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