Bump to WebRTC M120 release
Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone. We're continuing to carry iSAC even though it's gone upstream, but maybe we'll want to drop that soon.
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@ -11,6 +11,8 @@
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#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
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#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
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#include <stddef.h>
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#include "modules/audio_coding/codecs/isac/main/source/structs.h"
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void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);
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@ -25,8 +25,8 @@
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* Post-filtering:
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* y(z) = x(z) - damper(z) * gain * (x(z) + y(z)) * z ^ (-lag);
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*
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* Note that |lag| is a floating number so we perform an interpolation to
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* obtain the correct |lag|.
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* Note that `lag` is a floating number so we perform an interpolation to
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* obtain the correct `lag`.
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*
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*/
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@ -86,7 +86,7 @@ typedef enum {
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* buffer : a buffer where the sum of previous inputs and outputs
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* are stored.
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* damper_state : the state of the damping filter. The filter is defined by
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* |kDampFilter|.
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* `kDampFilter`.
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* interpol_coeff : pointer to a set of coefficient which are used to utilize
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* fractional pitch by interpolation.
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* gain : pitch-gain to be applied to the current segment of input.
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@ -140,9 +140,9 @@ static void FilterSegment(const double* in_data, PitchFilterParam* parameters,
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int j;
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double sum;
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double sum2;
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/* Index of |parameters->buffer| where the output is written to. */
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/* Index of `parameters->buffer` where the output is written to. */
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int pos = parameters->index + PITCH_BUFFSIZE;
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/* Index of |parameters->buffer| where samples are read for fractional-lag
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/* Index of `parameters->buffer` where samples are read for fractional-lag
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* computation. */
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int pos_lag = pos - parameters->lag_offset;
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@ -174,9 +174,9 @@ static void FilterSegment(const double* in_data, PitchFilterParam* parameters,
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/* Filter for fractional pitch. */
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sum2 = 0.0;
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for (m = PITCH_FRACORDER-1; m >= m_tmp; --m) {
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/* |lag_index + m| is always larger than or equal to zero, see how
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/* `lag_index + m` is always larger than or equal to zero, see how
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* m_tmp is computed. This is equivalent to assume samples outside
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* |out_dg[j]| are zero. */
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* `out_dg[j]` are zero. */
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sum2 += out_dg[j][lag_index + m] * parameters->interpol_coeff[m];
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}
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/* Add the contribution of differential gain change. */
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@ -353,7 +353,7 @@ static void FilterFrame(const double* in_data, PitchFiltstr* filter_state,
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if ((mode == kPitchFilterPreGain) || (mode == kPitchFilterPreLa)) {
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/* Filter the lookahead segment, this is treated as the last sub-frame. So
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* set |pf_param| to last sub-frame. */
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* set `pf_param` to last sub-frame. */
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filter_parameters.sub_frame = PITCH_SUBFRAMES - 1;
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filter_parameters.num_samples = QLOOKAHEAD;
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FilterSegment(in_data, &filter_parameters, out_data, out_dg);
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