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audio-resampler: Use n_phases when calculating taps offset

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Tweak linear interpolation oversampling.
Clear filter cache on rate changes when using a full filter.
This commit is contained in:
Wim Taymans 2016-02-16 09:18:13 +01:00
parent 524ea147cc
commit 167a415717
1 changed files with 12 additions and 6 deletions
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18
gst-libs/gst/audio/audio-resampler.c
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@ -509,25 +509,24 @@ fill_taps (GstAudioResampler * resampler,
gdouble x = 1.0 - n_taps / 2 - (gdouble) phase / n_phases; gdouble x = 1.0 - n_taps / 2 - (gdouble) phase / n_phases;
res = make_taps (resampler, tmp_taps, x, n_taps, 1); res = make_taps (resampler, tmp_taps, x, n_taps, 1);
} else { } else {
gint out_rate = resampler->out_rate;
gint offset, pos, frac; gint offset, pos, frac;
gint oversample = resampler->oversample; gint oversample = resampler->oversample;
gint taps_stride = resampler->taps_stride; gint taps_stride = resampler->taps_stride;
gdouble ic[4], *taps; gdouble ic[4], *taps;
pos = phase * oversample; pos = phase * oversample;
offset = (oversample - 1) - (pos / out_rate); offset = (oversample - 1) - (pos / n_phases);
frac = pos % out_rate; frac = pos % n_phases;
taps = (gdouble *) ((gint8 *) resampler->taps + offset * taps_stride); taps = (gdouble *) ((gint8 *) resampler->taps + offset * taps_stride);
switch (resampler->filter_interpolation) { switch (resampler->filter_interpolation) {
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR: case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR:
make_coeff_gdouble_linear (frac, out_rate, ic); make_coeff_gdouble_linear (frac, n_phases, ic);
interpolate_gdouble_linear (tmp_taps, taps, n_taps, ic); interpolate_gdouble_linear (tmp_taps, taps, n_taps, ic);
break; break;
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC: case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC:
make_coeff_gdouble_cubic (frac, out_rate, ic); make_coeff_gdouble_cubic (frac, n_phases, ic);
interpolate_gdouble_cubic (tmp_taps, taps, n_taps, ic); interpolate_gdouble_cubic (tmp_taps, taps, n_taps, ic);
break; break;
default: default:
@ -1172,7 +1171,7 @@ resampler_calculate_taps (GstAudioResampler * resampler)
switch (filter_interpolation) { switch (filter_interpolation) {
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR: case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR:
oversample <<= 5; oversample *= 11;
break; break;
default: default:
break; break;
@ -1218,9 +1217,11 @@ resampler_calculate_taps (GstAudioResampler * resampler)
switch (resampler->filter_interpolation) { switch (resampler->filter_interpolation) {
default: default:
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR: case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR:
GST_DEBUG ("using linear interpolation to build filter");
isize = 2; isize = 2;
break; break;
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC: case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC:
GST_DEBUG ("using cubic interpolation to build filter");
isize = 4; isize = 4;
break; break;
} }
@ -1621,6 +1622,11 @@ gst_audio_resampler_update (GstAudioResampler * resampler,
resampler->samples_avail += diff; resampler->samples_avail += diff;
} }
} else if (resampler->filter_mode == GST_AUDIO_RESAMPLER_FILTER_MODE_FULL) {
GST_DEBUG ("setting up filter cache");
resampler->n_phases = resampler->out_rate;
alloc_cache_mem (resampler, resampler->bps, resampler->n_taps,
resampler->n_phases);
} }
return TRUE; return TRUE;
} }