/* * GStreamer * Copyright (C) 2007-2009 Sebastian Dröge * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include #include #include "audiofxbaseiirfilter.h" #define GST_CAT_DEFAULT gst_audio_fx_base_iir_filter_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); #define ALLOWED_CAPS \ "audio/x-raw-float," \ " width = (int) { 32, 64 }, " \ " endianness = (int) BYTE_ORDER," \ " rate = (int) [ 1, MAX ]," \ " channels = (int) [ 1, MAX ]" #define DEBUG_INIT(bla) \ GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_iir_filter_debug, "audiofxbaseiirfilter", 0, "Audio IIR Filter Base Class"); GST_BOILERPLATE_FULL (GstAudioFXBaseIIRFilter, gst_audio_fx_base_iir_filter, GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT); static gboolean gst_audio_fx_base_iir_filter_setup (GstAudioFilter * filter, GstRingBufferSpec * format); static GstFlowReturn gst_audio_fx_base_iir_filter_transform_ip (GstBaseTransform * base, GstBuffer * buf); static gboolean gst_audio_fx_base_iir_filter_stop (GstBaseTransform * base); static void process_64 (GstAudioFXBaseIIRFilter * filter, gdouble * data, guint num_samples); static void process_32 (GstAudioFXBaseIIRFilter * filter, gfloat * data, guint num_samples); /* GObject vmethod implementations */ static void gst_audio_fx_base_iir_filter_base_init (gpointer klass) { GstCaps *caps; caps = gst_caps_from_string (ALLOWED_CAPS); gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass), caps); gst_caps_unref (caps); } static void gst_audio_fx_base_iir_filter_dispose (GObject * object) { GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (object); if (filter->a) { g_free (filter->a); filter->a = NULL; } if (filter->b) { g_free (filter->b); filter->b = NULL; } if (filter->channels) { GstAudioFXBaseIIRFilterChannelCtx *ctx; guint i; for (i = 0; i < filter->nchannels; i++) { ctx = &filter->channels[i]; g_free (ctx->x); g_free (ctx->y); } g_free (filter->channels); filter->channels = NULL; } G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_audio_fx_base_iir_filter_class_init (GstAudioFXBaseIIRFilterClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass; GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass; gobject_class->dispose = gst_audio_fx_base_iir_filter_dispose; filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_fx_base_iir_filter_setup); trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_audio_fx_base_iir_filter_transform_ip); trans_class->stop = GST_DEBUG_FUNCPTR (gst_audio_fx_base_iir_filter_stop); } static void gst_audio_fx_base_iir_filter_init (GstAudioFXBaseIIRFilter * filter, GstAudioFXBaseIIRFilterClass * klass) { gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE); filter->a = NULL; filter->na = 0; filter->b = NULL; filter->nb = 0; filter->channels = NULL; filter->nchannels = 0; } /* Evaluate the transfer function that corresponds to the IIR * coefficients at zr + zi*I and return the magnitude */ gdouble gst_audio_fx_base_iir_filter_calculate_gain (gdouble * a, guint na, gdouble * b, guint nb, gdouble zr, gdouble zi) { gdouble sum_ar, sum_ai; gdouble sum_br, sum_bi; gdouble gain_r, gain_i; gdouble sum_r_old; gdouble sum_i_old; gint i; sum_ar = 0.0; sum_ai = 0.0; for (i = na - 1; i >= 0; i--) { sum_r_old = sum_ar; sum_i_old = sum_ai; sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i]; sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0; } sum_br = 0.0; sum_bi = 0.0; for (i = nb - 1; i >= 0; i--) { sum_r_old = sum_br; sum_i_old = sum_bi; sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i]; sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0; } sum_br += 1.0; sum_bi += 0.0; gain_r = (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); gain_i = (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); return (sqrt (gain_r * gain_r + gain_i * gain_i)); } void gst_audio_fx_base_iir_filter_set_coefficients (GstAudioFXBaseIIRFilter * filter, gdouble * a, guint na, gdouble * b, guint nb) { guint i; g_return_if_fail (GST_IS_AUDIO_FX_BASE_IIR_FILTER (filter)); GST_BASE_TRANSFORM_LOCK (filter); g_free (filter->a); g_free (filter->b); filter->a = filter->b = NULL; if (filter->channels) { GstAudioFXBaseIIRFilterChannelCtx *ctx; gboolean free = (na != filter->na || nb != filter->nb); for (i = 0; i < filter->nchannels; i++) { ctx = &filter->channels[i]; if (free) g_free (ctx->x); else memset (ctx->x, 0, filter->na * sizeof (gdouble)); if (free) g_free (ctx->y); else memset (ctx->y, 0, filter->nb * sizeof (gdouble)); } g_free (filter->channels); filter->channels = NULL; } filter->na = na; filter->nb = nb; filter->a = a; filter->b = b; if (filter->nchannels && !filter->channels) { GstAudioFXBaseIIRFilterChannelCtx *ctx; filter->channels = g_new0 (GstAudioFXBaseIIRFilterChannelCtx, filter->nchannels); for (i = 0; i < filter->nchannels; i++) { ctx = &filter->channels[i]; ctx->x = g_new0 (gdouble, filter->na); ctx->y = g_new0 (gdouble, filter->nb); } } GST_BASE_TRANSFORM_UNLOCK (filter); } /* GstAudioFilter vmethod implementations */ static gboolean gst_audio_fx_base_iir_filter_setup (GstAudioFilter * base, GstRingBufferSpec * format) { GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (base); gboolean ret = TRUE; if (format->width == 32) filter->process = (GstAudioFXBaseIIRFilterProcessFunc) process_32; else if (format->width == 64) filter->process = (GstAudioFXBaseIIRFilterProcessFunc) process_64; else ret = FALSE; if (format->channels != filter->nchannels) { guint i; GstAudioFXBaseIIRFilterChannelCtx *ctx; if (filter->channels) { for (i = 0; i < filter->nchannels; i++) { ctx = &filter->channels[i]; g_free (ctx->x); g_free (ctx->y); } g_free (filter->channels); filter->channels = NULL; } filter->nchannels = format->channels; filter->channels = g_new0 (GstAudioFXBaseIIRFilterChannelCtx, filter->nchannels); for (i = 0; i < filter->nchannels; i++) { ctx = &filter->channels[i]; ctx->x = g_new0 (gdouble, filter->na); ctx->y = g_new0 (gdouble, filter->nb); } } return ret; } static inline gdouble process (GstAudioFXBaseIIRFilter * filter, GstAudioFXBaseIIRFilterChannelCtx * ctx, gdouble x0) { gdouble val = filter->a[0] * x0; gint i, j; for (i = 1, j = ctx->x_pos; i < filter->na; i++) { val += filter->a[i] * ctx->x[j]; j--; if (j < 0) j = filter->na - 1; } for (i = 1, j = ctx->y_pos; i < filter->nb; i++) { val += filter->b[i] * ctx->y[j]; j--; if (j < 0) j = filter->nb - 1; } if (ctx->x) { ctx->x_pos++; if (ctx->x_pos >= filter->na) ctx->x_pos = 0; ctx->x[ctx->x_pos] = x0; } if (ctx->y) { ctx->y_pos++; if (ctx->y_pos >= filter->nb) ctx->y_pos = 0; ctx->y[ctx->y_pos] = val; } return val; } #define DEFINE_PROCESS_FUNC(width,ctype) \ static void \ process_##width (GstAudioFXBaseIIRFilter * filter, \ g##ctype * data, guint num_samples) \ { \ gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; \ gdouble val; \ \ for (i = 0; i < num_samples / channels; i++) { \ for (j = 0; j < channels; j++) { \ val = process (filter, &filter->channels[j], *data); \ *data++ = val; \ } \ } \ } DEFINE_PROCESS_FUNC (32, float); DEFINE_PROCESS_FUNC (64, double); #undef DEFINE_PROCESS_FUNC /* GstBaseTransform vmethod implementations */ static GstFlowReturn gst_audio_fx_base_iir_filter_transform_ip (GstBaseTransform * base, GstBuffer * buf) { GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (base); guint num_samples = GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8); if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf)); if (gst_base_transform_is_passthrough (base)) return GST_FLOW_OK; g_return_val_if_fail (filter->a != NULL, GST_FLOW_ERROR); filter->process (filter, GST_BUFFER_DATA (buf), num_samples); return GST_FLOW_OK; } static gboolean gst_audio_fx_base_iir_filter_stop (GstBaseTransform * base) { GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (base); guint channels = GST_AUDIO_FILTER (filter)->format.channels; GstAudioFXBaseIIRFilterChannelCtx *ctx; guint i; /* Reset the history of input and output values if * already existing */ if (channels && filter->channels) { for (i = 0; i < channels; i++) { ctx = &filter->channels[i]; g_free (ctx->x); g_free (ctx->y); } g_free (filter->channels); } filter->channels = NULL; return TRUE; }