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940 lines
28 KiB
C
940 lines
28 KiB
C
/*
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* GStreamer
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* Copyright (C) 2011 Stefan Sauer <ensonic@users.sf.net>
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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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/*
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* Freeverb
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*
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* Written by Jezar at Dreampoint, June 2000
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* http://www.dreampoint.co.uk
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* This code is public domain
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*
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* Translated to C by Peter Hanappe, Mai 2001
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* Transformed into a GStreamer plugin by Stefan Sauer, Nov 2011
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*/
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/**
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* SECTION:element-freeverb
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* @title: freeverb
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*
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* Reverberation/room effect.
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 audiotestsrc wave=saw ! freeverb ! autoaudiosink
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* gst-launch-1.0 filesrc location="melo1.ogg" ! decodebin ! audioconvert ! freeverb ! autoaudiosink
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* ]|
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*
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*/
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/* FIXME:
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* - add mono-to-mono, then we might also need stereo-to-mono ?
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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 <math.h>
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#include <stdlib.h>
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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 "gstfreeverb.h"
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#define GST_CAT_DEFAULT gst_freeverb_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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enum
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{
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PROP_0,
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PROP_ROOM_SIZE,
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PROP_DAMPING,
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PROP_PAN_WIDTH,
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PROP_LEVEL
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};
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) { " GST_AUDIO_NE (F32) ", " GST_AUDIO_NE (S16) "}, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ], "
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"layout = (string) interleaved")
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);
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) { " GST_AUDIO_NE (F32) ", " GST_AUDIO_NE (S16) "}, "
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"rate = (int) [ 1, MAX ], " "channels = (int) 2, "
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"layout = (string) interleaved")
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);
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static void gst_freeverb_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_freeverb_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_freeverb_finalize (GObject * object);
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static gboolean gst_freeverb_get_unit_size (GstBaseTransform * base,
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GstCaps * caps, gsize * size);
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static GstCaps *gst_freeverb_transform_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps, GstCaps * filter);
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static gboolean gst_freeverb_set_caps (GstBaseTransform * base,
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GstCaps * incaps, GstCaps * outcaps);
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static GstFlowReturn gst_freeverb_transform (GstBaseTransform * base,
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GstBuffer * inbuf, GstBuffer * outbuf);
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static gboolean gst_freeverb_transform_m2s_int (GstFreeverb * filter,
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gint16 * idata, gint16 * odata, guint num_samples);
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static gboolean gst_freeverb_transform_s2s_int (GstFreeverb * filter,
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gint16 * idata, gint16 * odata, guint num_samples);
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static gboolean gst_freeverb_transform_m2s_float (GstFreeverb * filter,
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gfloat * idata, gfloat * odata, guint num_samples);
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static gboolean gst_freeverb_transform_s2s_float (GstFreeverb * filter,
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gfloat * idata, gfloat * odata, guint num_samples);
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/* Table with processing functions: [channels][format] */
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static const GstFreeverbProcessFunc process_functions[2][2] = {
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{
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(GstFreeverbProcessFunc) gst_freeverb_transform_m2s_int,
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(GstFreeverbProcessFunc) gst_freeverb_transform_m2s_float,
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},
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{
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(GstFreeverbProcessFunc) gst_freeverb_transform_s2s_int,
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(GstFreeverbProcessFunc) gst_freeverb_transform_s2s_float,
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}
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};
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/***************************************************************
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*
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* REVERB
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*/
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/* Denormalising:
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*
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* Another method fixes the problem cheaper: Use a small DC-offset in
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* the filter calculations. Now the signals converge not against 0,
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* but against the offset. The constant offset is invisible from the
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* outside world (i.e. it does not appear at the output. There is a
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* very small turn-on transient response, which should not cause
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* problems.
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*/
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//#define DC_OFFSET 0
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#define DC_OFFSET 1e-8
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//#define DC_OFFSET 0.001f
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/* all pass filter */
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typedef struct _freeverb_allpass
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{
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gfloat feedback;
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gfloat *buffer;
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gint bufsize;
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gint bufidx;
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} freeverb_allpass;
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static void
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freeverb_allpass_setbuffer (freeverb_allpass * allpass, gint size)
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{
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allpass->bufidx = 0;
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allpass->buffer = g_new (gfloat, size);
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allpass->bufsize = size;
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}
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static void
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freeverb_allpass_release (freeverb_allpass * allpass)
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{
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g_free (allpass->buffer);
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}
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static void
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freeverb_allpass_init (freeverb_allpass * allpass)
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{
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gint i, len = allpass->bufsize;
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gfloat *buf = allpass->buffer;
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for (i = 0; i < len; i++) {
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buf[i] = (gfloat) DC_OFFSET; /* this is not 100 % correct. */
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}
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}
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static void
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freeverb_allpass_setfeedback (freeverb_allpass * allpass, gfloat val)
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{
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allpass->feedback = val;
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}
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/*
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static gfloat
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freeverb_allpass_getfeedback(freeverb_allpass* allpass)
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{
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return allpass->feedback;
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}*/
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#define freeverb_allpass_process(_allpass, _input_1) \
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{ \
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gfloat output; \
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gfloat bufout; \
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bufout = _allpass.buffer[_allpass.bufidx]; \
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output = bufout-_input_1; \
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_allpass.buffer[_allpass.bufidx] = _input_1 + (bufout * _allpass.feedback); \
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if (++_allpass.bufidx >= _allpass.bufsize) { \
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_allpass.bufidx = 0; \
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} \
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_input_1 = output; \
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}
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/* comb filter */
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typedef struct _freeverb_comb
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{
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gfloat feedback;
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gfloat filterstore;
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gfloat damp1;
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gfloat damp2;
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gfloat *buffer;
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gint bufsize;
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gint bufidx;
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} freeverb_comb;
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static void
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freeverb_comb_setbuffer (freeverb_comb * comb, gint size)
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{
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comb->filterstore = 0;
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comb->bufidx = 0;
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comb->buffer = g_new (gfloat, size);
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comb->bufsize = size;
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}
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static void
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freeverb_comb_release (freeverb_comb * comb)
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{
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g_free (comb->buffer);
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}
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static void
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freeverb_comb_init (freeverb_comb * comb)
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{
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gint i, len = comb->bufsize;
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gfloat *buf = comb->buffer;
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for (i = 0; i < len; i++) {
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buf[i] = (gfloat) DC_OFFSET; /* This is not 100 % correct. */
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}
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}
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static void
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freeverb_comb_setdamp (freeverb_comb * comb, gfloat val)
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{
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comb->damp1 = val;
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comb->damp2 = 1 - val;
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}
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/*
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static gfloat
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freeverb_comb_getdamp(freeverb_comb* comb)
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{
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return comb->damp1;
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}*/
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static void
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freeverb_comb_setfeedback (freeverb_comb * comb, gfloat val)
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{
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comb->feedback = val;
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}
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/*
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static gfloat
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freeverb_comb_getfeedback(freeverb_comb* comb)
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{
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return comb->feedback;
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}*/
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#define freeverb_comb_process(_comb, _input_1, _output) \
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{ \
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gfloat _tmp = _comb.buffer[_comb.bufidx]; \
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_comb.filterstore = (_tmp * _comb.damp2) + (_comb.filterstore * _comb.damp1); \
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_comb.buffer[_comb.bufidx] = _input_1 + (_comb.filterstore * _comb.feedback); \
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if (++_comb.bufidx >= _comb.bufsize) { \
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_comb.bufidx = 0; \
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} \
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_output += _tmp; \
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}
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#define numcombs 8
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#define numallpasses 4
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#define fixedgain 0.015f
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#define scalewet 1.0f
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#define scaledry 1.0f
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#define scaledamp 1.0f
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#define scaleroom 0.28f
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#define offsetroom 0.7f
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#define stereospread 23
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/* These values assume 44.1KHz sample rate
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* they will need scaling for 96KHz (or other) sample rates.
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* The values were obtained by listening tests.
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*/
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#define combtuningL1 1116
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#define combtuningR1 (1116 + stereospread)
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#define combtuningL2 1188
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#define combtuningR2 (1188 + stereospread)
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#define combtuningL3 1277
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#define combtuningR3 (1277 + stereospread)
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#define combtuningL4 1356
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#define combtuningR4 (1356 + stereospread)
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#define combtuningL5 1422
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#define combtuningR5 (1422 + stereospread)
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#define combtuningL6 1491
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#define combtuningR6 (1491 + stereospread)
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#define combtuningL7 1557
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#define combtuningR7 (1557 + stereospread)
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#define combtuningL8 1617
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#define combtuningR8 (1617 + stereospread)
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#define allpasstuningL1 556
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#define allpasstuningR1 (556 + stereospread)
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#define allpasstuningL2 441
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#define allpasstuningR2 (441 + stereospread)
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#define allpasstuningL3 341
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#define allpasstuningR3 (341 + stereospread)
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#define allpasstuningL4 225
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#define allpasstuningR4 (225 + stereospread)
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struct _GstFreeverbPrivate
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{
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gfloat roomsize;
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gfloat damp;
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gfloat wet, wet1, wet2, dry;
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gfloat width;
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gfloat gain;
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/*
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The following are all declared inline
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to remove the need for dynamic allocation
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with its subsequent error-checking messiness
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*/
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/* Comb filters */
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freeverb_comb combL[numcombs];
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freeverb_comb combR[numcombs];
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/* Allpass filters */
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freeverb_allpass allpassL[numallpasses];
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freeverb_allpass allpassR[numallpasses];
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};
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G_DEFINE_TYPE_WITH_CODE (GstFreeverb, gst_freeverb, GST_TYPE_BASE_TRANSFORM,
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G_ADD_PRIVATE (GstFreeverb)
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G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, NULL));
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GST_ELEMENT_REGISTER_DEFINE (freeverb, "freeverb",
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GST_RANK_NONE, GST_TYPE_FREEVERB);
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static void
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freeverb_revmodel_init (GstFreeverb * filter)
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{
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GstFreeverbPrivate *priv = filter->priv;
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gint i;
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for (i = 0; i < numcombs; i++) {
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freeverb_comb_init (&priv->combL[i]);
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freeverb_comb_init (&priv->combR[i]);
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}
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for (i = 0; i < numallpasses; i++) {
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freeverb_allpass_init (&priv->allpassL[i]);
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freeverb_allpass_init (&priv->allpassR[i]);
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}
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}
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static void
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freeverb_revmodel_free (GstFreeverb * filter)
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{
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GstFreeverbPrivate *priv = filter->priv;
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gint i;
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for (i = 0; i < numcombs; i++) {
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freeverb_comb_release (&priv->combL[i]);
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freeverb_comb_release (&priv->combR[i]);
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}
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for (i = 0; i < numallpasses; i++) {
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freeverb_allpass_release (&priv->allpassL[i]);
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freeverb_allpass_release (&priv->allpassR[i]);
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}
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}
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/* GObject vmethod implementations */
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static void
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gst_freeverb_class_init (GstFreeverbClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *element_class;
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GST_DEBUG_CATEGORY_INIT (gst_freeverb_debug, "freeverb", 0,
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"freeverb element");
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gobject_class = (GObjectClass *) klass;
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element_class = (GstElementClass *) klass;
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gobject_class->set_property = gst_freeverb_set_property;
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gobject_class->get_property = gst_freeverb_get_property;
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gobject_class->finalize = gst_freeverb_finalize;
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g_object_class_install_property (gobject_class, PROP_ROOM_SIZE,
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g_param_spec_float ("room-size", "Room size",
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"Size of the simulated room", 0.0, 1.0, 0.5,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_DAMPING,
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g_param_spec_float ("damping", "Damping", "Damping of high frequencies",
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0.0, 1.0, 0.2f,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_PAN_WIDTH,
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g_param_spec_float ("width", "Width", "Stereo panorama width", 0.0, 1.0,
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1.0,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_LEVEL,
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g_param_spec_float ("level", "Level", "dry/wet level", 0.0, 1.0, 0.5,
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G_PARAM_CONSTRUCT | G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
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G_PARAM_STATIC_STRINGS));
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gst_element_class_set_static_metadata (element_class,
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"Reverberation/room effect", "Filter/Effect/Audio",
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"Add reverberation to audio streams",
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"Stefan Sauer <ensonic@users.sf.net>");
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gst_element_class_add_static_pad_template (element_class, &src_template);
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gst_element_class_add_static_pad_template (element_class, &sink_template);
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GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
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GST_DEBUG_FUNCPTR (gst_freeverb_get_unit_size);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
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GST_DEBUG_FUNCPTR (gst_freeverb_transform_caps);
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GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
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GST_DEBUG_FUNCPTR (gst_freeverb_set_caps);
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GST_BASE_TRANSFORM_CLASS (klass)->transform =
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GST_DEBUG_FUNCPTR (gst_freeverb_transform);
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}
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static void
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gst_freeverb_init (GstFreeverb * filter)
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{
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filter->priv = gst_freeverb_get_instance_private (filter);
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gst_audio_info_init (&filter->info);
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filter->process = NULL;
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gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
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freeverb_revmodel_init (filter);
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}
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static void
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gst_freeverb_finalize (GObject * object)
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{
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GstFreeverb *filter = GST_FREEVERB (object);
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freeverb_revmodel_free (filter);
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G_OBJECT_CLASS (gst_freeverb_parent_class)->finalize (object);
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}
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static gboolean
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gst_freeverb_set_process_function (GstFreeverb * filter, GstAudioInfo * info)
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{
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gint channel_index, format_index;
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const GstAudioFormatInfo *finfo = info->finfo;
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/* set processing function */
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channel_index = GST_AUDIO_INFO_CHANNELS (info) - 1;
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if (channel_index > 1 || channel_index < 0) {
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filter->process = NULL;
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return FALSE;
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}
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format_index = GST_AUDIO_FORMAT_INFO_IS_FLOAT (finfo) ? 1 : 0;
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filter->process = process_functions[channel_index][format_index];
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return TRUE;
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}
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static void
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gst_freeverb_init_rev_model (GstFreeverb * filter)
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{
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gfloat srfactor = GST_AUDIO_INFO_RATE (&filter->info) / 44100.0f;
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GstFreeverbPrivate *priv = filter->priv;
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freeverb_revmodel_free (filter);
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priv->gain = fixedgain;
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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 (freeverb, plugin);
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
freeverb,
|
|
"Reverberation/room effect",
|
|
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
|