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/* GStreamer
* Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#define FLOAT_SAMPLES 1
#include <soundtouch/BPMDetect.h>
/* FIXME: workaround for SoundTouch.h of version 1.3.1 defining those
* variables while it shouldn't. */
#undef VERSION
#undef PACKAGE_VERSION
#undef PACKAGE_TARNAME
#undef PACKAGE_STRING
#undef PACKAGE_NAME
#undef PACKAGE_BUGREPORT
#undef PACKAGE
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#include <math.h>
#include <string.h>
#include "gstbpmdetect.hh"
GST_DEBUG_CATEGORY_STATIC (gst_bpm_detect_debug);
#define GST_CAT_DEFAULT gst_bpm_detect_debug
#define GST_BPM_DETECT_GET_PRIVATE(o) (o->priv)
struct _GstBPMDetectPrivate
{
gfloat bpm;
BPMDetect *detect;
};
#define ALLOWED_CAPS \
"audio/x-raw-float, " \
" width = (int) 32, " \
" endianness = (int) BYTE_ORDER, " \
" rate = (int) [ 8000, MAX ], " \
" channels = (int) [ 1, 2 ]"
GST_BOILERPLATE (GstBPMDetect, gst_bpm_detect, GstAudioFilter,
GST_TYPE_AUDIO_FILTER);
static void gst_bpm_detect_finalize (GObject * object);
static gboolean gst_bpm_detect_stop (GstBaseTransform * trans);
static gboolean gst_bpm_detect_event (GstBaseTransform * trans,
GstEvent * event);
static GstFlowReturn gst_bpm_detect_transform_ip (GstBaseTransform * trans,
GstBuffer * in);
static gboolean gst_bpm_detect_setup (GstAudioFilter * filter,
GstRingBufferSpec * format);
static void
gst_bpm_detect_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstCaps *caps;
gst_element_class_set_details_simple (element_class, "BPM Detector",
"Filter/Analyzer/Audio", "Detect the BPM of an audio stream",
"Sebastian Dröge <slomo@circular-chaos.org>");
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
caps);
gst_caps_unref (caps);
}
static void
gst_bpm_detect_class_init (GstBPMDetectClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseTransformClass *trans_class = GST_BASE_TRANSFORM_CLASS (klass);
GstAudioFilterClass *filter_class = GST_AUDIO_FILTER_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (gst_bpm_detect_debug, "bpm_detect", 0,
"audio bpm detection element");
gobject_class->finalize = gst_bpm_detect_finalize;
trans_class->stop = GST_DEBUG_FUNCPTR (gst_bpm_detect_stop);
trans_class->event = GST_DEBUG_FUNCPTR (gst_bpm_detect_event);
trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_bpm_detect_transform_ip);
trans_class->passthrough_on_same_caps = TRUE;
filter_class->setup = GST_DEBUG_FUNCPTR (gst_bpm_detect_setup);
g_type_class_add_private (gobject_class, sizeof (GstBPMDetectPrivate));
}
static void
gst_bpm_detect_init (GstBPMDetect * bpm_detect, GstBPMDetectClass * g_class)
{
bpm_detect->priv = G_TYPE_INSTANCE_GET_PRIVATE ((bpm_detect),
GST_TYPE_BPM_DETECT, GstBPMDetectPrivate);
bpm_detect->priv->detect = NULL;
bpm_detect->bpm = 0.0;
}
static void
gst_bpm_detect_finalize (GObject * object)
{
GstBPMDetect *bpm_detect = GST_BPM_DETECT (object);
if (bpm_detect->priv->detect) {
delete bpm_detect->priv->detect;
bpm_detect->priv->detect = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_bpm_detect_stop (GstBaseTransform * trans)
{
GstBPMDetect *bpm_detect = GST_BPM_DETECT (trans);
if (bpm_detect->priv->detect) {
delete bpm_detect->priv->detect;
bpm_detect->priv->detect = NULL;
}
bpm_detect->bpm = 0.0;
return TRUE;
}
static gboolean
gst_bpm_detect_event (GstBaseTransform * trans, GstEvent * event)
{
GstBPMDetect *bpm_detect = GST_BPM_DETECT (trans);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
case GST_EVENT_EOS:
case GST_EVENT_NEWSEGMENT:
if (bpm_detect->priv->detect) {
delete bpm_detect->priv->detect;
bpm_detect->priv->detect = NULL;
}
bpm_detect->bpm = 0.0;
break;
default:
break;
}
return TRUE;
}
static gboolean
gst_bpm_detect_setup (GstAudioFilter * filter, GstRingBufferSpec * format)
{
GstBPMDetect *bpm_detect = GST_BPM_DETECT (filter);
if (bpm_detect->priv->detect) {
delete bpm_detect->priv->detect;
bpm_detect->priv->detect = NULL;
}
return TRUE;
}
static GstFlowReturn
gst_bpm_detect_transform_ip (GstBaseTransform * trans, GstBuffer * in)
{
GstBPMDetect *bpm_detect = GST_BPM_DETECT (trans);
GstAudioFilter *filter = GST_AUDIO_FILTER (trans);
gint nsamples;
gfloat *data;
gfloat bpm;
if (filter->format.channels == 0 || filter->format.rate == 0) {
GST_ERROR_OBJECT (bpm_detect, "No channels or rate set yet");
return GST_FLOW_ERROR;
}
nsamples = GST_BUFFER_SIZE (in) / (4 * filter->format.channels);
if (!bpm_detect->priv->detect)
bpm_detect->priv->detect =
new BPMDetect (filter->format.channels, filter->format.rate);
data = (gfloat *) g_memdup (GST_BUFFER_DATA (in), GST_BUFFER_SIZE (in));
bpm_detect->priv->detect->inputSamples (data, nsamples);
g_free (data);
bpm = bpm_detect->priv->detect->getBpm ();
if (bpm >= 1.0 && fabs (bpm_detect->bpm - bpm) >= 1.0) {
GstTagList *tags = gst_tag_list_new ();
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE_ALL, GST_TAG_BEATS_PER_MINUTE,
bpm, NULL);
gst_element_found_tags (GST_ELEMENT (bpm_detect), tags);
GST_INFO_OBJECT (bpm_detect, "Detected BPM: %lf\n", bpm);
bpm_detect->bpm = bpm;
}
return GST_FLOW_OK;
}
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