/* 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif /* 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 #include #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; #ifdef HAVE_SOUNDTOUCH_1_4 soundtouch::BPMDetect * detect; #else BPMDetect *detect; #endif }; /* For soundtouch 1.4 */ #if defined(INTEGER_SAMPLES) #define SOUNDTOUCH_INTEGER_SAMPLES 1 #elif defined(FLOAT_SAMPLES) #define SOUNDTOUCH_FLOAT_SAMPLES 1 #endif #if defined(SOUNDTOUCH_FLOAT_SAMPLES) #define ALLOWED_CAPS \ "audio/x-raw, " \ "format = (string) " GST_AUDIO_NE (F32) ", " \ "rate = (int) [ 8000, MAX ], " \ "channels = (int) [ 1, 2 ]" #elif defined(SOUNDTOUCH_INTEGER_SAMPLES) #define ALLOWED_CAPS \ "audio/x-raw, " \ "format = (string) " GST_AUDIO_NE (S16) ", " \ "rate = (int) [ 8000, MAX ], " \ "channels = (int) [ 1, 2 ]" #else #error "Only integer or float samples are supported" #endif #define gst_bpm_detect_parent_class parent_class G_DEFINE_TYPE_WITH_PRIVATE (GstBPMDetect, gst_bpm_detect, GST_TYPE_AUDIO_FILTER); GST_ELEMENT_REGISTER_DEFINE (bpmdetect, "bpmdetect", GST_RANK_NONE, GST_TYPE_BPM_DETECT); 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, const GstAudioInfo * info); static void gst_bpm_detect_class_init (GstBPMDetectClass * klass) { GstCaps *caps; GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_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; gst_element_class_set_static_metadata (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 (klass), caps); gst_caps_unref (caps); trans_class->stop = GST_DEBUG_FUNCPTR (gst_bpm_detect_stop); trans_class->sink_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); } static void gst_bpm_detect_init (GstBPMDetect * bpm_detect) { bpm_detect->priv = (GstBPMDetectPrivate *) gst_bpm_detect_get_instance_private (bpm_detect); 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_SEGMENT: if (bpm_detect->priv->detect) { delete bpm_detect->priv->detect; bpm_detect->priv->detect = NULL; } bpm_detect->bpm = 0.0; break; default: break; } return GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (trans, event); } static gboolean gst_bpm_detect_setup (GstAudioFilter * filter, const GstAudioInfo * info) { 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; GstMapInfo info; if (G_UNLIKELY (!bpm_detect->priv->detect)) { if (GST_AUDIO_INFO_FORMAT (&filter->info) == GST_AUDIO_FORMAT_UNKNOWN) { GST_ERROR_OBJECT (bpm_detect, "No channels or rate set yet"); return GST_FLOW_ERROR; } #ifdef HAVE_SOUNDTOUCH_1_4 bpm_detect->priv->detect = new soundtouch::BPMDetect (GST_AUDIO_INFO_CHANNELS (&filter->info), GST_AUDIO_INFO_RATE (&filter->info)); #else bpm_detect->priv->detect = new BPMDetect (GST_AUDIO_INFO_CHANNELS (&filter->info), GST_AUDIO_INFO_RATE (&filter->info)); #endif } gst_buffer_map (in, &info, GST_MAP_READ); nsamples = info.size / (GST_AUDIO_INFO_BPF (&filter->info) * GST_AUDIO_INFO_CHANNELS (&filter->info)); /* For stereo BPMDetect->inputSamples() does downmixing into the input * data but our buffer data shouldn't be modified. */ if (GST_AUDIO_INFO_CHANNELS (&filter->info) == 1) { soundtouch::SAMPLETYPE *inbuf = (soundtouch::SAMPLETYPE *) info.data; while (nsamples > 0) { bpm_detect->priv->detect->inputSamples (inbuf, MIN (nsamples, 2048)); nsamples -= 2048; inbuf += 2048; } } else { soundtouch::SAMPLETYPE *inbuf, *intmp, data[2 * 2048]; inbuf = (soundtouch::SAMPLETYPE *) info.data; intmp = data; while (nsamples > 0) { memcpy (intmp, inbuf, sizeof (soundtouch::SAMPLETYPE) * 2 * MIN (nsamples, 2048)); bpm_detect->priv->detect->inputSamples (intmp, MIN (nsamples, 2048)); nsamples -= 2048; inbuf += 2048 * 2; } } gst_buffer_unmap (in, &info); bpm = bpm_detect->priv->detect->getBpm (); if (bpm >= 1.0 && fabs (bpm_detect->bpm - bpm) >= 1.0) { GstTagList *tags = gst_tag_list_new_empty (); gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE_ALL, GST_TAG_BEATS_PER_MINUTE, bpm, (void *) NULL); gst_pad_push_event (trans->srcpad, gst_event_new_tag (tags)); GST_INFO_OBJECT (bpm_detect, "Detected BPM: %lf", bpm); bpm_detect->bpm = bpm; } return GST_FLOW_OK; }