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77a7c4c8fb
Original commit message from CVS: 2005-07-04 Andy Wingo <wingo@pobox.com> * examples/level/: * examples/level/Makefile.am: * examples/level/README: * examples/level/demo.c: * examples/level/plot.c: Examples moved out of the source dir. Not updated tho. * configure.ac: Add level to the build. * gst/level/Makefile.am: * gst/level/gstlevel.h: * gst/level/gstlevel.c: Cleaned up, ported to 0.9.
464 lines
15 KiB
C
464 lines
15 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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*
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* gstlevel.c: signals RMS, peak and decaying peak levels
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* Copyright (C) 2000,2001,2002,2003
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* Thomas Vander Stichele <thomas at apestaart dot org>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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 <gst/gst.h>
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#include "gstlevel.h"
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#include "math.h"
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GST_DEBUG_CATEGORY (level_debug);
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#define GST_CAT_DEFAULT level_debug
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static GstElementDetails level_details = {
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"Level",
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"Filter/Analyzer/Audio",
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"RMS/Peak/Decaying Peak Level signaller for audio/raw",
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"Thomas <thomas@apestaart.org>"
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};
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static GstStaticPadTemplate sink_template_factory =
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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-int, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, 2 ], "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) { 8, 16 }, "
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"depth = (int) { 8, 16 }, " "signed = (boolean) true")
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);
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static GstStaticPadTemplate src_template_factory =
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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-int, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, 2 ], "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) { 8, 16 }, "
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"depth = (int) { 8, 16 }, " "signed = (boolean) true")
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);
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enum
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{
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PROP_0,
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PROP_SIGNAL_LEVEL,
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PROP_SIGNAL_INTERVAL,
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PROP_PEAK_TTL,
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PROP_PEAK_FALLOFF
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};
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GST_BOILERPLATE (GstLevel, gst_level, GstBaseTransform,
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GST_TYPE_BASE_TRANSFORM);
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static void gst_level_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_level_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_level_set_caps (GstBaseTransform * trans, GstCaps * in,
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GstCaps * out);
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static GstFlowReturn gst_level_transform (GstBaseTransform * trans,
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GstBuffer * in, GstBuffer ** out);
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static void
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gst_level_base_init (gpointer g_class)
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{
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GstElementClass *element_class = g_class;
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template_factory));
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gst_element_class_set_details (element_class, &level_details);
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}
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static void
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gst_level_class_init (GstLevelClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseTransformClass *trans_class;
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gobject_class = (GObjectClass *) klass;
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trans_class = (GstBaseTransformClass *) klass;
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gobject_class->set_property = gst_level_set_property;
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gobject_class->get_property = gst_level_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SIGNAL_LEVEL,
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g_param_spec_boolean ("signal", "Signal",
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"Emit level signals for each interval", TRUE, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SIGNAL_INTERVAL,
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g_param_spec_double ("interval", "Interval",
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"Interval between emissions (in seconds)",
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0.01, 100.0, 0.1, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PEAK_TTL,
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g_param_spec_double ("peak_ttl", "Peak TTL",
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"Time To Live of decay peak before it falls back",
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0, 100.0, 0.3, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PEAK_FALLOFF,
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g_param_spec_double ("peak_falloff", "Peak Falloff",
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"Decay rate of decay peak after TTL (in dB/sec)",
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0.0, G_MAXDOUBLE, 10.0, G_PARAM_READWRITE));
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GST_DEBUG_CATEGORY_INIT (level_debug, "level", 0, "Level calculation");
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trans_class->set_caps = gst_level_set_caps;
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trans_class->transform = gst_level_transform;
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}
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static void
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gst_level_init (GstLevel * filter)
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{
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filter->CS = NULL;
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filter->peak = NULL;
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filter->MS = NULL;
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filter->RMS_dB = NULL;
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filter->rate = 0;
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filter->width = 0;
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filter->channels = 0;
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filter->interval = 0.1;
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filter->decay_peak_ttl = 0.4;
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filter->decay_peak_falloff = 10.0; /* dB falloff (/sec) */
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}
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static void
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gst_level_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstLevel *filter = GST_LEVEL (object);
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switch (prop_id) {
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case PROP_SIGNAL_LEVEL:
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filter->signal = g_value_get_boolean (value);
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break;
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case PROP_SIGNAL_INTERVAL:
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filter->interval = g_value_get_double (value);
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break;
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case PROP_PEAK_TTL:
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filter->decay_peak_ttl = g_value_get_double (value);
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break;
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case PROP_PEAK_FALLOFF:
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filter->decay_peak_falloff = g_value_get_double (value);
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break;
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default:
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break;
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}
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}
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static void
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gst_level_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstLevel *filter = GST_LEVEL (object);
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switch (prop_id) {
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case PROP_SIGNAL_LEVEL:
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g_value_set_boolean (value, filter->signal);
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break;
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case PROP_SIGNAL_INTERVAL:
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g_value_set_double (value, filter->interval);
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break;
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case PROP_PEAK_TTL:
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g_value_set_double (value, filter->decay_peak_ttl);
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break;
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case PROP_PEAK_FALLOFF:
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g_value_set_double (value, filter->decay_peak_falloff);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static gint
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structure_get_int (GstStructure * structure, const gchar * field)
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{
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gint ret;
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if (!gst_structure_get_int (structure, field, &ret))
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g_assert_not_reached ();
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return ret;
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}
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static gboolean
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gst_level_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
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{
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GstLevel *filter;
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GstStructure *structure;
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int i;
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filter = GST_LEVEL (trans);
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filter->num_samples = 0;
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structure = gst_caps_get_structure (in, 0);
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filter->rate = structure_get_int (structure, "rate");
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filter->width = structure_get_int (structure, "width");
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filter->channels = structure_get_int (structure, "channels");
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/* allocate channel variable arrays */
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g_free (filter->CS);
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g_free (filter->peak);
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g_free (filter->last_peak);
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g_free (filter->decay_peak);
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g_free (filter->decay_peak_age);
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g_free (filter->MS);
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g_free (filter->RMS_dB);
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filter->CS = g_new (double, filter->channels);
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filter->peak = g_new (double, filter->channels);
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filter->last_peak = g_new (double, filter->channels);
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filter->decay_peak = g_new (double, filter->channels);
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filter->decay_peak_age = g_new (double, filter->channels);
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filter->MS = g_new (double, filter->channels);
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filter->RMS_dB = g_new (double, filter->channels);
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for (i = 0; i < filter->channels; ++i) {
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filter->CS[i] = filter->peak[i] = filter->last_peak[i] =
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filter->decay_peak[i] = filter->decay_peak_age[i] =
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filter->MS[i] = filter->RMS_dB[i] = 0.0;
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}
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return TRUE;
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}
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#if 0
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#define DEBUG(str,...) g_print (str, ...)
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#else
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#define DEBUG(str,...) /*nop */
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#endif
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/* process one (interleaved) channel of incoming samples
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* calculate square sum of samples
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* normalize and return normalized Cumulative Square
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* caller must assure num is a multiple of channels
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* this filter only accepts signed audio data, so mid level is always 0
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*/
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#define DEFINE_LEVEL_CALCULATOR(TYPE) \
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static void inline \
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gst_level_calculate_##TYPE (TYPE * in, guint num, gint channels, \
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gint resolution, double *CS, double *peak) \
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{ \
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register int j; \
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double squaresum = 0.0; /* square sum of the integer samples */ \
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register double square = 0.0; /* Square */ \
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register double PSS = 0.0; /* Peak Square Sample */ \
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gdouble normalizer; \
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\
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*CS = 0.0; /* Cumulative Square for this block */ \
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\
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normalizer = (double) (1 << resolution); \
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\
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/* \
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* process data here \
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* input sample data enters in *in_data as 8 or 16 bit data \
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* samples for left and right channel are interleaved \
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* returns the Mean Square of the samples as a double between 0 and 1 \
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*/ \
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\
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for (j = 0; j < num; j += channels) \
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{ \
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DEBUG ("ch %d -> smp %d\n", j, in[j]); \
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square = (double) (in[j] * in[j]); \
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if (square > PSS) PSS = square; \
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squaresum += square; \
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} \
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*peak = PSS / ((double) normalizer * (double) normalizer); \
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\
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/* return normalized cumulative square */ \
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*CS = squaresum / ((double) normalizer * (double) normalizer); \
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}
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DEFINE_LEVEL_CALCULATOR (gint16);
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DEFINE_LEVEL_CALCULATOR (gint8);
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static GstMessage *
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gst_level_message_new (GstLevel * l, gdouble endtime)
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{
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GstStructure *s;
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GValue v = { 0, };
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g_value_init (&v, GST_TYPE_LIST);
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s = gst_structure_new ("level", "endtime", G_TYPE_DOUBLE, endtime, NULL);
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/* will copy-by-value */
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gst_structure_set_value (s, "rms", &v);
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gst_structure_set_value (s, "peak", &v);
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gst_structure_set_value (s, "decay", &v);
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return gst_message_new_application (GST_OBJECT (l), s);
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}
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static void
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gst_level_message_append_channel (GstMessage * m, gdouble rms, gdouble peak,
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gdouble decay)
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{
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GstStructure *s;
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GValue v = { 0, };
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GValue *l;
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g_value_init (&v, G_TYPE_DOUBLE);
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s = (GstStructure *) gst_message_get_structure (m);
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l = (GValue *) gst_structure_get_value (s, "rms");
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g_value_set_double (&v, rms);
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gst_value_list_append_value (l, &v); /* copies by value */
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l = (GValue *) gst_structure_get_value (s, "peak");
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g_value_set_double (&v, peak);
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gst_value_list_append_value (l, &v); /* copies by value */
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l = (GValue *) gst_structure_get_value (s, "decay");
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g_value_set_double (&v, decay);
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gst_value_list_append_value (l, &v); /* copies by value */
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}
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static GstFlowReturn
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gst_level_transform (GstBaseTransform * trans, GstBuffer * in, GstBuffer ** out)
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{
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GstLevel *filter;
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gpointer in_data;
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double CS = 0.0;
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gint num_samples = 0;
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gint i;
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filter = GST_LEVEL (trans);
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for (i = 0; i < filter->channels; ++i)
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filter->CS[i] = filter->peak[i] = filter->MS[i] = filter->RMS_dB[i] = 0.0;
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in_data = GST_BUFFER_DATA (in);
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num_samples = GST_BUFFER_SIZE (in) / (filter->width / 8);
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g_return_val_if_fail (num_samples % filter->channels == 0,
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GST_FLOW_UNEXPECTED);
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for (i = 0; i < filter->channels; ++i) {
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switch (filter->width) {
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case 16:
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gst_level_calculate_gint16 (in_data + i, num_samples,
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filter->channels, filter->width - 1, &CS, &filter->peak[i]);
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break;
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case 8:
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gst_level_calculate_gint8 (((gint8 *) in_data) + i, num_samples,
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filter->channels, filter->width - 1, &CS, &filter->peak[i]);
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break;
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}
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GST_LOG_OBJECT (filter, "channel %d, cumulative sum %f, peak %f", i, CS,
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filter->peak[i]);
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filter->CS[i] += CS;
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}
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filter->num_samples += num_samples;
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for (i = 0; i < filter->channels; ++i) {
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filter->decay_peak_age[i] += num_samples;
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DEBUG ("filter peak info [%d]: peak %f, age %f\n", i,
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filter->last_peak[i], filter->decay_peak_age[i]);
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/* update running peak */
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if (filter->peak[i] > filter->last_peak[i])
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filter->last_peak[i] = filter->peak[i];
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/* update decay peak */
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if (filter->peak[i] >= filter->decay_peak[i]) {
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DEBUG ("new peak, %f\n", filter->peak[i]);
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filter->decay_peak[i] = filter->peak[i];
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filter->decay_peak_age[i] = 0;
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} else {
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/* make decay peak fall off if too old */
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if (filter->decay_peak_age[i] > filter->rate * filter->decay_peak_ttl) {
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double falloff_dB;
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double falloff;
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double length; /* length of buffer in seconds */
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length = (double) num_samples / (filter->channels * filter->rate);
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falloff_dB = filter->decay_peak_falloff * length;
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falloff = pow (10, falloff_dB / -20.0);
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DEBUG ("falloff: length %f, dB falloff %f, falloff factor %e\n",
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length, falloff_dB, falloff);
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filter->decay_peak[i] *= falloff;
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DEBUG ("peak is %f samples old, decayed with factor %e to %f\n",
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filter->decay_peak_age[i], falloff, filter->decay_peak[i]);
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}
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}
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}
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/* do we need to emit ? */
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if (filter->num_samples >= filter->interval * (gdouble) filter->rate) {
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if (filter->signal) {
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GstMessage *m;
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double endtime, RMS;
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endtime = (double) GST_BUFFER_TIMESTAMP (in) / GST_SECOND
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+ (double) num_samples / (double) filter->rate;
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m = gst_level_message_new (filter, endtime);
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for (i = 0; i < filter->channels; ++i) {
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RMS = sqrt (filter->CS[i] / (filter->num_samples / filter->channels));
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gst_level_message_append_channel (m, 20 * log10 (RMS),
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20 * log10 (filter->last_peak[i]),
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20 * log10 (filter->decay_peak[i]));
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/* reset cumulative and normal peak */
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filter->CS[i] = 0.0;
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filter->last_peak[i] = 0.0;
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}
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gst_element_post_message (GST_ELEMENT (filter), m);
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}
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filter->num_samples = 0;
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}
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*out = in;
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return GST_FLOW_OK;
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}
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "level", GST_RANK_NONE, GST_TYPE_LEVEL);
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}
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GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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"level",
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"Audio level plugin",
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plugin_init, VERSION, GST_LICENSE, GST_PACKAGE, GST_ORIGIN)
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