/* * GStreamer * Copyright (C) 2011 Stefan Sauer * * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /* * Freeverb * * Written by Jezar at Dreampoint, June 2000 * http://www.dreampoint.co.uk * This code is public domain * * Translated to C by Peter Hanappe, Mai 2001 * Transformed into a GStreamer plugin by Stefan Sauer, Nov 2011 */ /** * SECTION:element-freeverb * @title: freeverb * * Reverberation/room effect. * * ## Example launch line * |[ * gst-launch-1.0 audiotestsrc wave=saw ! freeverb ! autoaudiosink * gst-launch-1.0 filesrc location="melo1.ogg" ! decodebin ! audioconvert ! freeverb ! autoaudiosink * ]| * */ /* FIXME: * - add mono-to-mono, then we might also need stereo-to-mono ? */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include #include "gstfreeverb.h" #define GST_CAT_DEFAULT gst_freeverb_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); enum { PROP_0, PROP_ROOM_SIZE, PROP_DAMPING, PROP_PAN_WIDTH, PROP_LEVEL }; static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) { " GST_AUDIO_NE (F32) ", " GST_AUDIO_NE (S16) "}, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ], " "layout = (string) interleaved") ); static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) { " GST_AUDIO_NE (F32) ", " GST_AUDIO_NE (S16) "}, " "rate = (int) [ 1, MAX ], " "channels = (int) 2, " "layout = (string) interleaved") ); static void gst_freeverb_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_freeverb_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_freeverb_finalize (GObject * object); static gboolean gst_freeverb_get_unit_size (GstBaseTransform * base, GstCaps * caps, gsize * size); static GstCaps *gst_freeverb_transform_caps (GstBaseTransform * base, GstPadDirection direction, GstCaps * caps, GstCaps * filter); static gboolean gst_freeverb_set_caps (GstBaseTransform * base, GstCaps * incaps, GstCaps * outcaps); static GstFlowReturn gst_freeverb_transform (GstBaseTransform * base, GstBuffer * inbuf, GstBuffer * outbuf); static gboolean gst_freeverb_transform_m2s_int (GstFreeverb * filter, gint16 * idata, gint16 * odata, guint num_samples); static gboolean gst_freeverb_transform_s2s_int (GstFreeverb * filter, gint16 * idata, gint16 * odata, guint num_samples); static gboolean gst_freeverb_transform_m2s_float (GstFreeverb * filter, gfloat * idata, gfloat * odata, guint num_samples); static gboolean gst_freeverb_transform_s2s_float (GstFreeverb * filter, gfloat * idata, gfloat * odata, guint num_samples); /* Table with processing functions: [channels][format] */ static const GstFreeverbProcessFunc process_functions[2][2] = { { (GstFreeverbProcessFunc) gst_freeverb_transform_m2s_int, (GstFreeverbProcessFunc) gst_freeverb_transform_m2s_float, }, { (GstFreeverbProcessFunc) gst_freeverb_transform_s2s_int, (GstFreeverbProcessFunc) gst_freeverb_transform_s2s_float, } }; /*************************************************************** * * REVERB */ /* Denormalising: * * Another method fixes the problem cheaper: Use a small DC-offset in * the filter calculations. Now the signals converge not against 0, * but against the offset. The constant offset is invisible from the * outside world (i.e. it does not appear at the output. There is a * very small turn-on transient response, which should not cause * problems. */ //#define DC_OFFSET 0 #define DC_OFFSET 1e-8 //#define DC_OFFSET 0.001f /* all pass filter */ typedef struct _freeverb_allpass { gfloat feedback; gfloat *buffer; gint bufsize; gint bufidx; } freeverb_allpass; static void freeverb_allpass_setbuffer (freeverb_allpass * allpass, gint size) { allpass->bufidx = 0; allpass->buffer = g_new (gfloat, size); allpass->bufsize = size; } static void freeverb_allpass_release (freeverb_allpass * allpass) { g_free (allpass->buffer); } static void freeverb_allpass_init (freeverb_allpass * allpass) { gint i, len = allpass->bufsize; gfloat *buf = allpass->buffer; for (i = 0; i < len; i++) { buf[i] = (gfloat) DC_OFFSET; /* this is not 100 % correct. */ } } static void freeverb_allpass_setfeedback (freeverb_allpass * allpass, gfloat val) { allpass->feedback = val; } /* static gfloat freeverb_allpass_getfeedback(freeverb_allpass* allpass) { return allpass->feedback; }*/ #define freeverb_allpass_process(_allpass, _input_1) \ { \ gfloat output; \ gfloat bufout; \ bufout = _allpass.buffer[_allpass.bufidx]; \ output = bufout-_input_1; \ _allpass.buffer[_allpass.bufidx] = _input_1 + (bufout * _allpass.feedback); \ if (++_allpass.bufidx >= _allpass.bufsize) { \ _allpass.bufidx = 0; \ } \ _input_1 = output; \ } /* comb filter */ typedef struct _freeverb_comb { gfloat feedback; gfloat filterstore; gfloat damp1; gfloat damp2; gfloat *buffer; gint bufsize; gint bufidx; } freeverb_comb; static void freeverb_comb_setbuffer (freeverb_comb * comb, gint size) { comb->filterstore = 0; comb->bufidx = 0; comb->buffer = g_new (gfloat, size); comb->bufsize = size; } static void freeverb_comb_release (freeverb_comb * comb) { g_free (comb->buffer); } static void freeverb_comb_init (freeverb_comb * comb) { gint i, len = comb->bufsize; gfloat *buf = comb->buffer; for (i = 0; i < len; i++) { buf[i] = (gfloat) DC_OFFSET; /* This is not 100 % correct. */ } } static void freeverb_comb_setdamp (freeverb_comb * comb, gfloat val) { comb->damp1 = val; comb->damp2 = 1 - val; } /* static gfloat freeverb_comb_getdamp(freeverb_comb* comb) { return comb->damp1; }*/ static void freeverb_comb_setfeedback (freeverb_comb * comb, gfloat val) { comb->feedback = val; } /* static gfloat freeverb_comb_getfeedback(freeverb_comb* comb) { return comb->feedback; }*/ #define freeverb_comb_process(_comb, _input_1, _output) \ { \ gfloat _tmp = _comb.buffer[_comb.bufidx]; \ _comb.filterstore = (_tmp * _comb.damp2) + (_comb.filterstore * _comb.damp1); \ _comb.buffer[_comb.bufidx] = _input_1 + (_comb.filterstore * _comb.feedback); \ if (++_comb.bufidx >= _comb.bufsize) { \ _comb.bufidx = 0; \ } \ _output += _tmp; \ } #define numcombs 8 #define numallpasses 4 #define fixedgain 0.015f #define scalewet 1.0f #define scaledry 1.0f #define scaledamp 1.0f #define scaleroom 0.28f #define offsetroom 0.7f #define stereospread 23 /* These values assume 44.1KHz sample rate * they will need scaling for 96KHz (or other) sample rates. * The values were obtained by listening tests. */ #define combtuningL1 1116 #define combtuningR1 (1116 + stereospread) #define combtuningL2 1188 #define combtuningR2 (1188 + stereospread) #define combtuningL3 1277 #define combtuningR3 (1277 + stereospread) #define combtuningL4 1356 #define combtuningR4 (1356 + stereospread) #define combtuningL5 1422 #define combtuningR5 (1422 + stereospread) #define combtuningL6 1491 #define combtuningR6 (1491 + stereospread) #define combtuningL7 1557 #define combtuningR7 (1557 + stereospread) #define combtuningL8 1617 #define combtuningR8 (1617 + stereospread) #define allpasstuningL1 556 #define allpasstuningR1 (556 + stereospread) #define allpasstuningL2 441 #define allpasstuningR2 (441 + stereospread) #define allpasstuningL3 341 #define allpasstuningR3 (341 + stereospread) #define allpasstuningL4 225 #define allpasstuningR4 (225 + stereospread) struct _GstFreeverbPrivate { gfloat roomsize; gfloat damp; gfloat wet, wet1, wet2, dry; gfloat width; gfloat gain; /* The following are all declared inline to remove the need for dynamic allocation with its subsequent error-checking messiness */ /* Comb filters */ freeverb_comb combL[numcombs]; freeverb_comb combR[numcombs]; /* Allpass filters */ freeverb_allpass allpassL[numallpasses]; freeverb_allpass allpassR[numallpasses]; }; G_DEFINE_TYPE_WITH_CODE (GstFreeverb, gst_freeverb, GST_TYPE_BASE_TRANSFORM, G_ADD_PRIVATE (GstFreeverb) G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, NULL)); static void freeverb_revmodel_init (GstFreeverb * filter) { GstFreeverbPrivate *priv = filter->priv; gint i; for (i = 0; i < numcombs; i++) { freeverb_comb_init (&priv->combL[i]); freeverb_comb_init (&priv->combR[i]); } for (i = 0; i < numallpasses; i++) { freeverb_allpass_init (&priv->allpassL[i]); freeverb_allpass_init (&priv->allpassR[i]); } } static void freeverb_revmodel_free (GstFreeverb * filter) { GstFreeverbPrivate *priv = filter->priv; gint i; for (i = 0; i < numcombs; i++) { freeverb_comb_release (&priv->combL[i]); freeverb_comb_release (&priv->combR[i]); } for (i = 0; i < numallpasses; i++) { freeverb_allpass_release (&priv->allpassL[i]); freeverb_allpass_release (&priv->allpassR[i]); } } /* GObject vmethod implementations */ static void gst_freeverb_class_init (GstFreeverbClass * klass) { GObjectClass *gobject_class; GstElementClass *element_class; GST_DEBUG_CATEGORY_INIT (gst_freeverb_debug, "freeverb", 0, "freeverb element"); gobject_class = (GObjectClass *) klass; element_class = (GstElementClass *) klass; gobject_class->set_property = gst_freeverb_set_property; gobject_class->get_property = gst_freeverb_get_property; gobject_class->finalize = gst_freeverb_finalize; g_object_class_install_property (gobject_class, PROP_ROOM_SIZE, g_param_spec_float ("room-size", "Room size", "Size of the simulated room", 0.0, 1.0, 0.5, G_PARAM_CONSTRUCT | G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_DAMPING, g_param_spec_float ("damping", "Damping", "Damping of high frequencies", 0.0, 1.0, 0.2f, G_PARAM_CONSTRUCT | G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PAN_WIDTH, g_param_spec_float ("width", "Width", "Stereo panorama width", 0.0, 1.0, 1.0, G_PARAM_CONSTRUCT | G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_LEVEL, g_param_spec_float ("level", "Level", "dry/wet level", 0.0, 1.0, 0.5, G_PARAM_CONSTRUCT | G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); gst_element_class_set_static_metadata (element_class, "Reverberation/room effect", "Filter/Effect/Audio", "Add reverberation to audio streams", "Stefan Sauer "); gst_element_class_add_static_pad_template (element_class, &src_template); gst_element_class_add_static_pad_template (element_class, &sink_template); GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size = GST_DEBUG_FUNCPTR (gst_freeverb_get_unit_size); GST_BASE_TRANSFORM_CLASS (klass)->transform_caps = GST_DEBUG_FUNCPTR (gst_freeverb_transform_caps); GST_BASE_TRANSFORM_CLASS (klass)->set_caps = GST_DEBUG_FUNCPTR (gst_freeverb_set_caps); GST_BASE_TRANSFORM_CLASS (klass)->transform = GST_DEBUG_FUNCPTR (gst_freeverb_transform); } static void gst_freeverb_init (GstFreeverb * filter) { filter->priv = gst_freeverb_get_instance_private (filter); gst_audio_info_init (&filter->info); filter->process = NULL; gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE); freeverb_revmodel_init (filter); } static void gst_freeverb_finalize (GObject * object) { GstFreeverb *filter = GST_FREEVERB (object); freeverb_revmodel_free (filter); G_OBJECT_CLASS (gst_freeverb_parent_class)->finalize (object); } static gboolean gst_freeverb_set_process_function (GstFreeverb * filter, GstAudioInfo * info) { gint channel_index, format_index; const GstAudioFormatInfo *finfo = info->finfo; /* set processing function */ channel_index = GST_AUDIO_INFO_CHANNELS (info) - 1; if (channel_index > 1 || channel_index < 0) { filter->process = NULL; return FALSE; } format_index = GST_AUDIO_FORMAT_INFO_IS_FLOAT (finfo) ? 1 : 0; filter->process = process_functions[channel_index][format_index]; return TRUE; } static void gst_freeverb_init_rev_model (GstFreeverb * filter) { gfloat srfactor = GST_AUDIO_INFO_RATE (&filter->info) / 44100.0f; GstFreeverbPrivate *priv = filter->priv; freeverb_revmodel_free (filter); priv->gain = fixedgain; freeverb_comb_setbuffer (&priv->combL[0], combtuningL1 * srfactor); freeverb_comb_setbuffer (&priv->combR[0], combtuningR1 * srfactor); freeverb_comb_setbuffer (&priv->combL[1], combtuningL2 * srfactor); freeverb_comb_setbuffer (&priv->combR[1], combtuningR2 * srfactor); freeverb_comb_setbuffer (&priv->combL[2], combtuningL3 * srfactor); freeverb_comb_setbuffer (&priv->combR[2], combtuningR3 * srfactor); freeverb_comb_setbuffer (&priv->combL[3], combtuningL4 * srfactor); freeverb_comb_setbuffer (&priv->combR[3], combtuningR4 * srfactor); freeverb_comb_setbuffer (&priv->combL[4], combtuningL5 * srfactor); freeverb_comb_setbuffer (&priv->combR[4], combtuningR5 * srfactor); freeverb_comb_setbuffer (&priv->combL[5], combtuningL6 * srfactor); freeverb_comb_setbuffer (&priv->combR[5], combtuningR6 * srfactor); freeverb_comb_setbuffer (&priv->combL[6], combtuningL7 * srfactor); freeverb_comb_setbuffer (&priv->combR[6], combtuningR7 * srfactor); freeverb_comb_setbuffer (&priv->combL[7], combtuningL8 * srfactor); freeverb_comb_setbuffer (&priv->combR[7], combtuningR8 * srfactor); freeverb_allpass_setbuffer (&priv->allpassL[0], allpasstuningL1 * srfactor); freeverb_allpass_setbuffer (&priv->allpassR[0], allpasstuningR1 * srfactor); freeverb_allpass_setbuffer (&priv->allpassL[1], allpasstuningL2 * srfactor); freeverb_allpass_setbuffer (&priv->allpassR[1], allpasstuningR2 * srfactor); freeverb_allpass_setbuffer (&priv->allpassL[2], allpasstuningL3 * srfactor); freeverb_allpass_setbuffer (&priv->allpassR[2], allpasstuningR3 * srfactor); freeverb_allpass_setbuffer (&priv->allpassL[3], allpasstuningL4 * srfactor); freeverb_allpass_setbuffer (&priv->allpassR[3], allpasstuningR4 * srfactor); /* clear buffers */ freeverb_revmodel_init (filter); /* set default values */ freeverb_allpass_setfeedback (&priv->allpassL[0], 0.5f); freeverb_allpass_setfeedback (&priv->allpassR[0], 0.5f); freeverb_allpass_setfeedback (&priv->allpassL[1], 0.5f); freeverb_allpass_setfeedback (&priv->allpassR[1], 0.5f); freeverb_allpass_setfeedback (&priv->allpassL[2], 0.5f); freeverb_allpass_setfeedback (&priv->allpassR[2], 0.5f); freeverb_allpass_setfeedback (&priv->allpassL[3], 0.5f); freeverb_allpass_setfeedback (&priv->allpassR[3], 0.5f); } static void gst_freeverb_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstFreeverb *filter = GST_FREEVERB (object); GstFreeverbPrivate *priv = filter->priv; gint i; switch (prop_id) { case PROP_ROOM_SIZE: filter->room_size = g_value_get_float (value); priv->roomsize = (filter->room_size * scaleroom) + offsetroom; for (i = 0; i < numcombs; i++) { freeverb_comb_setfeedback (&priv->combL[i], priv->roomsize); freeverb_comb_setfeedback (&priv->combR[i], priv->roomsize); } break; case PROP_DAMPING: filter->damping = g_value_get_float (value); priv->damp = filter->damping * scaledamp; for (i = 0; i < numcombs; i++) { freeverb_comb_setdamp (&priv->combL[i], priv->damp); freeverb_comb_setdamp (&priv->combR[i], priv->damp); } break; case PROP_PAN_WIDTH: filter->pan_width = g_value_get_float (value); priv->width = filter->pan_width; priv->wet1 = priv->wet * (priv->width / 2.0f + 0.5f); priv->wet2 = priv->wet * ((1.0f - priv->width) / 2.0f); break; case PROP_LEVEL: filter->level = g_value_get_float (value); priv->wet = filter->level * scalewet; priv->dry = (1.0 - filter->level) * scaledry; priv->wet1 = priv->wet * (priv->width / 2.0f + 0.5f); priv->wet2 = priv->wet * ((1.0f - priv->width) / 2.0f); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_freeverb_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstFreeverb *filter = GST_FREEVERB (object); switch (prop_id) { case PROP_ROOM_SIZE: g_value_set_float (value, filter->room_size); break; case PROP_DAMPING: g_value_set_float (value, filter->damping); break; case PROP_PAN_WIDTH: g_value_set_float (value, filter->pan_width); break; case PROP_LEVEL: g_value_set_float (value, filter->level); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } /* GstBaseTransform vmethod implementations */ static gboolean gst_freeverb_get_unit_size (GstBaseTransform * base, GstCaps * caps, gsize * size) { GstAudioInfo info; g_assert (size); if (!gst_audio_info_from_caps (&info, caps)) return FALSE; *size = GST_AUDIO_INFO_BPF (&info); GST_INFO_OBJECT (base, "unit size: %" G_GSIZE_FORMAT, *size); return TRUE; } static GstCaps * gst_freeverb_transform_caps (GstBaseTransform * base, GstPadDirection direction, GstCaps * caps, GstCaps * filter) { GstCaps *res; GstStructure *structure; gint i; /* replace the channel property with our range. */ res = gst_caps_copy (caps); for (i = 0; i < gst_caps_get_size (res); i++) { structure = gst_caps_get_structure (res, i); if (direction == GST_PAD_SRC) { GST_INFO_OBJECT (base, "[%d] allow 1-2 channels", i); gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL); } else { GST_INFO_OBJECT (base, "[%d] allow 2 channels", i); gst_structure_set (structure, "channels", G_TYPE_INT, 2, NULL); } gst_structure_remove_field (structure, "channel-mask"); } GST_DEBUG_OBJECT (base, "transformed %" GST_PTR_FORMAT, res); if (filter) { GstCaps *intersection; GST_DEBUG_OBJECT (base, "Using filter caps %" GST_PTR_FORMAT, filter); intersection = gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (res); res = intersection; GST_DEBUG_OBJECT (base, "Intersection %" GST_PTR_FORMAT, res); } return res; } static gboolean gst_freeverb_set_caps (GstBaseTransform * base, GstCaps * incaps, GstCaps * outcaps) { GstFreeverb *filter = GST_FREEVERB (base); GstAudioInfo info; /*GST_INFO ("incaps are %" GST_PTR_FORMAT, incaps); */ if (!gst_audio_info_from_caps (&info, incaps)) goto no_format; GST_DEBUG ("try to process %d input with %d channels", GST_AUDIO_INFO_FORMAT (&info), GST_AUDIO_INFO_CHANNELS (&info)); if (!gst_freeverb_set_process_function (filter, &info)) goto no_format; filter->info = info; gst_freeverb_init_rev_model (filter); filter->drained = FALSE; GST_INFO_OBJECT (base, "model configured"); return TRUE; no_format: { GST_DEBUG ("invalid caps"); return FALSE; } } static gboolean gst_freeverb_transform_m2s_int (GstFreeverb * filter, gint16 * idata, gint16 * odata, guint num_samples) { GstFreeverbPrivate *priv = filter->priv; gint i, k; gfloat out_l1, out_r1, input_1; gfloat out_l2, out_r2, input_2; gboolean drained = TRUE; for (k = 0; k < num_samples; k++) { out_l1 = out_r1 = 0.0; /* The original Freeverb code expects a stereo signal and 'input_1' * is set to the sum of the left and right input_1 sample. Since * this code works on a mono signal, 'input_1' is set to twice the * input_1 sample. */ input_2 = (gfloat) * idata++; input_1 = (2.0f * input_2 + DC_OFFSET) * priv->gain; /* Accumulate comb filters in parallel */ for (i = 0; i < numcombs; i++) { freeverb_comb_process (priv->combL[i], input_1, out_l1); freeverb_comb_process (priv->combR[i], input_1, out_r1); } /* Feed through allpasses in series */ for (i = 0; i < numallpasses; i++) { freeverb_allpass_process (priv->allpassL[i], out_l1); freeverb_allpass_process (priv->allpassR[i], out_r1); } /* Remove the DC offset */ out_l1 -= (gfloat) DC_OFFSET; out_r1 -= (gfloat) DC_OFFSET; /* Calculate output */ out_l2 = out_l1 * priv->wet1 + out_r1 * priv->wet2 + input_2 * priv->dry; out_r2 = out_r1 * priv->wet1 + out_l1 * priv->wet2 + input_2 * priv->dry; out_l2 = CLAMP (out_l2, G_MININT16, G_MAXINT16); out_r2 = CLAMP (out_r2, G_MININT16, G_MAXINT16); *odata++ = (gint16) out_l2; *odata++ = (gint16) out_r2; if (abs ((gint16) out_l2) > 0 || abs ((gint16) out_r2) > 0) drained = FALSE; } return drained; } static gboolean gst_freeverb_transform_s2s_int (GstFreeverb * filter, gint16 * idata, gint16 * odata, guint num_samples) { GstFreeverbPrivate *priv = filter->priv; gint i, k; gfloat out_l1, out_r1, input_1l, input_1r; gfloat out_l2, out_r2, input_2l, input_2r; gboolean drained = TRUE; for (k = 0; k < num_samples; k++) { out_l1 = out_r1 = 0.0; input_2l = (gfloat) * idata++; input_2r = (gfloat) * idata++; input_1l = (input_2l + DC_OFFSET) * priv->gain; input_1r = (input_2r + DC_OFFSET) * priv->gain; /* Accumulate comb filters in parallel */ for (i = 0; i < numcombs; i++) { freeverb_comb_process (priv->combL[i], input_1l, out_l1); freeverb_comb_process (priv->combR[i], input_1r, out_r1); } /* Feed through allpasses in series */ for (i = 0; i < numallpasses; i++) { freeverb_allpass_process (priv->allpassL[i], out_l1); freeverb_allpass_process (priv->allpassR[i], out_r1); } /* Remove the DC offset */ out_l1 -= (gfloat) DC_OFFSET; out_r1 -= (gfloat) DC_OFFSET; /* Calculate output */ out_l2 = out_l1 * priv->wet1 + out_r1 * priv->wet2 + input_2l * priv->dry; out_r2 = out_r1 * priv->wet1 + out_l1 * priv->wet2 + input_2r * priv->dry; out_l2 = CLAMP (out_l2, G_MININT16, G_MAXINT16); out_r2 = CLAMP (out_r2, G_MININT16, G_MAXINT16); *odata++ = (gint16) out_l2; *odata++ = (gint16) out_r2; if (abs ((gint16) out_l2) > 0 || abs ((gint16) out_r2) > 0) drained = FALSE; } return drained; } static gboolean gst_freeverb_transform_m2s_float (GstFreeverb * filter, gfloat * idata, gfloat * odata, guint num_samples) { GstFreeverbPrivate *priv = filter->priv; gint i, k; gfloat out_l1, out_r1, input_1; gfloat out_l2, out_r2, input_2; gboolean drained = TRUE; for (k = 0; k < num_samples; k++) { out_l1 = out_r1 = 0.0; /* The original Freeverb code expects a stereo signal and 'input_1' * is set to the sum of the left and right input_1 sample. Since * this code works on a mono signal, 'input_1' is set to twice the * input_1 sample. */ input_2 = *idata++; input_1 = (2.0f * input_2 + DC_OFFSET) * priv->gain; /* Accumulate comb filters in parallel */ for (i = 0; i < numcombs; i++) { freeverb_comb_process (priv->combL[i], input_1, out_l1); freeverb_comb_process (priv->combR[i], input_1, out_r1); } /* Feed through allpasses in series */ for (i = 0; i < numallpasses; i++) { freeverb_allpass_process (priv->allpassL[i], out_l1); freeverb_allpass_process (priv->allpassR[i], out_r1); } /* Remove the DC offset */ out_l1 -= (gfloat) DC_OFFSET; out_r1 -= (gfloat) DC_OFFSET; /* Calculate output */ out_l2 = out_l1 * priv->wet1 + out_r1 * priv->wet2 + input_2 * priv->dry; out_r2 = out_r1 * priv->wet1 + out_l1 * priv->wet2 + input_2 * priv->dry; *odata++ = out_l2; *odata++ = out_r2; if (fabs (out_l2) > 0 || fabs (out_r2) > 0) drained = FALSE; } return drained; } static gboolean gst_freeverb_transform_s2s_float (GstFreeverb * filter, gfloat * idata, gfloat * odata, guint num_samples) { GstFreeverbPrivate *priv = filter->priv; gint i, k; gfloat out_l1, out_r1, input_1l, input_1r; gfloat out_l2, out_r2, input_2l, input_2r; gboolean drained = TRUE; for (k = 0; k < num_samples; k++) { out_l1 = out_r1 = 0.0; input_2l = *idata++; input_2r = *idata++; input_1l = (input_2l + DC_OFFSET) * priv->gain; input_1r = (input_2r + DC_OFFSET) * priv->gain; /* Accumulate comb filters in parallel */ for (i = 0; i < numcombs; i++) { freeverb_comb_process (priv->combL[i], input_1l, out_l1); freeverb_comb_process (priv->combR[i], input_1r, out_r1); } /* Feed through allpasses in series */ for (i = 0; i < numallpasses; i++) { freeverb_allpass_process (priv->allpassL[i], out_l1); freeverb_allpass_process (priv->allpassR[i], out_r1); } /* Remove the DC offset */ out_l1 -= (gfloat) DC_OFFSET; out_r1 -= (gfloat) DC_OFFSET; /* Calculate output */ out_l2 = out_l1 * priv->wet1 + out_r1 * priv->wet2 + input_2l * priv->dry; out_r2 = out_r1 * priv->wet1 + out_l1 * priv->wet2 + input_2r * priv->dry; *odata++ = out_l2; *odata++ = out_r2; if (fabs (out_l2) > 0 || fabs (out_r2) > 0) drained = FALSE; } return drained; } /* this function does the actual processing */ static GstFlowReturn gst_freeverb_transform (GstBaseTransform * base, GstBuffer * inbuf, GstBuffer * outbuf) { GstFreeverb *filter = GST_FREEVERB (base); guint num_samples; GstClockTime timestamp; GstMapInfo inmap, outmap; timestamp = GST_BUFFER_TIMESTAMP (inbuf); timestamp = gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp); gst_buffer_map (inbuf, &inmap, GST_MAP_READ); gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE); num_samples = outmap.size / (2 * GST_AUDIO_INFO_BPS (&filter->info)); GST_DEBUG_OBJECT (filter, "processing %u samples at %" GST_TIME_FORMAT, num_samples, GST_TIME_ARGS (timestamp)); if (GST_CLOCK_TIME_IS_VALID (timestamp)) gst_object_sync_values (GST_OBJECT (filter), timestamp); if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DISCONT))) { filter->drained = FALSE; } if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))) { if (filter->drained) { memset (outmap.data, 0, outmap.size); } } else { filter->drained = FALSE; } if (!filter->drained) { filter->drained = filter->process (filter, inmap.data, outmap.data, num_samples); } if (filter->drained) { GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP); } gst_buffer_unmap (inbuf, &inmap); gst_buffer_unmap (outbuf, &outmap); return GST_FLOW_OK; } static gboolean plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "freeverb", GST_RANK_NONE, GST_TYPE_FREEVERB); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, freeverb, "Reverberation/room effect", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)