From: Wim Taymans Date: Thu, 4 Feb 2016 14:19:53 +0000 (+0100) Subject: audio-resampler: improve tap calculation X-Git-Tag: 1.19.3~511^2~2963 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=13e5b986cd09fe81b7cd827cfb3af52f210c77cc;p=platform%2Fupstream%2Fgstreamer.git audio-resampler: improve tap calculation Return the taps from make_taps, this makes it possible to not actually have to cache the taps when we want to. Fix overflow in phase calculation. --- diff --git a/gst-libs/gst/audio/audio-resampler.c b/gst-libs/gst/audio/audio-resampler.c index 1fb956a..121e2a7 100644 --- a/gst-libs/gst/audio/audio-resampler.c +++ b/gst-libs/gst/audio/audio-resampler.c @@ -259,7 +259,7 @@ get_kaiser_tap (GstAudioResampler * resampler, gdouble x) #define CONVERT_TAPS(type, precision) \ G_STMT_START { \ - type *taps = res = t->taps = (type *) ((gint8*)resampler->coeff + j * resampler->cstride); \ + type *taps = res = (type *) ((gint8*)resampler->coeff + j * resampler->cstride); \ gdouble multiplier = (1 << precision); \ gint i, j; \ gdouble offset, l_offset, h_offset; \ @@ -304,6 +304,7 @@ make_taps (GstAudioResampler * resampler, Tap * t, gint j) gdouble x, weight = 0.0; gdouble *tmpcoeff = resampler->tmpcoeff; gint tap_offs = n_taps / 2; + gint in_rate = resampler->in_rate; gint out_rate = resampler->out_rate; gint l; @@ -342,7 +343,7 @@ make_taps (GstAudioResampler * resampler, Tap * t, gint j) switch (resampler->format) { case GST_AUDIO_FORMAT_F64: { - gdouble *taps = res = t->taps = + gdouble *taps = res = (gdouble *) ((gint8 *) resampler->coeff + j * resampler->cstride); for (l = 0; l < n_taps; l++) taps[l] = tmpcoeff[l] / weight; @@ -350,7 +351,7 @@ make_taps (GstAudioResampler * resampler, Tap * t, gint j) } case GST_AUDIO_FORMAT_F32: { - gfloat *taps = res = t->taps = + gfloat *taps = res = (gfloat *) ((gint8 *) resampler->coeff + j * resampler->cstride); for (l = 0; l < n_taps; l++) taps[l] = tmpcoeff[l] / weight; @@ -366,6 +367,11 @@ make_taps (GstAudioResampler * resampler, Tap * t, gint j) g_assert_not_reached (); break; } + if (t) { + t->taps = res; + t->sample_inc = (j + in_rate) / out_rate; + t->next_phase = (j + in_rate) % out_rate; + } return res; } @@ -662,7 +668,8 @@ resampler_calculate_taps (GstAudioResampler * resampler) if (out_rate < in_rate) { resampler->cutoff = resampler->cutoff * out_rate / in_rate; - resampler->n_taps = resampler->n_taps * in_rate / out_rate; + resampler->n_taps = + gst_util_uint64_scale_int (resampler->n_taps, in_rate, out_rate); } /* only round up for bigger taps, the small taps are used for nearest, * linear and cubic and we want to use less taps for those. */ @@ -690,8 +697,6 @@ resampler_calculate_taps (GstAudioResampler * resampler) for (j = 0; j < out_rate; j++) { Tap *t = &resampler->taps[j]; t->taps = NULL; - t->sample_inc = (j + in_rate) / out_rate; - t->next_phase = (j + in_rate) % out_rate; } non_interleaved = @@ -1253,7 +1258,7 @@ gst_audio_resampler_resample (GstAudioResampler * resampler, resampler->resample (resampler, sbuf, samples_avail, out, out_frames, &consumed); - GST_LOG ("in %" G_GSIZE_FORMAT ", used %" G_GSIZE_FORMAT ", consumed %" + GST_LOG ("in %" G_GSIZE_FORMAT ", avail %" G_GSIZE_FORMAT ", consumed %" G_GSIZE_FORMAT, in_frames, samples_avail, consumed); /* update pointers */