From 27fbdb97d126d2b29611778362e73b03dc75e472 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Sun, 27 Jan 2008 05:56:04 +0000 Subject: ext/soundtouch/: Add BPM detection plugin based on SoundTouch's libBPM. Original commit message from CVS: * ext/soundtouch/Makefile.am: * ext/soundtouch/gstbpmdetect.cc: * ext/soundtouch/gstbpmdetect.hh: * ext/soundtouch/plugin.c: (plugin_init): Add BPM detection plugin based on SoundTouch's libBPM. * ext/soundtouch/gstpitch.cc: Allow sample rates until MAX instead of only 48kHz and remove the buffer-frames field from that caps. Clear the remaining samples completely when necessary to get into a clean state again. --- ext/soundtouch/gstbpmdetect.cc | 223 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 223 insertions(+) create mode 100644 ext/soundtouch/gstbpmdetect.cc (limited to 'ext/soundtouch/gstbpmdetect.cc') diff --git a/ext/soundtouch/gstbpmdetect.cc b/ext/soundtouch/gstbpmdetect.cc new file mode 100644 index 00000000..e9e2e274 --- /dev/null +++ b/ext/soundtouch/gstbpmdetect.cc @@ -0,0 +1,223 @@ +/* GStreamer + * Copyright (C) 2008 Sebastian Dröge + * + * 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 +/* 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 +#include +#include +#include +#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 "); + + 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 != 0.0 && 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; +} -- cgit v1.2.1