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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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audio-resample: small optimizations
Remove some inline functions that are called in the slow path. Unroll C fallback functions a little.
This commit is contained in:
Wim Taymans
parent
167a415717
commit
ea497b509f
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@ -31,6 +31,26 @@
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#include "audio-resampler.h"
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/**
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* Contains a collection of all things found in other resamplers:
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* speex (optimizations), ffmpeg (fixed phase filter, blackman filter),
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* SRC (linear interpolation, fixed precomputed tables),...
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*
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* Supports:
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* - S16, S32, F32 and F64 formats
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* - linear interpolation
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* - cubic interpolation
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* - sinc based interpolation with kaiser or blackman-nutall windows
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* - fully configurable kaiser parameters
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* - dynamic linear or cubic interpolation of filter table, this can
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* use less memory but more CPU
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* - full filter table, generated from optionally linear or cubic
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* interpolation of filter table
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* - fixed filter table size with nearest neighbour phase, optionally
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* using a precomputed table
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* - dynamic samplerate changes
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* - x86 optimizations
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*/
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typedef void (*ResampleFunc) (GstAudioResampler * resampler, gpointer in[],
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gsize in_len, gpointer out[], gsize out_len, gsize * consumed);
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typedef void (*DeinterleaveFunc) (GstAudioResampler * resampler,
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@ -67,13 +87,13 @@ struct _GstAudioResampler
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/* oversampled main filter table */
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gint oversample;
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guint n_taps;
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gint n_taps;
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gpointer taps;
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gpointer taps_mem;
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gsize taps_stride;
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gint n_phases;
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guint alloc_taps;
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guint alloc_phases;
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gint alloc_taps;
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gint alloc_phases;
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/* cached taps */
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gpointer *cached_phases;
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@ -84,8 +104,8 @@ struct _GstAudioResampler
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DeinterleaveFunc deinterleave;
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ResampleFunc resample;
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guint blocks;
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guint inc;
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gint blocks;
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gint inc;
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gint samp_inc;
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gint samp_frac;
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gint samp_index;
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@ -138,7 +158,7 @@ static const KaiserQualityMap kaiser_qualities[] = {
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typedef struct
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{
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guint n_taps;
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gint n_taps;
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gdouble cutoff;
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} BlackmanQualityMap;
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@ -294,7 +314,7 @@ get_kaiser_tap (gdouble x, gint n_taps, gdouble Fc, gdouble beta)
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#define PRECISION_S16 15
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#define PRECISION_S32 31
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static inline gdouble
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static gdouble
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make_taps (GstAudioResampler * resampler,
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gdouble * tmp_taps, gdouble x, gint n_taps, gint oversample)
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{
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@ -339,11 +359,11 @@ make_taps (GstAudioResampler * resampler,
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}
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#define MAKE_CONVERT_TAPS_INT_FUNC(type, precision) \
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static inline void \
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convert_taps_##type (gdouble *tmp_taps, type *taps, \
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static void \
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convert_taps_##type (gdouble *tmp_taps, type *taps, \
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gdouble weight, gint n_taps) \
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{ \
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gint64 one = (1L << precision) - 1; \
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gint64 one = (1LL << precision) - 1; \
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gdouble multiplier = one; \
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gint i, j; \
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gdouble offset, l_offset, h_offset; \
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@ -356,7 +376,7 @@ convert_taps_##type (gdouble *tmp_taps, type *taps, \
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for (i = 0; i < 32; i++) { \
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gint64 sum = 0; \
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for (j = 0; j < n_taps; j++) \
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sum += floor (offset + tmp_taps[j] * multiplier / weight); \
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sum += floor (offset + tmp_taps[j] * multiplier / weight); \
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if (sum == one) { \
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exact = TRUE; \
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break; \
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@ -374,19 +394,19 @@ convert_taps_##type (gdouble *tmp_taps, type *taps, \
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} \
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} \
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for (j = 0; j < n_taps; j++) \
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taps[j] = floor (offset + tmp_taps[j] * multiplier / weight); \
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taps[j] = floor (offset + tmp_taps[j] * multiplier / weight); \
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if (!exact) \
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GST_WARNING ("can't find exact taps"); \
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}
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#define MAKE_CONVERT_TAPS_FLOAT_FUNC(type) \
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static inline void \
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convert_taps_##type (gdouble *tmp_taps, type *taps, \
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static void \
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convert_taps_##type (gdouble *tmp_taps, type *taps, \
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gdouble weight, gint n_taps) \
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{ \
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gint i; \
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for (i = 0; i < n_taps; i++) \
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taps[i] = tmp_taps[i] / weight; \
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taps[i] = tmp_taps[i] / weight; \
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}
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MAKE_CONVERT_TAPS_INT_FUNC (gint16, PRECISION_S16);
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@ -399,10 +419,10 @@ static inline void \
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extract_taps_##type (GstAudioResampler * resampler, type *tmp_taps, \
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gint n_taps, gint oversample, gint mult) \
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{ \
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gint i, j, k; \
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for (i = 0; i < oversample; i++) { \
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gint i, j, k, o = oversample - 1; \
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for (i = 0; i < oversample; i++, o--) { \
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type *taps = (type *) ((gint8*)resampler->taps + \
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i * resampler->taps_stride); \
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o * resampler->taps_stride); \
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for (j = 0; j < n_taps; j++) { \
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for (k = 0; k < mult; k++) { \
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*taps++ = tmp_taps[i + j*oversample + k]; \
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@ -454,7 +474,7 @@ make_coeff_##type##_linear (gint frac, gint out_rate, type *icoeff) \
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{ \
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type x = ((gint64)frac << prec) / out_rate; \
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icoeff[0] = icoeff[2] = x; \
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icoeff[1] = icoeff[3] = (1L << prec) -1 - x; \
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icoeff[1] = icoeff[3] = (1LL << prec) -1 - x; \
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}
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#define MAKE_COEFF_LINEAR_FLOAT_FUNC(type) \
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static inline void \
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@ -462,7 +482,7 @@ make_coeff_##type##_linear (gint frac, gint out_rate, type *icoeff) \
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{ \
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type x = (type)frac / out_rate; \
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icoeff[0] = icoeff[2] = x; \
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icoeff[1] = icoeff[3] = 1.0 - x; \
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icoeff[1] = icoeff[3] = (type)1.0 - x; \
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}
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MAKE_COEFF_LINEAR_INT_FUNC (gint16, PRECISION_S16);
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MAKE_COEFF_LINEAR_INT_FUNC (gint32, PRECISION_S32);
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@ -473,7 +493,7 @@ MAKE_COEFF_LINEAR_FLOAT_FUNC (gdouble);
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static inline void \
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make_coeff_##type##_cubic (gint frac, gint out_rate, type *icoeff) \
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{ \
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type2 one = (1L << prec) - 1; \
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type2 one = (1LL << prec) - 1; \
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type2 x = ((gint64) frac << prec) / out_rate; \
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type2 x2 = (x * x) >> prec; \
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type2 x3 = (x2 * x) >> prec; \
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@ -491,14 +511,14 @@ make_coeff_##type##_cubic (gint frac, gint out_rate, type *icoeff) \
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icoeff[0] = 0.16667f * (x3 - x); \
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icoeff[1] = x + 0.5f * (x2 - x3); \
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icoeff[3] = -0.33333f * x + 0.5f * x2 - 0.16667f * x3; \
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icoeff[2] = 1. - icoeff[0] - icoeff[1] - icoeff[3]; \
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icoeff[2] = (type)1.0 - icoeff[0] - icoeff[1] - icoeff[3]; \
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}
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MAKE_COEFF_CUBIC_INT_FUNC (gint16, gint32, PRECISION_S16);
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MAKE_COEFF_CUBIC_INT_FUNC (gint32, gint64, PRECISION_S32);
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MAKE_COEFF_CUBIC_FLOAT_FUNC (gfloat);
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MAKE_COEFF_CUBIC_FLOAT_FUNC (gdouble);
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static inline gdouble
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static gdouble
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fill_taps (GstAudioResampler * resampler,
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gdouble * tmp_taps, gint phase, gint n_phases, gint n_taps)
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{
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@ -515,7 +535,7 @@ fill_taps (GstAudioResampler * resampler,
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gdouble ic[4], *taps;
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pos = phase * oversample;
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offset = (oversample - 1) - (pos / n_phases);
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offset = pos / n_phases;
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frac = pos % n_phases;
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taps = (gdouble *) ((gint8 *) resampler->taps + offset * taps_stride);
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@ -587,7 +607,7 @@ get_taps_##type##_##inter (GstAudioResampler * resampler, \
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gint taps_stride = resampler->taps_stride; \
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\
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pos = *samp_phase * oversample; \
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offset = (oversample - 1) - (pos / out_rate); \
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offset = pos / out_rate; \
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frac = pos % out_rate; \
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\
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res = (gint8 *) resampler->taps + offset * taps_stride; \
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@ -620,15 +640,16 @@ inner_product_##type##_none_1_c (type * o, const type * a, \
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gint i; \
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type2 res = 0; \
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\
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for (i = 0; i < len; i++) \
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res += (type2) a[i] * (type2) b[i]; \
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\
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res = (res + (1L << ((prec) - 1))) >> (prec); \
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for (i = 0; i < len; i += 2) { \
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res += (type2) a[2*i+0] * (type2) b[2*i+0]; \
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res += (type2) a[2*i+1] * (type2) b[2*i+1]; \
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} \
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res = (res + (1LL << ((prec) - 1))) >> (prec); \
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*o = CLAMP (res, -(limit), (limit) - 1); \
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}
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INNER_PRODUCT_INT_NONE_FUNC (gint16, gint32, PRECISION_S16, 1L << 15);
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INNER_PRODUCT_INT_NONE_FUNC (gint32, gint64, PRECISION_S32, 1L << 31);
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INNER_PRODUCT_INT_NONE_FUNC (gint16, gint32, PRECISION_S16, 1LL << 15);
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INNER_PRODUCT_INT_NONE_FUNC (gint32, gint64, PRECISION_S32, 1LL << 31);
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#define INNER_PRODUCT_INT_LINEAR_FUNC(type,type2,prec,limit) \
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static inline void \
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@ -644,12 +665,12 @@ inner_product_##type##_linear_1_c (type * o, const type * a, \
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} \
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res[0] = (res[0] >> (prec)) * (type2) ic[0] + \
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(res[1] >> (prec)) * (type2) ic[1]; \
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res[0] = (res[0] + (1L << ((prec) - 1))) >> (prec); \
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res[0] = (res[0] + (1LL << ((prec) - 1))) >> (prec); \
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*o = CLAMP (res[0], -(limit), (limit) - 1); \
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}
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INNER_PRODUCT_INT_LINEAR_FUNC (gint16, gint32, PRECISION_S16, 1L << 15);
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INNER_PRODUCT_INT_LINEAR_FUNC (gint32, gint64, PRECISION_S32, 1L << 31);
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INNER_PRODUCT_INT_LINEAR_FUNC (gint16, gint32, PRECISION_S16, 1LL << 15);
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INNER_PRODUCT_INT_LINEAR_FUNC (gint32, gint64, PRECISION_S32, 1LL << 31);
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#define INNER_PRODUCT_INT_CUBIC_FUNC(type,type2,prec,limit) \
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static inline void \
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@ -669,12 +690,12 @@ inner_product_##type##_cubic_1_c (type * o, const type * a, \
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(res[1] >> (prec)) * (type2) ic[1] + \
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(res[2] >> (prec)) * (type2) ic[2] + \
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(res[3] >> (prec)) * (type2) ic[3]; \
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res[0] = (res[0] + (1L << ((prec) - 1))) >> (prec); \
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res[0] = (res[0] + (1LL << ((prec) - 1))) >> (prec); \
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*o = CLAMP (res[0], -(limit), (limit) - 1); \
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}
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INNER_PRODUCT_INT_CUBIC_FUNC (gint16, gint32, PRECISION_S16, 1L << 15);
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INNER_PRODUCT_INT_CUBIC_FUNC (gint32, gint64, PRECISION_S32, 1L << 31);
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INNER_PRODUCT_INT_CUBIC_FUNC (gint16, gint32, PRECISION_S16, 1LL << 15);
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INNER_PRODUCT_INT_CUBIC_FUNC (gint32, gint64, PRECISION_S32, 1LL << 31);
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#define INNER_PRODUCT_FLOAT_NONE_FUNC(type) \
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static inline void \
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@ -684,9 +705,10 @@ inner_product_##type##_none_1_c (type * o, const type * a, \
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gint i; \
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type res = 0.0; \
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\
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for (i = 0; i < len; i++) \
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res += a[i] * b[i]; \
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\
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for (i = 0; i < len; i += 2) { \
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res += a[2 * i + 0] * b[2 * i + 0]; \
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res += a[2 * i + 1] * b[2 * i + 1]; \
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} \
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*o = res; \
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}
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@ -885,7 +907,7 @@ static void \
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deinterleave_ ##type (GstAudioResampler * resampler, gpointer sbuf[], \
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gpointer in[], gsize in_frames) \
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{ \
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guint i, c, channels = resampler->channels; \
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gint i, c, channels = resampler->channels; \
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gsize samples_avail = resampler->samples_avail; \
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for (c = 0; c < channels; c++) { \
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type *s = (type *) sbuf[c] + samples_avail; \
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@ -916,7 +938,7 @@ static void
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deinterleave_copy (GstAudioResampler * resampler, gpointer sbuf[],
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gpointer in[], gsize in_frames)
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{
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guint c, blocks = resampler->blocks;
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gint c, blocks = resampler->blocks;
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gsize bytes_avail, in_bytes, bpf;
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bpf = resampler->bps * resampler->inc;
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@ -925,9 +947,9 @@ deinterleave_copy (GstAudioResampler * resampler, gpointer sbuf[],
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for (c = 0; c < blocks; c++) {
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if (G_UNLIKELY (in == NULL))
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memset ((guint8 *) sbuf[c] + bytes_avail, 0, in_bytes);
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memset ((gint8 *) sbuf[c] + bytes_avail, 0, in_bytes);
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else
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memcpy ((guint8 *) sbuf[c] + bytes_avail, in[c], in_bytes);
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memcpy ((gint8 *) sbuf[c] + bytes_avail, in[c], in_bytes);
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}
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}
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@ -1448,7 +1470,7 @@ static inline gpointer *
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get_sample_bufs (GstAudioResampler * resampler, gsize need)
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{
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if (G_LIKELY (resampler->samples_len < need)) {
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guint c, blocks = resampler->blocks;
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gint c, blocks = resampler->blocks;
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gsize bytes, to_move = 0;
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gint8 *ptr, *samples;
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@ -1654,23 +1676,6 @@ gst_audio_resampler_free (GstAudioResampler * resampler)
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g_slice_free (GstAudioResampler, resampler);
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}
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static inline gsize
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calc_out (GstAudioResampler * resampler, gsize in)
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{
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gsize out;
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out = in * resampler->out_rate;
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if (out < resampler->samp_phase)
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return 0;
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out = ((out - resampler->samp_phase) / resampler->in_rate) + 1;
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GST_LOG ("out %d = ((%d * %d - %d) / %d) + 1", (gint) out,
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(gint) in, resampler->out_rate, resampler->samp_phase,
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resampler->in_rate);
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return out;
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}
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/**
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* gst_audio_resampler_get_out_frames:
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* @resampler: a #GstAudioResampler
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@ -1686,7 +1691,7 @@ gsize
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gst_audio_resampler_get_out_frames (GstAudioResampler * resampler,
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gsize in_frames)
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{
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gsize need, avail;
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gsize need, avail, out;
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g_return_val_if_fail (resampler != NULL, 0);
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@ -1698,7 +1703,16 @@ gst_audio_resampler_get_out_frames (GstAudioResampler * resampler,
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if (avail < need)
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return 0;
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return calc_out (resampler, avail - need);
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out = (avail - need) * resampler->out_rate;
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if (out < resampler->samp_phase)
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return 0;
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out = ((out - resampler->samp_phase) / resampler->in_rate) + 1;
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GST_LOG ("out %d = ((%d * %d - %d) / %d) + 1", (gint) out,
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(gint) (avail - need), resampler->out_rate, resampler->samp_phase,
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resampler->in_rate);
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return out;
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}
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/**
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