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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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416 lines
11 KiB
C
416 lines
11 KiB
C
/*
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* GStreamer
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* Copyright (C) 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <gst/gst.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/gstaudiofilter.h>
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#include <math.h>
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#include "audiofxbaseiirfilter.h"
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#define GST_CAT_DEFAULT gst_audio_fx_base_iir_filter_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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#define ALLOWED_CAPS \
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"audio/x-raw," \
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" format=(string){"GST_AUDIO_NE(F32)","GST_AUDIO_NE(F64)"}," \
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" rate = (int) [ 1, MAX ]," \
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" channels = (int) [ 1, MAX ]," \
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" layout=(string) interleaved"
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#define gst_audio_fx_base_iir_filter_parent_class parent_class
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G_DEFINE_TYPE (GstAudioFXBaseIIRFilter,
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gst_audio_fx_base_iir_filter, GST_TYPE_AUDIO_FILTER);
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static gboolean gst_audio_fx_base_iir_filter_setup (GstAudioFilter * filter,
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const GstAudioInfo * info);
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static GstFlowReturn
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gst_audio_fx_base_iir_filter_transform_ip (GstBaseTransform * base,
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GstBuffer * buf);
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static gboolean gst_audio_fx_base_iir_filter_stop (GstBaseTransform * base);
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static void process_64 (GstAudioFXBaseIIRFilter * filter,
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gdouble * data, guint num_samples);
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static void process_32 (GstAudioFXBaseIIRFilter * filter,
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gfloat * data, guint num_samples);
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/* GObject vmethod implementations */
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static void
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gst_audio_fx_base_iir_filter_finalize (GObject * object)
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{
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GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (object);
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if (filter->a) {
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g_free (filter->a);
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filter->a = NULL;
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}
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if (filter->b) {
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g_free (filter->b);
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filter->b = NULL;
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}
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if (filter->channels) {
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GstAudioFXBaseIIRFilterChannelCtx *ctx;
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guint i;
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for (i = 0; i < filter->nchannels; i++) {
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ctx = &filter->channels[i];
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g_free (ctx->x);
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g_free (ctx->y);
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}
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g_free (filter->channels);
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filter->channels = NULL;
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}
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g_mutex_clear (&filter->lock);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_audio_fx_base_iir_filter_class_init (GstAudioFXBaseIIRFilterClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass;
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GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
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GstCaps *caps;
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GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_iir_filter_debug,
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"audiofxbaseiirfilter", 0, "Audio IIR Filter Base Class");
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gobject_class->finalize = gst_audio_fx_base_iir_filter_finalize;
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caps = gst_caps_from_string (ALLOWED_CAPS);
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gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
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caps);
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gst_caps_unref (caps);
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filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_fx_base_iir_filter_setup);
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trans_class->transform_ip =
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GST_DEBUG_FUNCPTR (gst_audio_fx_base_iir_filter_transform_ip);
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trans_class->transform_ip_on_passthrough = FALSE;
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trans_class->stop = GST_DEBUG_FUNCPTR (gst_audio_fx_base_iir_filter_stop);
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}
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static void
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gst_audio_fx_base_iir_filter_init (GstAudioFXBaseIIRFilter * filter)
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{
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gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
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filter->a = NULL;
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filter->na = 0;
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filter->b = NULL;
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filter->nb = 0;
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filter->channels = NULL;
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filter->nchannels = 0;
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g_mutex_init (&filter->lock);
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}
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/* Evaluate the transfer function that corresponds to the IIR
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* coefficients at (zr + zi*I)^-1 and return the magnitude */
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gdouble
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gst_audio_fx_base_iir_filter_calculate_gain (gdouble * a, guint na, gdouble * b,
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guint nb, gdouble zr, gdouble zi)
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{
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gdouble sum_ar, sum_ai;
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gdouble sum_br, sum_bi;
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gdouble gain_r, gain_i;
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gdouble sum_r_old;
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gdouble sum_i_old;
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gint i;
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sum_ar = a[na - 1];
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sum_ai = 0.0;
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for (i = na - 2; i >= 0; i--) {
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sum_r_old = sum_ar;
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sum_i_old = sum_ai;
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sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
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sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
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}
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sum_br = b[nb - 1];
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sum_bi = 0.0;
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for (i = nb - 2; i >= 0; i--) {
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sum_r_old = sum_br;
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sum_i_old = sum_bi;
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sum_br = (sum_r_old * zr - sum_i_old * zi) + b[i];
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sum_bi = (sum_r_old * zi + sum_i_old * zr) + 0.0;
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}
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gain_r =
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(sum_br * sum_ar + sum_bi * sum_ai) / (sum_ar * sum_ar + sum_ai * sum_ai);
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gain_i =
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(sum_bi * sum_ar - sum_br * sum_ai) / (sum_ar * sum_ar + sum_ai * sum_ai);
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return (sqrt (gain_r * gain_r + gain_i * gain_i));
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}
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void
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gst_audio_fx_base_iir_filter_set_coefficients (GstAudioFXBaseIIRFilter * filter,
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gdouble * a, guint na, gdouble * b, guint nb)
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{
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guint i;
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g_return_if_fail (GST_IS_AUDIO_FX_BASE_IIR_FILTER (filter));
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g_mutex_lock (&filter->lock);
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g_free (filter->a);
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g_free (filter->b);
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filter->a = filter->b = NULL;
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if (filter->channels) {
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GstAudioFXBaseIIRFilterChannelCtx *ctx;
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gboolean free = (na != filter->na || nb != filter->nb);
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for (i = 0; i < filter->nchannels; i++) {
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ctx = &filter->channels[i];
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if (free)
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g_free (ctx->x);
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else
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memset (ctx->x, 0, filter->nb * sizeof (gdouble));
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if (free)
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g_free (ctx->y);
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else
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memset (ctx->y, 0, filter->na * sizeof (gdouble));
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}
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g_free (filter->channels);
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filter->channels = NULL;
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}
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filter->na = na;
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filter->nb = nb;
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filter->a = a;
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filter->b = b;
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if (filter->nchannels && !filter->channels) {
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GstAudioFXBaseIIRFilterChannelCtx *ctx;
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filter->channels =
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g_new0 (GstAudioFXBaseIIRFilterChannelCtx, filter->nchannels);
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for (i = 0; i < filter->nchannels; i++) {
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ctx = &filter->channels[i];
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ctx->x = g_new0 (gdouble, filter->nb);
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ctx->y = g_new0 (gdouble, filter->na);
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}
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}
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g_mutex_unlock (&filter->lock);
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}
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/* GstAudioFilter vmethod implementations */
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static gboolean
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gst_audio_fx_base_iir_filter_setup (GstAudioFilter * base,
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const GstAudioInfo * info)
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{
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GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (base);
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gboolean ret = TRUE;
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gint channels;
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g_mutex_lock (&filter->lock);
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switch (GST_AUDIO_INFO_FORMAT (info)) {
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case GST_AUDIO_FORMAT_F32:
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filter->process = (GstAudioFXBaseIIRFilterProcessFunc)
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process_32;
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break;
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case GST_AUDIO_FORMAT_F64:
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filter->process = (GstAudioFXBaseIIRFilterProcessFunc)
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process_64;
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break;
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default:
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ret = FALSE;
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break;
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}
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channels = GST_AUDIO_INFO_CHANNELS (info);
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if (channels != filter->nchannels) {
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guint i;
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GstAudioFXBaseIIRFilterChannelCtx *ctx;
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if (filter->channels) {
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for (i = 0; i < filter->nchannels; i++) {
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ctx = &filter->channels[i];
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g_free (ctx->x);
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g_free (ctx->y);
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}
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g_free (filter->channels);
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}
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filter->channels = g_new0 (GstAudioFXBaseIIRFilterChannelCtx, channels);
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for (i = 0; i < channels; i++) {
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ctx = &filter->channels[i];
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ctx->x = g_new0 (gdouble, filter->nb);
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ctx->y = g_new0 (gdouble, filter->na);
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}
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filter->nchannels = channels;
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}
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g_mutex_unlock (&filter->lock);
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return ret;
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}
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static inline gdouble
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process (GstAudioFXBaseIIRFilter * filter,
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GstAudioFXBaseIIRFilterChannelCtx * ctx, gdouble x0)
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{
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gdouble val = filter->b[0] * x0;
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gint i, j;
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for (i = 1, j = ctx->x_pos; i < filter->nb; i++) {
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val += filter->b[i] * ctx->x[j];
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j--;
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if (j < 0)
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j = filter->nb - 1;
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}
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for (i = 1, j = ctx->y_pos; i < filter->na; i++) {
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val -= filter->a[i] * ctx->y[j];
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j--;
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if (j < 0)
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j = filter->na - 1;
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}
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val /= filter->a[0];
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if (ctx->x) {
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ctx->x_pos++;
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if (ctx->x_pos >= filter->nb)
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ctx->x_pos = 0;
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ctx->x[ctx->x_pos] = x0;
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}
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if (ctx->y) {
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ctx->y_pos++;
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if (ctx->y_pos >= filter->na)
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ctx->y_pos = 0;
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ctx->y[ctx->y_pos] = val;
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}
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return val;
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}
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#define DEFINE_PROCESS_FUNC(width,ctype) \
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static void \
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process_##width (GstAudioFXBaseIIRFilter * filter, \
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g##ctype * data, guint num_samples) \
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{ \
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gint i, j, channels = filter->nchannels; \
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gdouble val; \
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\
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for (i = 0; i < num_samples / channels; i++) { \
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for (j = 0; j < channels; j++) { \
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val = process (filter, &filter->channels[j], *data); \
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*data++ = val; \
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} \
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} \
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}
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DEFINE_PROCESS_FUNC (32, float);
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DEFINE_PROCESS_FUNC (64, double);
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#undef DEFINE_PROCESS_FUNC
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/* GstBaseTransform vmethod implementations */
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static GstFlowReturn
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gst_audio_fx_base_iir_filter_transform_ip (GstBaseTransform * base,
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GstBuffer * buf)
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{
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GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (base);
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guint num_samples;
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GstClockTime timestamp, stream_time;
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GstMapInfo map;
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timestamp = GST_BUFFER_TIMESTAMP (buf);
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stream_time =
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gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
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GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
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GST_TIME_ARGS (timestamp));
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if (GST_CLOCK_TIME_IS_VALID (stream_time))
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gst_object_sync_values (GST_OBJECT (filter), stream_time);
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gst_buffer_map (buf, &map, GST_MAP_READWRITE);
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num_samples = map.size / GST_AUDIO_FILTER_BPS (filter);
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g_mutex_lock (&filter->lock);
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if (filter->a == NULL || filter->b == NULL) {
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g_warn_if_fail (filter->a != NULL && filter->b != NULL);
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gst_buffer_unmap (buf, &map);
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g_mutex_unlock (&filter->lock);
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return GST_FLOW_ERROR;
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}
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filter->process (filter, map.data, num_samples);
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g_mutex_unlock (&filter->lock);
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gst_buffer_unmap (buf, &map);
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return GST_FLOW_OK;
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}
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static gboolean
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gst_audio_fx_base_iir_filter_stop (GstBaseTransform * base)
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{
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GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (base);
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guint channels = filter->nchannels;
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GstAudioFXBaseIIRFilterChannelCtx *ctx;
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guint i;
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/* Reset the history of input and output values if
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* already existing */
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if (channels && filter->channels) {
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for (i = 0; i < channels; i++) {
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ctx = &filter->channels[i];
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g_free (ctx->x);
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g_free (ctx->y);
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}
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g_free (filter->channels);
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}
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filter->channels = NULL;
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filter->nchannels = 0;
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return TRUE;
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}
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