation follows: https://tools.ietf.org/html/rfc6465
// with the intent that it can provide the RTP audio level indication.
//
- // Frames passed to ProcessStream() with an |_energy| of zero are considered
+ // Frames passed to ProcessStream() with an `_energy` of zero are considered
// to have been muted. The RMS of the frame will be interpreted as -127.
int RMS() { return rms_.Average(); }
diff --git a/modules/audio_processing/optionally_built_submodule_creators.h b/modules/audio_processing/optionally_built_submodule_creators.h
index c96e66f..7de337b 100644
--- a/modules/audio_processing/optionally_built_submodule_creators.h
+++ b/modules/audio_processing/optionally_built_submodule_creators.h
@@ -20,7 +20,7 @@
// These overrides are only to be used for testing purposes.
// Each flag emulates a preprocessor macro to exclude a submodule of APM from
// the build, e.g. WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR. If the corresponding
-// flag |transient_suppression| is enabled, then the creators will return
+// flag `transient_suppression` is enabled, then the creators will return
// nullptr instead of a submodule instance, as if the macro had been defined.
struct ApmSubmoduleCreationOverrides {
bool transient_suppression = false;
@@ -29,7 +29,7 @@
// Creates a transient suppressor.
// Will instead return nullptr if one of the following is true:
// * WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR is defined
-// * The corresponding override in |overrides| is enabled.
+// * The corresponding override in `overrides` is enabled.
std::unique_ptr<TransientSuppressor> CreateTransientSuppressor(
const ApmSubmoduleCreationOverrides& overrides);
diff --git a/modules/audio_processing/residual_echo_detector.h b/modules/audio_processing/residual_echo_detector.h
index 5d18ecb..44252af 100644
--- a/modules/audio_processing/residual_echo_detector.h
+++ b/modules/audio_processing/residual_echo_detector.h
@@ -51,12 +51,12 @@
private:
static int instance_count_;
std::unique_ptr<ApmDataDumper> data_dumper_;
- // Keep track if the |Process| function has been previously called.
+ // Keep track if the `Process` function has been previously called.
bool first_process_call_ = true;
// Buffer for storing the power of incoming farend buffers. This is needed for
// cases where calls to BufferFarend and Process are jittery.
CircularBuffer render_buffer_;
- // Count how long ago it was that the size of |render_buffer_| was zero. This
+ // Count how long ago it was that the size of `render_buffer_` was zero. This
// value is also reset to zero when clock drift is detected and a value from
// the renderbuffer is discarded, even though the buffer is not actually zero
// at that point. This is done to avoid repeatedly removing elements in this
diff --git a/modules/audio_processing/rms_level.h b/modules/audio_processing/rms_level.h
index e1a6d56..4955d1b 100644
--- a/modules/audio_processing/rms_level.h
+++ b/modules/audio_processing/rms_level.h
@@ -47,7 +47,7 @@
void Analyze(rtc::ArrayView<const int16_t> data);
void Analyze(rtc::ArrayView<const float> data);
- // If all samples with the given |length| have a magnitude of zero, this is
+ // If all samples with the given `length` have a magnitude of zero, this is
// a shortcut to avoid some computation.
void AnalyzeMuted(size_t length);
@@ -62,7 +62,7 @@
Levels AverageAndPeak();
private:
- // Compares |block_size| with |block_size_|. If they are different, calls
+ // Compares `block_size` with `block_size_`. If they are different, calls
// Reset() and stores the new size.
void CheckBlockSize(size_t block_size);
diff --git a/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc
index 1f05f43..c61110f 100644
--- a/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/modules/audio_processing/test/audio_processing_simulator.cc
@@ -206,7 +206,7 @@
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 @@
// 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_);
diff --git a/modules/audio_processing/test/audioproc_float_impl.h b/modules/audio_processing/test/audioproc_float_impl.h
index 0687c43..5ed3aef 100644
--- a/modules/audio_processing/test/audioproc_float_impl.h
+++ b/modules/audio_processing/test/audioproc_float_impl.h
@@ -19,11 +19,11 @@
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 @@
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[],
diff --git a/modules/audio_processing/test/conversational_speech/simulator.cc b/modules/audio_processing/test/conversational_speech/simulator.cc
index 0591252..20c8608 100644
--- a/modules/audio_processing/test/conversational_speech/simulator.cc
+++ b/modules/audio_processing/test/conversational_speech/simulator.cc
@@ -125,8 +125,8 @@
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 @@
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());
diff --git a/modules/audio_processing/test/fake_recording_device.h b/modules/audio_processing/test/fake_recording_device.h
index b4d2a10..4017037 100644
--- a/modules/audio_processing/test/fake_recording_device.h
+++ b/modules/audio_processing/test/fake_recording_device.h
@@ -52,14 +52,14 @@
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);
diff --git a/modules/audio_processing/test/fake_recording_device_unittest.cc b/modules/audio_processing/test/fake_recording_device_unittest.cc
index 74bb47f..2ac8b1d 100644
--- a/modules/audio_processing/test/fake_recording_device_unittest.cc
+++ b/modules/audio_processing/test/fake_recording_device_unittest.cc
@@ -75,7 +75,7 @@
}
// 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());
diff --git a/modules/audio_processing/test/performance_timer.h b/modules/audio_processing/test/performance_timer.h
index b6e0da7..5375ba7 100644
--- a/modules/audio_processing/test/performance_timer.h
+++ b/modules/audio_processing/test/performance_timer.h
@@ -31,7 +31,7 @@
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;
diff --git a/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_boxplot.py b/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_boxplot.py
index 60d1e85..c425885 100644
--- a/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_boxplot.py
+++ b/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_boxplot.py
@@ -88,7 +88,7 @@
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
]
diff --git a/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py b/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py
index b0be37c..ecae2ed 100644
--- a/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py
+++ b/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py
@@ -133,7 +133,7 @@
{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.
"""
diff --git a/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores.py b/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores.py
index 23f6eff..59c5f74 100644
--- a/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores.py
+++ b/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores.py
@@ -397,7 +397,7 @@
# 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]
diff --git a/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py b/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py
index fb3aae0..0affbed 100644
--- a/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py
+++ b/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py
@@ -363,7 +363,7 @@
@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 @@
@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
diff --git a/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer.py b/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer.py
index f9125fa..af022bd 100644
--- a/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer.py
+++ b/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer.py
@@ -47,7 +47,7 @@
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:
diff --git a/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py b/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py
index e41637c..95e8019 100644
--- a/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py
+++ b/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py
@@ -174,7 +174,7 @@
"""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 @@
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 @@
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')
diff --git a/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation.py b/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation.py
index fe30c9c..69b3a16 100644
--- a/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation.py
+++ b/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation.py
@@ -264,7 +264,7 @@
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.
diff --git a/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_unittest.py b/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_unittest.py
index 6d0cb79..f75098a 100644
--- a/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_unittest.py
+++ b/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_unittest.py
@@ -116,7 +116,7 @@
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 @@
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 @@
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)
diff --git a/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc b/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc
index 9906eca..b47f622 100644
--- a/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc
+++ b/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc
@@ -63,7 +63,7 @@
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 =
diff --git a/modules/audio_processing/test/test_utils.h b/modules/audio_processing/test/test_utils.h
index e2d243e..30674cb 100644
--- a/modules/audio_processing/test/test_utils.h
+++ b/modules/audio_processing/test/test_utils.h
@@ -78,7 +78,7 @@
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 @@
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);
diff --git a/modules/audio_processing/three_band_filter_bank.cc b/modules/audio_processing/three_band_filter_bank.cc
index 2a7d272..fc665ef 100644
--- a/modules/audio_processing/three_band_filter_bank.cc
+++ b/modules/audio_processing/three_band_filter_bank.cc
@@ -39,16 +39,16 @@
namespace webrtc {
namespace {
-// Factors to take into account when choosing |kFilterSize|:
-// 1. Higher |kFilterSize|, means faster transition, which ensures less
+// Factors to take into account when choosing `kFilterSize`:
+// 1. Higher `kFilterSize`, means faster transition, which ensures less
// aliasing. This is especially important when there is non-linear
// processing between the splitting and merging.
// 2. The delay that this filter bank introduces is
-// |kNumBands| * |kSparsity| * |kFilterSize| / 2, so it increases linearly
-// with |kFilterSize|.
-// 3. The computation complexity also increases linearly with |kFilterSize|.
+// `kNumBands` * `kSparsity` * `kFilterSize` / 2, so it increases linearly
+// with `kFilterSize`.
+// 3. The computation complexity also increases linearly with `kFilterSize`.
-// The Matlab code to generate these |kFilterCoeffs| is:
+// The Matlab code to generate these `kFilterCoeffs` is:
//
// N = kNumBands * kSparsity * kFilterSize - 1;
// h = fir1(N, 1 / (2 * kNumBands), kaiser(N + 1, 3.5));
@@ -59,7 +59,7 @@
// Because the total bandwidth of the lower and higher band is double the middle
// one (because of the spectrum parity), the low-pass prototype is half the
-// bandwidth of 1 / (2 * |kNumBands|) and is then shifted with cosine modulation
+// bandwidth of 1 / (2 * `kNumBands`) and is then shifted with cosine modulation
// to the right places.
// A Kaiser window is used because of its flexibility and the alpha is set to
// 3.5, since that sets a stop band attenuation of 40dB ensuring a fast
@@ -100,8 +100,8 @@
{1.f, -2.f, 1.f},
{1.73205077f, 0.f, -1.73205077f}};
-// Filters the input signal |in| with the filter |filter| using a shift by
-// |in_shift|, taking into account the previous state.
+// Filters the input signal `in` with the filter `filter` using a shift by
+// `in_shift`, taking into account the previous state.
void FilterCore(
rtc::ArrayView<const float, kFilterSize> filter,
rtc::ArrayView<const float, ThreeBandFilterBank::kSplitBandSize> in,
@@ -164,10 +164,10 @@
ThreeBandFilterBank::~ThreeBandFilterBank() = default;
// The analysis can be separated in these steps:
-// 1. Serial to parallel downsampling by a factor of |kNumBands|.
-// 2. Filtering of |kSparsity| different delayed signals with polyphase
+// 1. Serial to parallel downsampling by a factor of `kNumBands`.
+// 2. Filtering of `kSparsity` different delayed signals with polyphase
// decomposition of the low-pass prototype filter and upsampled by a factor
-// of |kSparsity|.
+// of `kSparsity`.
// 3. Modulating with cosines and accumulating to get the desired band.
void ThreeBandFilterBank::Analysis(
rtc::ArrayView<const float, kFullBandSize> in,
@@ -222,9 +222,9 @@
// The synthesis can be separated in these steps:
// 1. Modulating with cosines.
// 2. Filtering each one with a polyphase decomposition of the low-pass
-// prototype filter upsampled by a factor of |kSparsity| and accumulating
-// |kSparsity| signals with different delays.
-// 3. Parallel to serial upsampling by a factor of |kNumBands|.
+// prototype filter upsampled by a factor of `kSparsity` and accumulating
+// `kSparsity` signals with different delays.
+// 3. Parallel to serial upsampling by a factor of `kNumBands`.
void ThreeBandFilterBank::Synthesis(
rtc::ArrayView<const rtc::ArrayView<float>, ThreeBandFilterBank::kNumBands>
in,
diff --git a/modules/audio_processing/three_band_filter_bank.h b/modules/audio_processing/three_band_filter_bank.h
index e6346de..db66cab 100644
--- a/modules/audio_processing/three_band_filter_bank.h
+++ b/modules/audio_processing/three_band_filter_bank.h
@@ -55,13 +55,13 @@
ThreeBandFilterBank();
~ThreeBandFilterBank();
- // Splits |in| of size kFullBandSize into 3 downsampled frequency bands in
- // |out|, each of size 160.
+ // Splits `in` of size kFullBandSize into 3 downsampled frequency bands in
+ // `out`, each of size 160.
void Analysis(rtc::ArrayView<const float, kFullBandSize> in,
rtc::ArrayView<const rtc::ArrayView<float>, kNumBands> out);
- // Merges the 3 downsampled frequency bands in |in|, each of size 160, into
- // |out|, which is of size kFullBandSize.
+ // Merges the 3 downsampled frequency bands in `in`, each of size 160, into
+ // `out`, which is of size kFullBandSize.
void Synthesis(rtc::ArrayView<const rtc::ArrayView<float>, kNumBands> in,
rtc::ArrayView<float, kFullBandSize> out);
diff --git a/modules/audio_processing/transient/click_annotate.cc b/modules/audio_processing/transient/click_annotate.cc
index 21641f8..f3f040f 100644
--- a/modules/audio_processing/transient/click_annotate.cc
+++ b/modules/audio_processing/transient/click_annotate.cc
@@ -26,7 +26,7 @@
// Creates a send times array, one for each step.
// Each block that contains a transient, has an infinite send time.
// The resultant array is written to a DAT file
-// Returns -1 on error or |lost_packets| otherwise.
+// Returns -1 on error or `lost_packets` otherwise.
int main(int argc, char* argv[]) {
if (argc != 5) {
printf("\n%s - Application to generate a RTP timing file.\n\n", argv[0]);
diff --git a/modules/audio_processing/transient/dyadic_decimator.h b/modules/audio_processing/transient/dyadic_decimator.h
index fcb56b7..52467e8 100644
--- a/modules/audio_processing/transient/dyadic_decimator.h
+++ b/modules/audio_processing/transient/dyadic_decimator.h
@@ -18,7 +18,7 @@
namespace webrtc {
// Returns the proper length of the output buffer that you should use for the
-// given |in_length| and decimation |odd_sequence|.
+// given `in_length` and decimation `odd_sequence`.
// Return -1 on error.
inline size_t GetOutLengthToDyadicDecimate(size_t in_length,
bool odd_sequence) {
@@ -34,10 +34,10 @@
// Performs a dyadic decimation: removes every odd/even member of a sequence
// halving its overall length.
// Arguments:
-// in: array of |in_length|.
+// in: array of `in_length`.
// odd_sequence: If false, the odd members will be removed (1, 3, 5, ...);
// if true, the even members will be removed (0, 2, 4, ...).
-// out: array of |out_length|. |out_length| must be large enough to
+// out: array of `out_length`. `out_length` must be large enough to
// hold the decimated output. The necessary length can be provided by
// GetOutLengthToDyadicDecimate().
// Must be previously allocated.
diff --git a/modules/audio_processing/transient/dyadic_decimator_unittest.cc b/modules/audio_processing/transient/dyadic_decimator_unittest.cc
index 3e65a7b..e4776d6 100644
--- a/modules/audio_processing/transient/dyadic_decimator_unittest.cc
+++ b/modules/audio_processing/transient/dyadic_decimator_unittest.cc
@@ -42,7 +42,7 @@
static_cast<int16_t*>(NULL), kOutBufferLength);
EXPECT_EQ(0u, out_samples);
- // Less than required |out_length|.
+ // Less than required `out_length`.
out_samples = DyadicDecimate(test_buffer_even_len, kEvenBufferLength,
false, // Even sequence.
test_buffer_out, 2);
diff --git a/modules/audio_processing/transient/file_utils.h b/modules/audio_processing/transient/file_utils.h
index 6184017..b748337 100644
--- a/modules/audio_processing/transient/file_utils.h
+++ b/modules/audio_processing/transient/file_utils.h
@@ -50,63 +50,63 @@
// Returns 0 if correct, -1 on error.
int ConvertDoubleToByteArray(double value, uint8_t out_bytes[8]);
-// Reads |length| 16-bit integers from |file| to |buffer|.
-// |file| must be previously opened.
+// Reads `length` 16-bit integers from `file` to `buffer`.
+// `file` must be previously opened.
// Returns the number of 16-bit integers read or -1 on error.
size_t ReadInt16BufferFromFile(FileWrapper* file,
size_t length,
int16_t* buffer);
-// Reads |length| 16-bit integers from |file| and stores those values
-// (converting them) in |buffer|.
-// |file| must be previously opened.
+// Reads `length` 16-bit integers from `file` and stores those values
+// (converting them) in `buffer`.
+// `file` must be previously opened.
// Returns the number of 16-bit integers read or -1 on error.
size_t ReadInt16FromFileToFloatBuffer(FileWrapper* file,
size_t length,
float* buffer);
-// Reads |length| 16-bit integers from |file| and stores those values
-// (converting them) in |buffer|.
-// |file| must be previously opened.
+// Reads `length` 16-bit integers from `file` and stores those values
+// (converting them) in `buffer`.
+// `file` must be previously opened.
// Returns the number of 16-bit integers read or -1 on error.
size_t ReadInt16FromFileToDoubleBuffer(FileWrapper* file,
size_t length,
double* buffer);
-// Reads |length| floats in binary representation (4 bytes) from |file| to
-// |buffer|.
-// |file| must be previously opened.
+// Reads `length` floats in binary representation (4 bytes) from `file` to
+// `buffer`.
+// `file` must be previously opened.
// Returns the number of floats read or -1 on error.
size_t ReadFloatBufferFromFile(FileWrapper* file, size_t length, float* buffer);
-// Reads |length| doubles in binary representation (8 bytes) from |file| to
-// |buffer|.
-// |file| must be previously opened.
+// Reads `length` doubles in binary representation (8 bytes) from `file` to
+// `buffer`.
+// `file` must be previously opened.
// Returns the number of doubles read or -1 on error.
size_t ReadDoubleBufferFromFile(FileWrapper* file,
size_t length,
double* buffer);
-// Writes |length| 16-bit integers from |buffer| in binary representation (2
-// bytes) to |file|. It flushes |file|, so after this call there are no
+// Writes `length` 16-bit integers from `buffer` in binary representation (2
+// bytes) to `file`. It flushes `file`, so after this call there are no
// writings pending.
-// |file| must be previously opened.
+// `file` must be previously opened.
// Returns the number of doubles written or -1 on error.
size_t WriteInt16BufferToFile(FileWrapper* file,
size_t length,
const int16_t* buffer);
-// Writes |length| floats from |buffer| in binary representation (4 bytes) to
-// |file|. It flushes |file|, so after this call there are no writtings pending.
-// |file| must be previously opened.
+// Writes `length` floats from `buffer` in binary representation (4 bytes) to
+// `file`. It flushes `file`, so after this call there are no writtings pending.
+// `file` must be previously opened.
// Returns the number of doubles written or -1 on error.
size_t WriteFloatBufferToFile(FileWrapper* file,
size_t length,
const float* buffer);
-// Writes |length| doubles from |buffer| in binary representation (8 bytes) to
-// |file|. It flushes |file|, so after this call there are no writings pending.
-// |file| must be previously opened.
+// Writes `length` doubles from `buffer` in binary representation (8 bytes) to
+// `file`. It flushes `file`, so after this call there are no writings pending.
+// `file` must be previously opened.
// Returns the number of doubles written or -1 on error.
size_t WriteDoubleBufferToFile(FileWrapper* file,
size_t length,
diff --git a/modules/audio_processing/transient/moving_moments.h b/modules/audio_processing/transient/moving_moments.h
index 6dc0520..70451dc 100644
--- a/modules/audio_processing/transient/moving_moments.h
+++ b/modules/audio_processing/transient/moving_moments.h
@@ -26,13 +26,13 @@
// the last values of the moments. When needed.
class MovingMoments {
public:
- // Creates a Moving Moments object, that uses the last |length| values
+ // Creates a Moving Moments object, that uses the last `length` values
// (including the new value introduced in every new calculation).
explicit MovingMoments(size_t length);
~MovingMoments();
- // Calculates the new values using |in|. Results will be in the out buffers.
- // |first| and |second| must be allocated with at least |in_length|.
+ // Calculates the new values using `in`. Results will be in the out buffers.
+ // `first` and `second` must be allocated with at least `in_length`.
void CalculateMoments(const float* in,
size_t in_length,
float* first,
@@ -40,7 +40,7 @@
private:
size_t length_;
- // A queue holding the |length_| latest input values.
+ // A queue holding the `length_` latest input values.
std::queue<float> queue_;
// Sum of the values of the queue.
float sum_;
diff --git a/modules/audio_processing/transient/transient_detector.cc b/modules/audio_processing/transient/transient_detector.cc
index f03a2ea..5c35505 100644
--- a/modules/audio_processing/transient/transient_detector.cc
+++ b/modules/audio_processing/transient/transient_detector.cc
@@ -43,8 +43,8 @@
sample_rate_hz == ts::kSampleRate48kHz);
int samples_per_transient = sample_rate_hz * kTransientLengthMs / 1000;
// Adjustment to avoid data loss while downsampling, making
- // |samples_per_chunk_| and |samples_per_transient| always divisible by
- // |kLeaves|.
+ // `samples_per_chunk_` and `samples_per_transient` always divisible by
+ // `kLeaves`.
samples_per_chunk_ -= samples_per_chunk_ % kLeaves;
samples_per_transient -= samples_per_transient % kLeaves;
@@ -137,7 +137,7 @@
// In the current implementation we return the max of the current result and
// the previous results, so the high results have a width equals to
- // |transient_length|.
+ // `transient_length`.
return *std::max_element(previous_results_.begin(), previous_results_.end());
}
diff --git a/modules/audio_processing/transient/transient_detector.h b/modules/audio_processing/transient/transient_detector.h
index 5ede2e8..a3dbb7f 100644
--- a/modules/audio_processing/transient/transient_detector.h
+++ b/modules/audio_processing/transient/transient_detector.h
@@ -37,8 +37,8 @@
~TransientDetector();
- // Calculates the log-likelihood of the existence of a transient in |data|.
- // |data_length| has to be equal to |samples_per_chunk_|.
+ // Calculates the log-likelihood of the existence of a transient in `data`.
+ // `data_length` has to be equal to `samples_per_chunk_`.
// Returns a value between 0 and 1, as a non linear representation of this
// likelihood.
// Returns a negative value on error.
@@ -71,7 +71,7 @@
float last_second_moment_[kLeaves];
// We keep track of the previous results from the previous chunks, so it can
- // be used to effectively give results according to the |transient_length|.
+ // be used to effectively give results according to the `transient_length`.
std::deque<float> previous_results_;
// Number of chunks that are going to return only zeros at the beginning of
diff --git a/modules/audio_processing/transient/transient_suppressor.h b/modules/audio_processing/transient/transient_suppressor.h
index bb262b0..982ddbd 100644
--- a/modules/audio_processing/transient/transient_suppressor.h
+++ b/modules/audio_processing/transient/transient_suppressor.h
@@ -27,22 +27,22 @@
int detector_rate_hz,
int num_channels) = 0;
- // Processes a |data| chunk, and returns it with keystrokes suppressed from
+ // Processes a `data` chunk, and returns it with keystrokes suppressed from
// it. The float format is assumed to be int16 ranged. If there are more than
- // one channel, the chunks are concatenated one after the other in |data|.
- // |data_length| must be equal to |data_length_|.
- // |num_channels| must be equal to |num_channels_|.
- // A sub-band, ideally the higher, can be used as |detection_data|. If it is
- // NULL, |data| is used for the detection too. The |detection_data| is always
+ // one channel, the chunks are concatenated one after the other in `data`.
+ // `data_length` must be equal to `data_length_`.
+ // `num_channels` must be equal to `num_channels_`.
+ // A sub-band, ideally the higher, can be used as `detection_data`. If it is
+ // NULL, `data` is used for the detection too. The `detection_data` is always
// assumed mono.
// If a reference signal (e.g. keyboard microphone) is available, it can be
- // passed in as |reference_data|. It is assumed mono and must have the same
- // length as |data|. NULL is accepted if unavailable.
+ // passed in as `reference_data`. It is assumed mono and must have the same
+ // length as `data`. NULL is accepted if unavailable.
// This suppressor performs better if voice information is available.
- // |voice_probability| is the probability of voice being present in this chunk
- // of audio. If voice information is not available, |voice_probability| must
+ // `voice_probability` is the probability of voice being present in this chunk
+ // of audio. If voice information is not available, `voice_probability` must
// always be set to 1.
- // |key_pressed| determines if a key was pressed on this audio chunk.
+ // `key_pressed` determines if a key was pressed on this audio chunk.
// Returns 0 on success and -1 otherwise.
virtual int Suppress(float* data,
size_t data_length,
diff --git a/modules/audio_processing/transient/transient_suppressor_impl.cc b/modules/audio_processing/transient/transient_suppressor_impl.cc
index d515d30..8e43d78 100644
--- a/modules/audio_processing/transient/transient_suppressor_impl.cc
+++ b/modules/audio_processing/transient/transient_suppressor_impl.cc
@@ -194,7 +194,7 @@
using_reference_ = detector_->using_reference();
- // |detector_smoothed_| follows the |detector_result| when this last one is
+ // `detector_smoothed_` follows the `detector_result` when this last one is
// increasing, but has an exponential decaying tail to be able to suppress
// the ringing of keyclicks.
float smooth_factor = using_reference_ ? 0.6 : 0.1;
@@ -223,7 +223,7 @@
}
// This should only be called when detection is enabled. UpdateBuffers() must
-// have been called. At return, |out_buffer_| will be filled with the
+// have been called. At return, `out_buffer_` will be filled with the
// processed output.
void TransientSuppressorImpl::Suppress(float* in_ptr,
float* spectral_mean,
@@ -325,7 +325,7 @@
}
// Shift buffers to make way for new data. Must be called after
-// |detection_enabled_| is updated by UpdateKeypress().
+// `detection_enabled_` is updated by UpdateKeypress().
void TransientSuppressorImpl::UpdateBuffers(float* data) {
// TODO(aluebs): Change to ring buffer.
memmove(in_buffer_.get(), &in_buffer_[data_length_],
@@ -350,9 +350,9 @@
}
// Restores the unvoiced signal if a click is present.
-// Attenuates by a certain factor every peak in the |fft_buffer_| that exceeds
-// the spectral mean. The attenuation depends on |detector_smoothed_|.
-// If a restoration takes place, the |magnitudes_| are updated to the new value.
+// Attenuates by a certain factor every peak in the `fft_buffer_` that exceeds
+// the spectral mean. The attenuation depends on `detector_smoothed_`.
+// If a restoration takes place, the `magnitudes_` are updated to the new value.
void TransientSuppressorImpl::HardRestoration(float* spectral_mean) {
const float detector_result =
1.f - std::pow(1.f - detector_smoothed_, using_reference_ ? 200.f : 50.f);
@@ -376,10 +376,10 @@
}
// Restores the voiced signal if a click is present.
-// Attenuates by a certain factor every peak in the |fft_buffer_| that exceeds
+// Attenuates by a certain factor every peak in the `fft_buffer_` that exceeds
// the spectral mean and that is lower than some function of the current block
-// frequency mean. The attenuation depends on |detector_smoothed_|.
-// If a restoration takes place, the |magnitudes_| are updated to the new value.
+// frequency mean. The attenuation depends on `detector_smoothed_`.
+// If a restoration takes place, the `magnitudes_` are updated to the new value.
void TransientSuppressorImpl::SoftRestoration(float* spectral_mean) {
// Get the spectral magnitude mean of the current block.
float block_frequency_mean = 0;
diff --git a/modules/audio_processing/transient/transient_suppressor_impl.h b/modules/audio_processing/transient/transient_suppressor_impl.h
index 4737af5..fa8186e 100644
--- a/modules/audio_processing/transient/transient_suppressor_impl.h
+++ b/modules/audio_processing/transient/transient_suppressor_impl.h
@@ -34,22 +34,22 @@
int detector_rate_hz,
int num_channels) override;
- // Processes a |data| chunk, and returns it with keystrokes suppressed from
+ // Processes a `data` chunk, and returns it with keystrokes suppressed from
// it. The float format is assumed to be int16 ranged. If there are more than
- // one channel, the chunks are concatenated one after the other in |data|.
- // |data_length| must be equal to |data_length_|.
- // |num_channels| must be equal to |num_channels_|.
- // A sub-band, ideally the higher, can be used as |detection_data|. If it is
- // NULL, |data| is used for the detection too. The |detection_data| is always
+ // one channel, the chunks are concatenated one after the other in `data`.
+ // `data_length` must be equal to `data_length_`.
+ // `num_channels` must be equal to `num_channels_`.
+ // A sub-band, ideally the higher, can be used as `detection_data`. If it is
+ // NULL, `data` is used for the detection too. The `detection_data` is always
// assumed mono.
// If a reference signal (e.g. keyboard microphone) is available, it can be
- // passed in as |reference_data|. It is assumed mono and must have the same
- // length as |data|. NULL is accepted if unavailable.
+ // passed in as `reference_data`. It is assumed mono and must have the same
+ // length as `data`. NULL is accepted if unavailable.
// This suppressor performs better if voice information is available.
- // |voice_probability| is the probability of voice being present in this chunk
- // of audio. If voice information is not available, |voice_probability| must
+ // `voice_probability` is the probability of voice being present in this chunk
+ // of audio. If voice information is not available, `voice_probability` must
// always be set to 1.
- // |key_pressed| determines if a key was pressed on this audio chunk.
+ // `key_pressed` determines if a key was pressed on this audio chunk.
// Returns 0 on success and -1 otherwise.
int Suppress(float* data,
size_t data_length,
diff --git a/modules/audio_processing/transient/wpd_node.h b/modules/audio_processing/transient/wpd_node.h
index 6a52fb7..41614fa 100644
--- a/modules/audio_processing/transient/wpd_node.h
+++ b/modules/audio_processing/transient/wpd_node.h
@@ -25,7 +25,7 @@
WPDNode(size_t length, const float* coefficients, size_t coefficients_length);
~WPDNode();
- // Updates the node data. |parent_data| / 2 must be equals to |length_|.
+ // Updates the node data. `parent_data` / 2 must be equals to `length_`.
// Returns 0 if correct, and -1 otherwise.
int Update(const float* parent_data, size_t parent_data_length);
diff --git a/modules/audio_processing/transient/wpd_tree.h b/modules/audio_processing/transient/wpd_tree.h
index c54220f..13cb8d9 100644
--- a/modules/audio_processing/transient/wpd_tree.h
+++ b/modules/audio_processing/transient/wpd_tree.h
@@ -65,7 +65,7 @@
// If level or index are out of bounds the function will return NULL.
WPDNode* NodeAt(int level, int index);
- // Updates all the nodes of the tree with the new data. |data_length| must be
+ // Updates all the nodes of the tree with the new data. `data_length` must be
// teh same that was used for the creation of the tree.
// Returns 0 if correct, and -1 otherwise.
int Update(const float* data, size_t data_length);
diff --git a/modules/audio_processing/typing_detection.h b/modules/audio_processing/typing_detection.h
index d8fb359..9d96583 100644
--- a/modules/audio_processing/typing_detection.h
+++ b/modules/audio_processing/typing_detection.h
@@ -22,7 +22,7 @@
// Run the detection algortihm. Shall be called every 10 ms. Returns true if
// typing is detected, or false if not, based on the update period as set with
- // SetParameters(). See |report_detection_update_period_| description below.
+ // SetParameters(). See `report_detection_update_period_` description below.
bool Process(bool key_pressed, bool vad_activity);
// Gets the time in seconds since the last detection.
@@ -43,14 +43,14 @@
int penalty_counter_;
// Counter since last time the detection status reported by Process() was
- // updated. See also |report_detection_update_period_|.
+ // updated. See also `report_detection_update_period_`.
int counter_since_last_detection_update_;
// The detection status to report. Updated every
- // |report_detection_update_period_| call to Process().
+ // `report_detection_update_period_` call to Process().
bool detection_to_report_;
- // What |detection_to_report_| should be set to next time it is updated.
+ // What `detection_to_report_` should be set to next time it is updated.
bool new_detection_to_report_;
// Settable threshold values.
@@ -61,10 +61,10 @@
// Penalty added for a typing + activity coincide.
int cost_per_typing_;
- // Threshold for |penalty_counter_|.
+ // Threshold for `penalty_counter_`.
int reporting_threshold_;
- // How much we reduce |penalty_counter_| every 10 ms.
+ // How much we reduce `penalty_counter_` every 10 ms.
int penalty_decay_;
// How old typing events we allow.
diff --git a/modules/audio_processing/utility/delay_estimator.cc b/modules/audio_processing/utility/delay_estimator.cc
index 73c70b0..6868392 100644
--- a/modules/audio_processing/utility/delay_estimator.cc
+++ b/modules/audio_processing/utility/delay_estimator.cc
@@ -55,7 +55,7 @@
return ((int)tmp);
}
-// Compares the |binary_vector| with all rows of the |binary_matrix| and counts
+// Compares the `binary_vector` with all rows of the `binary_matrix` and counts
// per row the number of times they have the same value.
//
// Inputs:
@@ -74,7 +74,7 @@
int32_t* bit_counts) {
int n = 0;
- // Compare |binary_vector| with all rows of the |binary_matrix|
+ // Compare `binary_vector` with all rows of the `binary_matrix`
for (; n < matrix_size; n++) {
bit_counts[n] = (int32_t)BitCount(binary_vector ^ binary_matrix[n]);
}
@@ -83,9 +83,9 @@
// Collects necessary statistics for the HistogramBasedValidation(). This
// function has to be called prior to calling HistogramBasedValidation(). The
// statistics updated and used by the HistogramBasedValidation() are:
-// 1. the number of |candidate_hits|, which states for how long we have had the
-// same |candidate_delay|
-// 2. the |histogram| of candidate delays over time. This histogram is
+// 1. the number of `candidate_hits`, which states for how long we have had the
+// same `candidate_delay`
+// 2. the `histogram` of candidate delays over time. This histogram is
// weighted with respect to a reliability measure and time-varying to cope
// with possible delay shifts.
// For further description see commented code.
@@ -93,7 +93,7 @@
// Inputs:
// - candidate_delay : The delay to validate.
// - valley_depth_q14 : The cost function has a valley/minimum at the
-// |candidate_delay| location. |valley_depth_q14| is the
+// `candidate_delay` location. `valley_depth_q14` is the
// cost function difference between the minimum and
// maximum locations. The value is in the Q14 domain.
// - valley_level_q14 : Is the cost function value at the minimum, in Q14.
@@ -109,37 +109,37 @@
int i = 0;
RTC_DCHECK_EQ(self->history_size, self->farend->history_size);
- // Reset |candidate_hits| if we have a new candidate.
+ // Reset `candidate_hits` if we have a new candidate.
if (candidate_delay != self->last_candidate_delay) {
self->candidate_hits = 0;
self->last_candidate_delay = candidate_delay;
}
self->candidate_hits++;
- // The |histogram| is updated differently across the bins.
- // 1. The |candidate_delay| histogram bin is increased with the
- // |valley_depth|, which is a simple measure of how reliable the
- // |candidate_delay| is. The histogram is not increased above
- // |kHistogramMax|.
+ // The `histogram` is updated differently across the bins.
+ // 1. The `candidate_delay` histogram bin is increased with the
+ // `valley_depth`, which is a simple measure of how reliable the
+ // `candidate_delay` is. The histogram is not increased above
+ // `kHistogramMax`.
self->histogram[candidate_delay] += valley_depth;
if (self->histogram[candidate_delay] > kHistogramMax) {
self->histogram[candidate_delay] = kHistogramMax;
}
- // 2. The histogram bins in the neighborhood of |candidate_delay| are
+ // 2. The histogram bins in the neighborhood of `candidate_delay` are
// unaffected. The neighborhood is defined as x + {-2, -1, 0, 1}.
- // 3. The histogram bins in the neighborhood of |last_delay| are decreased
- // with |decrease_in_last_set|. This value equals the difference between
- // the cost function values at the locations |candidate_delay| and
- // |last_delay| until we reach |max_hits_for_slow_change| consecutive hits
- // at the |candidate_delay|. If we exceed this amount of hits the
- // |candidate_delay| is a "potential" candidate and we start decreasing
- // these histogram bins more rapidly with |valley_depth|.
+ // 3. The histogram bins in the neighborhood of `last_delay` are decreased
+ // with `decrease_in_last_set`. This value equals the difference between
+ // the cost function values at the locations `candidate_delay` and
+ // `last_delay` until we reach `max_hits_for_slow_change` consecutive hits
+ // at the `candidate_delay`. If we exceed this amount of hits the
+ // `candidate_delay` is a "potential" candidate and we start decreasing
+ // these histogram bins more rapidly with `valley_depth`.
if (self->candidate_hits < max_hits_for_slow_change) {
decrease_in_last_set =
(self->mean_bit_counts[self->compare_delay] - valley_level_q14) *
kQ14Scaling;
}
- // 4. All other bins are decreased with |valley_depth|.
+ // 4. All other bins are decreased with `valley_depth`.
// TODO(bjornv): Investigate how to make this loop more efficient. Split up
// the loop? Remove parts that doesn't add too much.
for (i = 0; i < self->history_size; ++i) {
@@ -157,15 +157,15 @@
}
}
-// Validates the |candidate_delay|, estimated in WebRtc_ProcessBinarySpectrum(),
+// Validates the `candidate_delay`, estimated in WebRtc_ProcessBinarySpectrum(),
// based on a mix of counting concurring hits with a modified histogram
// of recent delay estimates. In brief a candidate is valid (returns 1) if it
// is the most likely according to the histogram. There are a couple of
// exceptions that are worth mentioning:
-// 1. If the |candidate_delay| < |last_delay| it can be that we are in a
+// 1. If the `candidate_delay` < `last_delay` it can be that we are in a
// non-causal state, breaking a possible echo control algorithm. Hence, we
// open up for a quicker change by allowing the change even if the
-// |candidate_delay| is not the most likely one according to the histogram.
+// `candidate_delay` is not the most likely one according to the histogram.
// 2. There's a minimum number of hits (kMinRequiredHits) and the histogram
// value has to reached a minimum (kMinHistogramThreshold) to be valid.
// 3. The action is also depending on the filter length used for echo control.
@@ -177,7 +177,7 @@
// - candidate_delay : The delay to validate.
//
// Return value:
-// - is_histogram_valid : 1 - The |candidate_delay| is valid.
+// - is_histogram_valid : 1 - The `candidate_delay` is valid.
// 0 - Otherwise.
static int HistogramBasedValidation(const BinaryDelayEstimator* self,
int candidate_delay) {
@@ -186,22 +186,22 @@
const int delay_difference = candidate_delay - self->last_delay;
int is_histogram_valid = 0;
- // The histogram based validation of |candidate_delay| is done by comparing
- // the |histogram| at bin |candidate_delay| with a |histogram_threshold|.
- // This |histogram_threshold| equals a |fraction| of the |histogram| at bin
- // |last_delay|. The |fraction| is a piecewise linear function of the
- // |delay_difference| between the |candidate_delay| and the |last_delay|
+ // The histogram based validation of `candidate_delay` is done by comparing
+ // the `histogram` at bin `candidate_delay` with a `histogram_threshold`.
+ // This `histogram_threshold` equals a `fraction` of the `histogram` at bin
+ // `last_delay`. The `fraction` is a piecewise linear function of the
+ // `delay_difference` between the `candidate_delay` and the `last_delay`
// allowing for a quicker move if
// i) a potential echo control filter can not handle these large differences.
- // ii) keeping |last_delay| instead of updating to |candidate_delay| could
+ // ii) keeping `last_delay` instead of updating to `candidate_delay` could
// force an echo control into a non-causal state.
// We further require the histogram to have reached a minimum value of
- // |kMinHistogramThreshold|. In addition, we also require the number of
- // |candidate_hits| to be more than |kMinRequiredHits| to remove spurious
+ // `kMinHistogramThreshold`. In addition, we also require the number of
+ // `candidate_hits` to be more than `kMinRequiredHits` to remove spurious
// values.
- // Calculate a comparison histogram value (|histogram_threshold|) that is
- // depending on the distance between the |candidate_delay| and |last_delay|.
+ // Calculate a comparison histogram value (`histogram_threshold`) that is
+ // depending on the distance between the `candidate_delay` and `last_delay`.
// TODO(bjornv): How much can we gain by turning the fraction calculation
// into tables?
if (delay_difference > self->allowed_offset) {
@@ -226,9 +226,9 @@
return is_histogram_valid;
}
-// Performs a robust validation of the |candidate_delay| estimated in
+// Performs a robust validation of the `candidate_delay` estimated in
// WebRtc_ProcessBinarySpectrum(). The algorithm takes the
-// |is_instantaneous_valid| and the |is_histogram_valid| and combines them
+// `is_instantaneous_valid` and the `is_histogram_valid` and combines them
// into a robust validation. The HistogramBasedValidation() has to be called
// prior to this call.
// For further description on how the combination is done, see commented code.
@@ -250,18 +250,18 @@
int is_robust = 0;
// The final robust validation is based on the two algorithms; 1) the
- // |is_instantaneous_valid| and 2) the histogram based with result stored in
- // |is_histogram_valid|.
- // i) Before we actually have a valid estimate (|last_delay| == -2), we say
+ // `is_instantaneous_valid` and 2) the histogram based with result stored in
+ // `is_histogram_valid`.
+ // i) Before we actually have a valid estimate (`last_delay` == -2), we say
// a candidate is valid if either algorithm states so
- // (|is_instantaneous_valid| OR |is_histogram_valid|).
+ // (`is_instantaneous_valid` OR `is_histogram_valid`).
is_robust =
(self->last_delay < 0) && (is_instantaneous_valid || is_histogram_valid);
// ii) Otherwise, we need both algorithms to be certain
- // (|is_instantaneous_valid| AND |is_histogram_valid|)
+ // (`is_instantaneous_valid` AND `is_histogram_valid`)
is_robust |= is_instantaneous_valid && is_histogram_valid;
// iii) With one exception, i.e., the histogram based algorithm can overrule
- // the instantaneous one if |is_histogram_valid| = 1 and the histogram
+ // the instantaneous one if `is_histogram_valid` = 1 and the histogram
// is significantly strong.
is_robust |= is_histogram_valid &&
(self->histogram[candidate_delay] > self->last_delay_histogram);
@@ -373,13 +373,13 @@
void WebRtc_AddBinaryFarSpectrum(BinaryDelayEstimatorFarend* handle,
uint32_t binary_far_spectrum) {
RTC_DCHECK(handle);
- // Shift binary spectrum history and insert current |binary_far_spectrum|.
+ // Shift binary spectrum history and insert current `binary_far_spectrum`.
memmove(&(handle->binary_far_history[1]), &(handle->binary_far_history[0]),
(handle->history_size - 1) * sizeof(uint32_t));
handle->binary_far_history[0] = binary_far_spectrum;
// Shift history of far-end binary spectrum bit counts and insert bit count
- // of current |binary_far_spectrum|.
+ // of current `binary_far_spectrum`.
memmove(&(handle->far_bit_counts[1]), &(handle->far_bit_counts[0]),
(handle->history_size - 1) * sizeof(int));
handle->far_bit_counts[0] = BitCount(binary_far_spectrum);
@@ -402,7 +402,7 @@
free(self->histogram);
self->histogram = NULL;
- // BinaryDelayEstimator does not have ownership of |farend|, hence we do not
+ // BinaryDelayEstimator does not have ownership of `farend`, hence we do not
// free the memory here. That should be handled separately by the user.
self->farend = NULL;
@@ -454,8 +454,8 @@
// Only update far-end buffers if we need.
history_size = WebRtc_AllocateFarendBufferMemory(far, history_size);
}
- // The extra array element in |mean_bit_counts| and |histogram| is a dummy
- // element only used while |last_delay| == -2, i.e., before we have a valid
+ // The extra array element in `mean_bit_counts` and `histogram` is a dummy
+ // element only used while `last_delay` == -2, i.e., before we have a valid
// estimate.
self->mean_bit_counts = static_cast<int32_t*>(
realloc(self->mean_bit_counts,
@@ -539,36 +539,36 @@
}
if (self->near_history_size > 1) {
// If we apply lookahead, shift near-end binary spectrum history. Insert
- // current |binary_near_spectrum| and pull out the delayed one.
+ // current `binary_near_spectrum` and pull out the delayed one.
memmove(&(self->binary_near_history[1]), &(self->binary_near_history[0]),
(self->near_history_size - 1) * sizeof(uint32_t));
self->binary_near_history[0] = binary_near_spectrum;
binary_near_spectrum = self->binary_near_history[self->lookahead];
}
- // Compare with delayed spectra and store the |bit_counts| for each delay.
+ // Compare with delayed spectra and store the `bit_counts` for each delay.
BitCountComparison(binary_near_spectrum, self->farend->binary_far_history,
self->history_size, self->bit_counts);
- // Update |mean_bit_counts|, which is the smoothed version of |bit_counts|.
+ // Update `mean_bit_counts`, which is the smoothed version of `bit_counts`.
for (i = 0; i < self->history_size; i++) {
- // |bit_counts| is constrained to [0, 32], meaning we can smooth with a
+ // `bit_counts` is constrained to [0, 32], meaning we can smooth with a
// factor up to 2^26. We use Q9.
int32_t bit_count = (self->bit_counts[i] << 9); // Q9.
- // Update |mean_bit_counts| only when far-end signal has something to
- // contribute. If |far_bit_counts| is zero the far-end signal is weak and
+ // Update `mean_bit_counts` only when far-end signal has something to
+ // contribute. If `far_bit_counts` is zero the far-end signal is weak and
// we likely have a poor echo condition, hence don't update.
if (self->farend->far_bit_counts[i] > 0) {
- // Make number of right shifts piecewise linear w.r.t. |far_bit_counts|.
+ // Make number of right shifts piecewise linear w.r.t. `far_bit_counts`.
int shifts = kShiftsAtZero;
shifts -= (kShiftsLinearSlope * self->farend->far_bit_counts[i]) >> 4;
WebRtc_MeanEstimatorFix(bit_count, shifts, &(self->mean_bit_counts[i]));
}
}
- // Find |candidate_delay|, |value_best_candidate| and |value_worst_candidate|
- // of |mean_bit_counts|.
+ // Find `candidate_delay`, `value_best_candidate` and `value_worst_candidate`
+ // of `mean_bit_counts`.
for (i = 0; i < self->history_size; i++) {
if (self->mean_bit_counts[i] < value_best_candidate) {
value_best_candidate = self->mean_bit_counts[i];
@@ -580,25 +580,25 @@
}
valley_depth = value_worst_candidate - value_best_candidate;
- // The |value_best_candidate| is a good indicator on the probability of
- // |candidate_delay| being an accurate delay (a small |value_best_candidate|
+ // The `value_best_candidate` is a good indicator on the probability of
+ // `candidate_delay` being an accurate delay (a small `value_best_candidate`
// means a good binary match). In the following sections we make a decision
- // whether to update |last_delay| or not.
+ // whether to update `last_delay` or not.
// 1) If the difference bit counts between the best and the worst delay
// candidates is too small we consider the situation to be unreliable and
- // don't update |last_delay|.
- // 2) If the situation is reliable we update |last_delay| if the value of the
+ // don't update `last_delay`.
+ // 2) If the situation is reliable we update `last_delay` if the value of the
// best candidate delay has a value less than
- // i) an adaptive threshold |minimum_probability|, or
- // ii) this corresponding value |last_delay_probability|, but updated at
+ // i) an adaptive threshold `minimum_probability`, or
+ // ii) this corresponding value `last_delay_probability`, but updated at
// this time instant.
- // Update |minimum_probability|.
+ // Update `minimum_probability`.
if ((self->minimum_probability > kProbabilityLowerLimit) &&
(valley_depth > kProbabilityMinSpread)) {
// The "hard" threshold can't be lower than 17 (in Q9).
// The valley in the curve also has to be distinct, i.e., the
- // difference between |value_worst_candidate| and |value_best_candidate| has
+ // difference between `value_worst_candidate` and `value_best_candidate` has
// to be large enough.
int32_t threshold = value_best_candidate + kProbabilityOffset;
if (threshold < kProbabilityLowerLimit) {
@@ -608,17 +608,17 @@
self->minimum_probability = threshold;
}
}
- // Update |last_delay_probability|.
+ // Update `last_delay_probability`.
// We use a Markov type model, i.e., a slowly increasing level over time.
self->last_delay_probability++;
- // Validate |candidate_delay|. We have a reliable instantaneous delay
+ // Validate `candidate_delay`. We have a reliable instantaneous delay
// estimate if
- // 1) The valley is distinct enough (|valley_depth| > |kProbabilityOffset|)
+ // 1) The valley is distinct enough (`valley_depth` > `kProbabilityOffset`)
// and
// 2) The depth of the valley is deep enough
- // (|value_best_candidate| < |minimum_probability|)
+ // (`value_best_candidate` < `minimum_probability`)
// and deeper than the best estimate so far
- // (|value_best_candidate| < |last_delay_probability|)
+ // (`value_best_candidate` < `last_delay_probability`)
valid_candidate = ((valley_depth > kProbabilityOffset) &&
((value_best_candidate < self->minimum_probability) ||
(value_best_candidate < self->last_delay_probability)));
@@ -650,7 +650,7 @@
(self->histogram[candidate_delay] > kLastHistogramMax
? kLastHistogramMax
: self->histogram[candidate_delay]);
- // Adjust the histogram if we made a change to |last_delay|, though it was
+ // Adjust the histogram if we made a change to `last_delay`, though it was
// not the most likely one according to the histogram.
if (self->histogram[candidate_delay] <
self->histogram[self->compare_delay]) {
@@ -680,7 +680,7 @@
// Simply a linear function of the histogram height at delay estimate.
quality = self->histogram[self->compare_delay] / kHistogramMax;
} else {
- // Note that |last_delay_probability| states how deep the minimum of the
+ // Note that `last_delay_probability` states how deep the minimum of the
// cost function is, so it is rather an error probability.
quality = (float)(kMaxBitCountsQ9 - self->last_delay_probability) /
kMaxBitCountsQ9;
diff --git a/modules/audio_processing/utility/delay_estimator.h b/modules/audio_processing/utility/delay_estimator.h
index df281bc..b6fc36a 100644
--- a/modules/audio_processing/utility/delay_estimator.h
+++ b/modules/audio_processing/utility/delay_estimator.h
@@ -81,7 +81,7 @@
//
// Return value:
// - BinaryDelayEstimatorFarend*
-// : Created |handle|. If the memory can't be allocated
+// : Created `handle`. If the memory can't be allocated
// or if any of the input parameters are invalid NULL
// is returned.
//
@@ -159,7 +159,7 @@
BinaryDelayEstimatorFarend* farend,
int max_lookahead);
-// Re-allocates |history_size| dependent buffers. The far-end buffers will be
+// Re-allocates `history_size` dependent buffers. The far-end buffers will be
// updated at the same time if needed.
//
// Input:
@@ -237,7 +237,7 @@
// delay value.
float WebRtc_binary_last_delay_quality(BinaryDelayEstimator* self);
-// Updates the |mean_value| recursively with a step size of 2^-|factor|. This
+// Updates the `mean_value` recursively with a step size of 2^-`factor`. This
// function is used internally in the Binary Delay Estimator as well as the
// Fixed point wrapper.
//
diff --git a/modules/audio_processing/utility/delay_estimator_internal.h b/modules/audio_processing/utility/delay_estimator_internal.h
index fce95d8..891e200 100644
--- a/modules/audio_processing/utility/delay_estimator_internal.h
+++ b/modules/audio_processing/utility/delay_estimator_internal.h
@@ -25,7 +25,7 @@
typedef struct {
// Pointers to mean values of spectrum.
SpectrumType* mean_far_spectrum;
- // |mean_far_spectrum| initialization indicator.
+ // `mean_far_spectrum` initialization indicator.
int far_spectrum_initialized;
int spectrum_size;
@@ -37,7 +37,7 @@
typedef struct {
// Pointers to mean values of spectrum.
SpectrumType* mean_near_spectrum;
- // |mean_near_spectrum| initialization indicator.
+ // `mean_near_spectrum` initialization indicator.
int near_spectrum_initialized;
int spectrum_size;
diff --git a/modules/audio_processing/utility/delay_estimator_unittest.cc b/modules/audio_processing/utility/delay_estimator_unittest.cc
index 65d8e14..651d836 100644
--- a/modules/audio_processing/utility/delay_estimator_unittest.cc
+++ b/modules/audio_processing/utility/delay_estimator_unittest.cc
@@ -80,7 +80,7 @@
memset(far_u16_, 1, sizeof(far_u16_));
memset(near_u16_, 2, sizeof(near_u16_));
// Construct a sequence of binary spectra used to verify delay estimate. The
- // |kSequenceLength| has to be long enough for the delay estimation to leave
+ // `kSequenceLength` has to be long enough for the delay estimation to leave
// the initialized state.
binary_spectrum_[0] = 1;
for (int i = 1; i < (kSequenceLength + kHistorySize); i++) {
@@ -132,7 +132,7 @@
// Initialize Binary Delay Estimator
WebRtc_InitBinaryDelayEstimator(binary_);
// Verify initialization. This does not guarantee a complete check, since
- // |last_delay| may be equal to -2 before initialization if done on the fly.
+ // `last_delay` may be equal to -2 before initialization if done on the fly.
EXPECT_EQ(-2, binary_->last_delay);
}
@@ -144,7 +144,7 @@
if (delay != -2) {
// Verify correct delay estimate. In the non-causal case the true delay
- // is equivalent with the |offset|.
+ // is equivalent with the `offset`.
EXPECT_EQ(offset, delay);
}
}
@@ -160,7 +160,7 @@
WebRtc_InitBinaryDelayEstimator(binary1);
WebRtc_InitBinaryDelayEstimator(binary2);
// Verify initialization. This does not guarantee a complete check, since
- // |last_delay| may be equal to -2 before initialization if done on the fly.
+ // `last_delay` may be equal to -2 before initialization if done on the fly.
EXPECT_EQ(-2, binary1->last_delay);
EXPECT_EQ(-2, binary2->last_delay);
for (int i = kLookahead; i < (kSequenceLength + kLookahead); i++) {
@@ -174,12 +174,12 @@
VerifyDelay(binary2,
far_offset + kLookahead + lookahead_offset + near_offset,
delay_2);
- // Expect the two delay estimates to be offset by |lookahead_offset| +
- // |near_offset| when we have left the initial state.
+ // Expect the two delay estimates to be offset by `lookahead_offset` +
+ // `near_offset` when we have left the initial state.
if ((delay_1 != -2) && (delay_2 != -2)) {
EXPECT_EQ(delay_1, delay_2 - lookahead_offset - near_offset);
}
- // For the case of identical signals |delay_1| and |delay_2| should match
+ // For the case of identical signals `delay_1` and `delay_2` should match
// all the time, unless one of them has robust validation turned on. In
// that case the robust validation leaves the initial state faster.
if ((near_offset == 0) && (lookahead_offset == 0)) {
@@ -208,8 +208,8 @@
BinaryDelayEstimator* binary2 = WebRtc_CreateBinaryDelayEstimator(
binary_farend_, kLookahead + lookahead_offset);
// Verify the delay for both causal and non-causal systems. For causal systems
- // the delay is equivalent with a positive |offset| of the far-end sequence.
- // For non-causal systems the delay is equivalent with a negative |offset| of
+ // the delay is equivalent with a positive `offset` of the far-end sequence.
+ // For non-causal systems the delay is equivalent with a negative `offset` of
// the far-end sequence.
binary_->robust_validation_enabled = ref_robust_validation;
binary2->robust_validation_enabled = robust_validation;
@@ -242,23 +242,23 @@
EXPECT_TRUE(handle == NULL);
// WebRtc_InitDelayEstimatorFarend() and WebRtc_InitDelayEstimator() should
- // return -1 if we have a NULL pointer as |handle|.
+ // return -1 if we have a NULL pointer as `handle`.
EXPECT_EQ(-1, WebRtc_InitDelayEstimatorFarend(NULL));
EXPECT_EQ(-1, WebRtc_InitDelayEstimator(NULL));
// WebRtc_AddFarSpectrumFloat() should return -1 if we have:
- // 1) NULL pointer as |handle|.
+ // 1) NULL pointer as `handle`.
// 2) NULL pointer as far-end spectrum.
// 3) Incorrect spectrum size.
EXPECT_EQ(-1, WebRtc_AddFarSpectrumFloat(NULL, far_f_, spectrum_size_));
- // Use |farend_handle_| which is properly created at SetUp().
+ // Use `farend_handle_` which is properly created at SetUp().
EXPECT_EQ(-1,
WebRtc_AddFarSpectrumFloat(farend_handle_, NULL, spectrum_size_));
EXPECT_EQ(-1, WebRtc_AddFarSpectrumFloat(farend_handle_, far_f_,
spectrum_size_ + 1));
// WebRtc_AddFarSpectrumFix() should return -1 if we have:
- // 1) NULL pointer as |handle|.
+ // 1) NULL pointer as `handle`.
// 2) NULL pointer as far-end spectrum.
// 3) Incorrect spectrum size.
// 4) Too high precision in far-end spectrum (Q-domain > 15).
@@ -271,8 +271,8 @@
spectrum_size_, 16));
// WebRtc_set_history_size() should return -1 if:
- // 1) |handle| is a NULL.
- // 2) |history_size| <= 1.
+ // 1) `handle` is a NULL.
+ // 2) `history_size` <= 1.
EXPECT_EQ(-1, WebRtc_set_history_size(NULL, 1));
EXPECT_EQ(-1, WebRtc_set_history_size(handle_, 1));
// WebRtc_history_size() should return -1 if:
@@ -293,43 +293,43 @@
EXPECT_EQ(-1, WebRtc_set_lookahead(handle_, -1));
// WebRtc_set_allowed_offset() should return -1 if we have:
- // 1) NULL pointer as |handle|.
- // 2) |allowed_offset| < 0.
+ // 1) NULL pointer as `handle`.
+ // 2) `allowed_offset` < 0.
EXPECT_EQ(-1, WebRtc_set_allowed_offset(NULL, 0));
EXPECT_EQ(-1, WebRtc_set_allowed_offset(handle_, -1));
EXPECT_EQ(-1, WebRtc_get_allowed_offset(NULL));
// WebRtc_enable_robust_validation() should return -1 if we have:
- // 1) NULL pointer as |handle|.
- // 2) Incorrect |enable| value (not 0 or 1).
+ // 1) NULL pointer as `handle`.
+ // 2) Incorrect `enable` value (not 0 or 1).
EXPECT_EQ(-1, WebRtc_enable_robust_validation(NULL, kEnable[0]));
EXPECT_EQ(-1, WebRtc_enable_robust_validation(handle_, -1));
EXPECT_EQ(-1, WebRtc_enable_robust_validation(handle_, 2));
// WebRtc_is_robust_validation_enabled() should return -1 if we have NULL
- // pointer as |handle|.
+ // pointer as `handle`.
EXPECT_EQ(-1, WebRtc_is_robust_validation_enabled(NULL));
// WebRtc_DelayEstimatorProcessFloat() should return -1 if we have:
- // 1) NULL pointer as |handle|.
+ // 1) NULL pointer as `handle`.
// 2) NULL pointer as near-end spectrum.
// 3) Incorrect spectrum size.
// 4) Non matching history sizes if multiple delay estimators using the same
// far-end reference.
EXPECT_EQ(-1,
WebRtc_DelayEstimatorProcessFloat(NULL, near_f_, spectrum_size_));
- // Use |handle_| which is properly created at SetUp().
+ // Use `handle_` which is properly created at SetUp().
EXPECT_EQ(-1,
WebRtc_DelayEstimatorProcessFloat(handle_, NULL, spectrum_size_));
EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFloat(handle_, near_f_,
spectrum_size_ + 1));
- // |tmp_handle| is already in a non-matching state.
+ // `tmp_handle` is already in a non-matching state.
EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFloat(tmp_handle, near_f_,
spectrum_size_));
// WebRtc_DelayEstimatorProcessFix() should return -1 if we have:
- // 1) NULL pointer as |handle|.
+ // 1) NULL pointer as `handle`.
// 2) NULL pointer as near-end spectrum.
// 3) Incorrect spectrum size.
// 4) Too high precision in near-end spectrum (Q-domain > 15).
@@ -343,12 +343,12 @@
spectrum_size_ + 1, 0));
EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(handle_, near_u16_,
spectrum_size_, 16));
- // |tmp_handle| is already in a non-matching state.
+ // `tmp_handle` is already in a non-matching state.
EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(tmp_handle, near_u16_,
spectrum_size_, 0));
WebRtc_FreeDelayEstimator(tmp_handle);
- // WebRtc_last_delay() should return -1 if we have a NULL pointer as |handle|.
+ // WebRtc_last_delay() should return -1 if we have a NULL pointer as `handle`.
EXPECT_EQ(-1, WebRtc_last_delay(NULL));
// Free any local memory if needed.
@@ -422,7 +422,7 @@
TEST_F(DelayEstimatorTest, CorrectLastDelay) {
// In this test we verify that we get the correct last delay upon valid call.
// We simply process the same data until we leave the initialized state
- // (|last_delay| = -2). Then we compare the Process() output with the
+ // (`last_delay` = -2). Then we compare the Process() output with the
// last_delay() call.
// TODO(bjornv): Update quality values for robust validation.
@@ -488,8 +488,8 @@
BinaryDelayEstimator* binary_handle = binary_;
// WebRtc_CreateBinaryDelayEstimator() should return -1 if we have a NULL
- // pointer as |binary_farend| or invalid input values. Upon failure, the
- // |binary_handle| should be NULL.
+ // pointer as `binary_farend` or invalid input values. Upon failure, the
+ // `binary_handle` should be NULL.
// Make sure we have a non-NULL value at start, so we can detect NULL after
// create failure.
binary_handle = WebRtc_CreateBinaryDelayEstimator(NULL, kLookahead);
@@ -506,12 +506,12 @@
int32_t mean_value_before = mean_value;
int32_t new_mean_value = mean_value * 2;
- // Increasing |mean_value|.
+ // Increasing `mean_value`.
WebRtc_MeanEstimatorFix(new_mean_value, 10, &mean_value);
EXPECT_LT(mean_value_before, mean_value);
EXPECT_GT(new_mean_value, mean_value);
- // Decreasing |mean_value|.
+ // Decreasing `mean_value`.
new_mean_value = mean_value / 2;
mean_value_before = mean_value;
WebRtc_MeanEstimatorFix(new_mean_value, 10, &mean_value);
@@ -569,7 +569,7 @@
TEST_F(DelayEstimatorTest, AllowedOffsetNoImpactWhenRobustValidationDisabled) {
// The same setup as in ExactDelayEstimateMultipleNearSameSpectrum with the
- // difference that |allowed_offset| is set for the reference binary delay
+ // difference that `allowed_offset` is set for the reference binary delay
// estimator.
binary_->allowed_offset = 10;
diff --git a/modules/audio_processing/utility/delay_estimator_wrapper.cc b/modules/audio_processing/utility/delay_estimator_wrapper.cc
index 8eac2f6..521a8a0 100644
--- a/modules/audio_processing/utility/delay_estimator_wrapper.cc
+++ b/modules/audio_processing/utility/delay_estimator_wrapper.cc
@@ -19,8 +19,8 @@
namespace webrtc {
-// Only bit |kBandFirst| through bit |kBandLast| are processed and
-// |kBandFirst| - |kBandLast| must be < 32.
+// Only bit `kBandFirst` through bit `kBandLast` are processed and
+// `kBandFirst` - `kBandLast` must be < 32.
enum { kBandFirst = 12 };
enum { kBandLast = 43 };
@@ -48,8 +48,8 @@
*mean_value += (new_value - *mean_value) * scale;
}
-// Computes the binary spectrum by comparing the input |spectrum| with a
-// |threshold_spectrum|. Float and fixed point versions.
+// Computes the binary spectrum by comparing the input `spectrum` with a
+// `threshold_spectrum`. Float and fixed point versions.
//
// Inputs:
// - spectrum : Spectrum of which the binary spectrum should be
@@ -69,11 +69,11 @@
RTC_DCHECK_LT(q_domain, 16);
if (!(*threshold_initialized)) {
- // Set the |threshold_spectrum| to half the input |spectrum| as starting
+ // Set the `threshold_spectrum` to half the input `spectrum` as starting
// value. This speeds up the convergence.
for (i = kBandFirst; i <= kBandLast; i++) {
if (spectrum[i] > 0) {
- // Convert input spectrum from Q(|q_domain|) to Q15.
+ // Convert input spectrum from Q(`q_domain`) to Q15.
int32_t spectrum_q15 = ((int32_t)spectrum[i]) << (15 - q_domain);
threshold_spectrum[i].int32_ = (spectrum_q15 >> 1);
*threshold_initialized = 1;
@@ -81,11 +81,11 @@
}
}
for (i = kBandFirst; i <= kBandLast; i++) {
- // Convert input spectrum from Q(|q_domain|) to Q15.
+ // Convert input spectrum from Q(`q_domain`) to Q15.
int32_t spectrum_q15 = ((int32_t)spectrum[i]) << (15 - q_domain);
- // Update the |threshold_spectrum|.
+ // Update the `threshold_spectrum`.
WebRtc_MeanEstimatorFix(spectrum_q15, 6, &(threshold_spectrum[i].int32_));
- // Convert |spectrum| at current frequency bin to a binary value.
+ // Convert `spectrum` at current frequency bin to a binary value.
if (spectrum_q15 > threshold_spectrum[i].int32_) {
out = SetBit(out, i - kBandFirst);
}
@@ -102,7 +102,7 @@
const float kScale = 1 / 64.0;
if (!(*threshold_initialized)) {
- // Set the |threshold_spectrum| to half the input |spectrum| as starting
+ // Set the `threshold_spectrum` to half the input `spectrum` as starting
// value. This speeds up the convergence.
for (i = kBandFirst; i <= kBandLast; i++) {
if (spectrum[i] > 0.0f) {
@@ -113,9 +113,9 @@
}
for (i = kBandFirst; i <= kBandLast; i++) {
- // Update the |threshold_spectrum|.
+ // Update the `threshold_spectrum`.
MeanEstimatorFloat(spectrum[i], kScale, &(threshold_spectrum[i].float_));
- // Convert |spectrum| at current frequency bin to a binary value.
+ // Convert `spectrum` at current frequency bin to a binary value.
if (spectrum[i] > threshold_spectrum[i].float_) {
out = SetBit(out, i - kBandFirst);
}
@@ -219,7 +219,7 @@
return -1;
}
if (far_q > 15) {
- // If |far_q| is larger than 15 we cannot guarantee no wrap around.
+ // If `far_q` is larger than 15 we cannot guarantee no wrap around.
return -1;
}
@@ -433,7 +433,7 @@
return -1;
}
if (near_q > 15) {
- // If |near_q| is larger than 15 we cannot guarantee no wrap around.
+ // If `near_q` is larger than 15 we cannot guarantee no wrap around.
return -1;
}
diff --git a/modules/audio_processing/utility/delay_estimator_wrapper.h b/modules/audio_processing/utility/delay_estimator_wrapper.h
index dbcafaf..a90cbe3 100644
--- a/modules/audio_processing/utility/delay_estimator_wrapper.h
+++ b/modules/audio_processing/utility/delay_estimator_wrapper.h
@@ -35,7 +35,7 @@
// determined together with WebRtc_set_lookahead().
//
// Return value:
-// - void* : Created |handle|. If the memory can't be allocated or
+// - void* : Created `handle`. If the memory can't be allocated or
// if any of the input parameters are invalid NULL is
// returned.
void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size);
@@ -85,13 +85,13 @@
// WebRtc_CreateDelayEstimatorFarend().
//
// Note that WebRtc_CreateDelayEstimator does not take
-// ownership of |farend_handle|, which has to be torn
+// ownership of `farend_handle`, which has to be torn
// down properly after this instance.
//
// - max_lookahead : Maximum amount of non-causal lookahead allowed. The
// actual amount of lookahead used can be controlled by
-// WebRtc_set_lookahead(...). The default |lookahead| is
-// set to |max_lookahead| at create time. Use
+// WebRtc_set_lookahead(...). The default `lookahead` is
+// set to `max_lookahead` at create time. Use
// WebRtc_set_lookahead(...) before start if a different
// value is desired.
//
@@ -106,12 +106,12 @@
// estimated.
//
// Note that the effective range of delay estimates is
-// [-|lookahead|,... ,|history_size|-|lookahead|)
-// where |history_size| is set through
+// [-`lookahead`,... ,`history_size`-`lookahead`)
+// where `history_size` is set through
// WebRtc_set_history_size().
//
// Return value:
-// - void* : Created |handle|. If the memory can't be allocated or
+// - void* : Created `handle`. If the memory can't be allocated or
// if any of the input parameters are invalid NULL is
// returned.
void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead);
@@ -129,12 +129,12 @@
// - actual_shifts : The actual number of shifts performed.
int WebRtc_SoftResetDelayEstimator(void* handle, int delay_shift);
-// Sets the effective |history_size| used. Valid values from 2. We simply need
-// at least two delays to compare to perform an estimate. If |history_size| is
+// Sets the effective `history_size` used. Valid values from 2. We simply need
+// at least two delays to compare to perform an estimate. If `history_size` is
// changed, buffers are reallocated filling in with zeros if necessary.
-// Note that changing the |history_size| affects both buffers in far-end and
+// Note that changing the `history_size` affects both buffers in far-end and
// near-end. Hence it is important to change all DelayEstimators that use the
-// same reference far-end, to the same |history_size| value.
+// same reference far-end, to the same `history_size` value.
// Inputs:
// - handle : Pointer to the delay estimation instance.
// - history_size : Effective history size to be used.
@@ -148,8 +148,8 @@
// - handle : Pointer to the delay estimation instance.
int WebRtc_history_size(const void* handle);
-// Sets the amount of |lookahead| to use. Valid values are [0, max_lookahead]
-// where |max_lookahead| was set at create time through
+// Sets the amount of `lookahead` to use. Valid values are [0, max_lookahead]
+// where `max_lookahead` was set at create time through
// WebRtc_CreateDelayEstimator(...).
//
// Input:
@@ -157,8 +157,8 @@
// - lookahead : The amount of lookahead to be used.
//
// Return value:
-// - new_lookahead : The actual amount of lookahead set, unless |handle| is
-// a NULL pointer or |lookahead| is invalid, for which an
+// - new_lookahead : The actual amount of lookahead set, unless `handle` is
+// a NULL pointer or `lookahead` is invalid, for which an
// error is returned.
int WebRtc_set_lookahead(void* handle, int lookahead);
@@ -167,12 +167,12 @@
// - handle : Pointer to the delay estimation instance.
int WebRtc_lookahead(void* handle);
-// Sets the |allowed_offset| used in the robust validation scheme. If the
+// Sets the `allowed_offset` used in the robust validation scheme. If the
// delay estimator is used in an echo control component, this parameter is
-// related to the filter length. In principle |allowed_offset| should be set to
+// related to the filter length. In principle `allowed_offset` should be set to
// the echo control filter length minus the expected echo duration, i.e., the
// delay offset the echo control can handle without quality regression. The
-// default value, used if not set manually, is zero. Note that |allowed_offset|
+// default value, used if not set manually, is zero. Note that `allowed_offset`
// has to be non-negative.
// Inputs:
// - handle : Pointer to the delay estimation instance.
@@ -180,7 +180,7 @@
// the echo control filter can handle.
int WebRtc_set_allowed_offset(void* handle, int allowed_offset);
-// Returns the |allowed_offset| in number of partitions.
+// Returns the `allowed_offset` in number of partitions.
int WebRtc_get_allowed_offset(const void* handle);
// Enables/Disables a robust validation functionality in the delay estimation.
diff --git a/modules/audio_processing/utility/pffft_wrapper.h b/modules/audio_processing/utility/pffft_wrapper.h
index 160f0da..983c2fd 100644
--- a/modules/audio_processing/utility/pffft_wrapper.h
+++ b/modules/audio_processing/utility/pffft_wrapper.h
@@ -51,7 +51,7 @@
// TODO(https://crbug.com/webrtc/9577): Consider adding a factory and making
// the ctor private.
// static std::unique_ptr<Pffft> Create(size_t fft_size,
- // FftType fft_type); Ctor. |fft_size| must be a supported size (see
+ // FftType fft_type); Ctor. `fft_size` must be a supported size (see
// Pffft::IsValidFftSize()). If not supported, the code will crash.
Pffft(size_t fft_size, FftType fft_type);
Pffft(const Pffft&) = delete;
@@ -73,9 +73,9 @@
// Computes the backward fast Fourier transform.
void BackwardTransform(const FloatBuffer& in, FloatBuffer* out, bool ordered);
- // Multiplies the frequency components of |fft_x| and |fft_y| and accumulates
- // them into |out|. The arrays must have been obtained with
- // ForwardTransform(..., /*ordered=*/false) - i.e., |fft_x| and |fft_y| must
+ // Multiplies the frequency components of `fft_x` and `fft_y` and accumulates
+ // them into `out`. The arrays must have been obtained with
+ // ForwardTransform(..., /*ordered=*/false) - i.e., `fft_x` and `fft_y` must
// not be ordered.
void FrequencyDomainConvolve(const FloatBuffer& fft_x,
const FloatBuffer& fft_y,
diff --git a/modules/audio_processing/vad/gmm.h b/modules/audio_processing/vad/gmm.h
index 93eb675..d9d68ec 100644
--- a/modules/audio_processing/vad/gmm.h
+++ b/modules/audio_processing/vad/gmm.h
@@ -20,13 +20,13 @@
// Where a 'mixture' is a Gaussian density.
struct GmmParameters {
- // weight[n] = log(w[n]) - |dimension|/2 * log(2*pi) - 1/2 * log(det(cov[n]));
+ // weight[n] = log(w[n]) - `dimension`/2 * log(2*pi) - 1/2 * log(det(cov[n]));
// where cov[n] is the covariance matrix of mixture n;
const double* weight;
- // pointer to the first element of a |num_mixtures|x|dimension| matrix
+ // pointer to the first element of a `num_mixtures`x`dimension` matrix
// where kth row is the mean of the kth mixture.
const double* mean;
- // pointer to the first element of a |num_mixtures|x|dimension|x|dimension|
+ // pointer to the first element of a `num_mixtures`x`dimension`x`dimension`
// 3D-matrix, where the kth 2D-matrix is the inverse of the covariance
// matrix of the kth mixture.
const double* covar_inverse;
@@ -36,8 +36,8 @@
int num_mixtures;
};
-// Evaluate the given GMM, according to |gmm_parameters|, at the given point
-// |x|. If the dimensionality of the given GMM is larger that the maximum
+// Evaluate the given GMM, according to `gmm_parameters`, at the given point
+// `x`. If the dimensionality of the given GMM is larger that the maximum
// acceptable dimension by the following function -1 is returned.
double EvaluateGmm(const double* x, const GmmParameters& gmm_parameters);
diff --git a/modules/audio_processing/vad/pitch_based_vad.h b/modules/audio_processing/vad/pitch_based_vad.h
index e005e23..fa3abc2 100644
--- a/modules/audio_processing/vad/pitch_based_vad.h
+++ b/modules/audio_processing/vad/pitch_based_vad.h
@@ -34,7 +34,7 @@
// p_combined: an array which contains the combined activity probabilities
// computed prior to the call of this function. The method,
// then, computes the voicing probabilities and combine them
- // with the given values. The result are returned in |p|.
+ // with the given values. The result are returned in `p`.
int VoicingProbability(const AudioFeatures& features, double* p_combined);
private:
diff --git a/modules/audio_processing/vad/pitch_internal.h b/modules/audio_processing/vad/pitch_internal.h
index 938745d..e382c1f 100644
--- a/modules/audio_processing/vad/pitch_internal.h
+++ b/modules/audio_processing/vad/pitch_internal.h
@@ -14,7 +14,7 @@
namespace webrtc {
// TODO(turajs): Write a description of this function. Also be consistent with
-// usage of |sampling_rate_hz| vs |kSamplingFreqHz|.
+// usage of `sampling_rate_hz` vs `kSamplingFreqHz`.
void GetSubframesPitchParameters(int sampling_rate_hz,
double* gains,
double* lags,
diff --git a/modules/audio_processing/vad/standalone_vad.h b/modules/audio_processing/vad/standalone_vad.h
index 3dff416..b084633 100644
--- a/modules/audio_processing/vad/standalone_vad.h
+++ b/modules/audio_processing/vad/standalone_vad.h
@@ -26,12 +26,12 @@
// Outputs
// p: a buffer where probabilities are written to.
- // length_p: number of elements of |p|.
+ // length_p: number of elements of `p`.
//
// return value:
// -1: if no audio is stored or VAD returns error.
// 0: in success.
- // In case of error the content of |activity| is unchanged.
+ // In case of error the content of `activity` is unchanged.
//
// Note that due to a high false-positive (VAD decision is active while the
// processed audio is just background noise) rate, stand-alone VAD is used as
diff --git a/modules/audio_processing/vad/standalone_vad_unittest.cc b/modules/audio_processing/vad/standalone_vad_unittest.cc
index 22b1f49..0fa2ed7 100644
--- a/modules/audio_processing/vad/standalone_vad_unittest.cc
+++ b/modules/audio_processing/vad/standalone_vad_unittest.cc
@@ -31,7 +31,7 @@
for (size_t n = 0; n < kMaxNumFrames; n++)
EXPECT_EQ(0, vad->AddAudio(data, kLength10Ms));
- // Pretend |p| is shorter that it should be.
+ // Pretend `p` is shorter that it should be.
EXPECT_EQ(-1, vad->GetActivity(p, kMaxNumFrames - 1));
EXPECT_EQ(0, vad->GetActivity(p, kMaxNumFrames));
diff --git a/modules/audio_processing/vad/vad_audio_proc.cc b/modules/audio_processing/vad/vad_audio_proc.cc
index 97cf651..aaf8214 100644
--- a/modules/audio_processing/vad/vad_audio_proc.cc
+++ b/modules/audio_processing/vad/vad_audio_proc.cc
@@ -132,7 +132,7 @@
kNumSubframeSamples + kNumPastSignalSamples, kLpcOrder);
}
-// Compute |kNum10msSubframes| sets of LPC coefficients, one per 10 ms input.
+// Compute `kNum10msSubframes` sets of LPC coefficients, one per 10 ms input.
// The analysis window is 15 ms long and it is centered on the first half of
// each 10ms sub-frame. This is equivalent to computing LPC coefficients for the
// first half of each 10 ms subframe.
@@ -169,7 +169,7 @@
return fractional_index;
}
-// 1 / A(z), where A(z) is defined by |lpc| is a model of the spectral envelope
+// 1 / A(z), where A(z) is defined by `lpc` is a model of the spectral envelope
// of the input signal. The local maximum of the spectral envelope corresponds
// with the local minimum of A(z). It saves complexity, as we save one
// inversion. Furthermore, we find the first local maximum of magnitude squared,
diff --git a/modules/audio_processing/vad/vad_circular_buffer.h b/modules/audio_processing/vad/vad_circular_buffer.h
index 46b03d4..c1806f9 100644
--- a/modules/audio_processing/vad/vad_circular_buffer.h
+++ b/modules/audio_processing/vad/vad_circular_buffer.h
@@ -38,8 +38,8 @@
// The mean value of the elements in the buffer. The return value is zero if
// buffer is empty, i.e. no value is inserted.
double Mean();
- // Remove transients. If the values exceed |val_threshold| for a period
- // shorter then or equal to |width_threshold|, then that period is considered
+ // Remove transients. If the values exceed `val_threshold` for a period
+ // shorter then or equal to `width_threshold`, then that period is considered
// transient and set to zero.
int RemoveTransient(int width_threshold, double val_threshold);
@@ -49,7 +49,7 @@
// insertion. |index = 1| is the one before the most recent insertion, and
// so on.
int Get(int index, double* value) const;
- // Set a given position to |value|. |index| is interpreted as above.
+ // Set a given position to `value`. `index` is interpreted as above.
int Set(int index, double value);
// Return the number of valid elements in the buffer.
int BufferLevel();
diff --git a/modules/audio_processing/vad/voice_activity_detector.cc b/modules/audio_processing/vad/voice_activity_detector.cc
index f0d34c6..ce4d46b 100644
--- a/modules/audio_processing/vad/voice_activity_detector.cc
+++ b/modules/audio_processing/vad/voice_activity_detector.cc
@@ -32,7 +32,7 @@
VoiceActivityDetector::~VoiceActivityDetector() = default;
// Because ISAC has a different chunk length, it updates
-// |chunkwise_voice_probabilities_| and |chunkwise_rms_| when there is new data.
+// `chunkwise_voice_probabilities_` and `chunkwise_rms_` when there is new data.
// Otherwise it clears them.
void VoiceActivityDetector::ProcessChunk(const int16_t* audio,
size_t length,
@@ -49,7 +49,7 @@
}
RTC_DCHECK_EQ(length, kLength10Ms);
- // Each chunk needs to be passed into |standalone_vad_|, because internally it
+ // Each chunk needs to be passed into `standalone_vad_`, because internally it
// buffers the audio and processes it all at once when GetActivity() is
// called.
RTC_CHECK_EQ(standalone_vad_->AddAudio(resampled_ptr, length), 0);
diff --git a/modules/audio_processing/vad/voice_activity_detector_unittest.cc b/modules/audio_processing/vad/voice_activity_detector_unittest.cc
index 3214bd9..80f21c8 100644
--- a/modules/audio_processing/vad/voice_activity_detector_unittest.cc
+++ b/modules/audio_processing/vad/voice_activity_detector_unittest.cc
@@ -133,7 +133,7 @@
vad.ProcessChunk(&data[0], data.size(), kSampleRateHz);
// Before the |vad has enough data to process an ISAC block it will return
- // the default value, 1.f, which would ruin the |max_probability| value.
+ // the default value, 1.f, which would ruin the `max_probability` value.
if (i > kNumChunksPerIsacBlock) {
max_probability = std::max(max_probability, vad.last_voice_probability());
}
@@ -156,7 +156,7 @@
vad.ProcessChunk(&data[0], data.size(), 2 * kSampleRateHz);
// Before the |vad has enough data to process an ISAC block it will return
- // the default value, 1.f, which would ruin the |max_probability| value.
+ // the default value, 1.f, which would ruin the `max_probability` value.
if (i > kNumChunksPerIsacBlock) {
max_probability = std::max(max_probability, vad.last_voice_probability());
}