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

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Bug: webrtc:12338
Change-Id: I02613d9fca45d00e2477f334b7a0416e7912e26b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227037
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34621}
This commit is contained in:
Artem Titov
2021-07-28 20:00:17 +02:00
committed by WebRTC LUCI CQ
parent 0146a34b3f
commit d00ce747c7
143 changed files with 809 additions and 809 deletions
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38
modules/audio_coding/neteq/expand.cc
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@ -167,7 +167,7 @@ int Expand::Process(AudioMultiVector* output) {
}
// Smooth the expanded if it has not been muted to a low amplitude and
// |current_voice_mix_factor| is larger than 0.5.
// `current_voice_mix_factor` is larger than 0.5.
if ((parameters.mute_factor > 819) &&
(parameters.current_voice_mix_factor > 8192)) {
size_t start_ix = sync_buffer_->Size() - overlap_length_;
@ -197,7 +197,7 @@ int Expand::Process(AudioMultiVector* output) {
}
// Unvoiced part.
// Filter |scaled_random_vector| through |ar_filter_|.
// Filter `scaled_random_vector` through `ar_filter_`.
memcpy(unvoiced_vector - kUnvoicedLpcOrder, parameters.ar_filter_state,
sizeof(int16_t) * kUnvoicedLpcOrder);
int32_t add_constant = 0;
@ -402,7 +402,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
// Calculate correlation in downsampled domain (4 kHz sample rate).
size_t correlation_length = 51; // TODO(hlundin): Legacy bit-exactness.
// If it is decided to break bit-exactness |correlation_length| should be
// If it is decided to break bit-exactness `correlation_length` should be
// initialized to the return value of Correlation().
Correlation(audio_history.get(), signal_length, correlation_vector);
@ -417,7 +417,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
best_correlation_index[1] += fs_mult_20;
best_correlation_index[2] += fs_mult_20;
// Calculate distortion around the |kNumCorrelationCandidates| best lags.
// Calculate distortion around the `kNumCorrelationCandidates` best lags.
int distortion_scale = 0;
for (size_t i = 0; i < kNumCorrelationCandidates; i++) {
size_t min_index =
@ -434,7 +434,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
WebRtcSpl_VectorBitShiftW32ToW16(best_distortion, kNumCorrelationCandidates,
best_distortion_w32, distortion_scale);
// Find the maximizing index |i| of the cost function
// Find the maximizing index `i` of the cost function
// f[i] = best_correlation[i] / best_distortion[i].
int32_t best_ratio = std::numeric_limits<int32_t>::min();
size_t best_index = std::numeric_limits<size_t>::max();
@ -458,7 +458,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
max_lag_ = std::max(distortion_lag, correlation_lag);
// Calculate the exact best correlation in the range between
// |correlation_lag| and |distortion_lag|.
// `correlation_lag` and `distortion_lag`.
correlation_length = std::max(std::min(distortion_lag + 10, fs_mult_120),
static_cast<size_t>(60 * fs_mult));
@ -487,7 +487,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
(31 - WebRtcSpl_NormW32(static_cast<int32_t>(correlation_length))) - 31;
correlation_scale = std::max(0, correlation_scale);
// Calculate the correlation, store in |correlation_vector2|.
// Calculate the correlation, store in `correlation_vector2`.
WebRtcSpl_CrossCorrelation(
correlation_vector2,
&(audio_history[signal_length - correlation_length]),
@ -537,7 +537,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
}
// Extract the two vectors expand_vector0 and expand_vector1 from
// |audio_history|.
// `audio_history`.
size_t expansion_length = max_lag_ + overlap_length_;
const int16_t* vector1 = &(audio_history[signal_length - expansion_length]);
const int16_t* vector2 = vector1 - distortion_lag;
@ -594,13 +594,13 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
expand_lags_[1] = distortion_lag;
expand_lags_[2] = distortion_lag;
} else {
// |distortion_lag| and |correlation_lag| are not equal; use different
// `distortion_lag` and `correlation_lag` are not equal; use different
// combinations of the two.
// First lag is |distortion_lag| only.
// First lag is `distortion_lag` only.
expand_lags_[0] = distortion_lag;
// Second lag is the average of the two.
expand_lags_[1] = (distortion_lag + correlation_lag) / 2;
// Third lag is the average again, but rounding towards |correlation_lag|.
// Third lag is the average again, but rounding towards `correlation_lag`.
if (distortion_lag > correlation_lag) {
expand_lags_[2] = (distortion_lag + correlation_lag - 1) / 2;
} else {
@ -638,7 +638,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
if (stability != 1) {
// Set first coefficient to 4096 (1.0 in Q12).
parameters.ar_filter[0] = 4096;
// Set remaining |kUnvoicedLpcOrder| coefficients to zero.
// Set remaining `kUnvoicedLpcOrder` coefficients to zero.
WebRtcSpl_MemSetW16(parameters.ar_filter + 1, 0, kUnvoicedLpcOrder);
}
}
@ -656,7 +656,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
sizeof(int16_t) * noise_length);
} else {
// Only applies to SWB where length could be larger than
// |kRandomTableSize|.
// `kRandomTableSize`.
memcpy(random_vector, RandomVector::kRandomTable,
sizeof(int16_t) * RandomVector::kRandomTableSize);
RTC_DCHECK_LE(noise_length, kMaxSampleRate / 8000 * 120 + 30);
@ -694,7 +694,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
int32_t unvoiced_energy = WebRtcSpl_DotProductWithScale(
unvoiced_vector, unvoiced_vector, 128, unvoiced_prescale);
// Normalize |unvoiced_energy| to 28 or 29 bits to preserve sqrt() accuracy.
// Normalize `unvoiced_energy` to 28 or 29 bits to preserve sqrt() accuracy.
int16_t unvoiced_scale = WebRtcSpl_NormW32(unvoiced_energy) - 3;
// Make sure we do an odd number of shifts since we already have 7 shifts
// from dividing with 128 earlier. This will make the total scale factor
@ -715,7 +715,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
// voice_mix_factor = 0;
if (corr_coefficient > 7875) {
int16_t x1, x2, x3;
// |corr_coefficient| is in Q14.
// `corr_coefficient` is in Q14.
x1 = static_cast<int16_t>(corr_coefficient);
x2 = (x1 * x1) >> 14; // Shift 14 to keep result in Q14.
x3 = (x1 * x2) >> 14;
@ -733,13 +733,13 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
}
// Calculate muting slope. Reuse value from earlier scaling of
// |expand_vector0| and |expand_vector1|.
// `expand_vector0` and `expand_vector1`.
int16_t slope = amplitude_ratio;
if (slope > 12288) {
// slope > 1.5.
// Calculate (1 - (1 / slope)) / distortion_lag =
// (slope - 1) / (distortion_lag * slope).
// |slope| is in Q13, so 1 corresponds to 8192. Shift up to Q25 before
// `slope` is in Q13, so 1 corresponds to 8192. Shift up to Q25 before
// the division.
// Shift the denominator from Q13 to Q5 before the division. The result of
// the division will then be in Q20.
@ -757,7 +757,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
parameters.onset = true;
} else {
// Calculate (1 - slope) / distortion_lag.
// Shift |slope| by 7 to Q20 before the division. The result is in Q20.
// Shift `slope` by 7 to Q20 before the division. The result is in Q20.
parameters.mute_slope = WebRtcSpl_DivW32W16(
(8192 - slope) * 128, static_cast<int16_t>(distortion_lag));
if (parameters.voice_mix_factor <= 13107) {
@ -826,7 +826,7 @@ void Expand::Correlation(const int16_t* input,
kDownsampledLength, filter_coefficients, num_coefficients,
downsampling_factor, kFilterDelay);
// Normalize |downsampled_input| to using all 16 bits.
// Normalize `downsampled_input` to using all 16 bits.
int16_t max_value =
WebRtcSpl_MaxAbsValueW16(downsampled_input, kDownsampledLength);
int16_t norm_shift = 16 - WebRtcSpl_NormW32(max_value);