gstreamer/ext/soundtouch/gstbpmdetect.cc
Sebastian Dröge e6e86c4f2d ext/soundtouch/gstbpmdetect.cc: Clean up a bit and only allocate a temporary buffer for the data if processing stereo...
Original commit message from CVS:
* ext/soundtouch/gstbpmdetect.cc:
Clean up a bit and only allocate a temporary buffer for the data
if processing stereo data as BPMDetect downmixes from stereo to
mono and stores the result in the input data. Thanks to
Stefan Kost for the suggestions.
2008-01-28 11:47:18 +00:00

233 lines
6.6 KiB
C++

/* 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 bpm;
if (G_UNLIKELY (!bpm_detect->priv->detect)) {
if (filter->format.channels == 0 || filter->format.rate == 0) {
GST_ERROR_OBJECT (bpm_detect, "No channels or rate set yet");
return GST_FLOW_ERROR;
}
bpm_detect->priv->detect =
new BPMDetect (filter->format.channels, filter->format.rate);
}
nsamples = GST_BUFFER_SIZE (in) / (4 * filter->format.channels);
/* For stereo BPMDetect->inputSamples() does downmixing into the input
* data but our buffer data shouldn't be modified.
*/
if (filter->format.channels == 1) {
bpm_detect->priv->detect->inputSamples ((gfloat *) GST_BUFFER_DATA (in),
nsamples);
} else {
gfloat *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;
}