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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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9d703439f1
Add pitch tests with different forward and backward playback rates. Those tests depend on the libSoundTouch version to validate the buffers checksums. The actual version uses libSoundTouch 2.3.2, use the `--force-fallback-for=soundtouch` meson option to build using the same version. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6247>
268 lines
7.7 KiB
C++
268 lines
7.7 KiB
C++
/* GStreamer
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* Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, 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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/* FIXME: workaround for SoundTouch.h of version 1.3.1 defining those
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* variables while it shouldn't. */
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#undef VERSION
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#undef PACKAGE_VERSION
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#undef PACKAGE_TARNAME
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#undef PACKAGE_STRING
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#undef PACKAGE_NAME
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#undef PACKAGE_BUGREPORT
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#undef PACKAGE
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#include <BPMDetect.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/gstaudiofilter.h>
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#include <math.h>
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#include <string.h>
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#include "gstbpmdetect.hh"
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GST_DEBUG_CATEGORY_STATIC (gst_bpm_detect_debug);
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#define GST_CAT_DEFAULT gst_bpm_detect_debug
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#define GST_BPM_DETECT_GET_PRIVATE(o) (o->priv)
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struct _GstBPMDetectPrivate
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{
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gfloat bpm;
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#ifdef HAVE_SOUNDTOUCH_1_4
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soundtouch::BPMDetect * detect;
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#else
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BPMDetect *detect;
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#endif
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};
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/* For soundtouch 1.4 */
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#if defined(INTEGER_SAMPLES)
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#define SOUNDTOUCH_INTEGER_SAMPLES 1
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#elif defined(FLOAT_SAMPLES)
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#define SOUNDTOUCH_FLOAT_SAMPLES 1
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#endif
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#if defined(SOUNDTOUCH_FLOAT_SAMPLES)
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#define ALLOWED_CAPS \
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"audio/x-raw, " \
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"format = (string) " GST_AUDIO_NE (F32) ", " \
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"rate = (int) [ 8000, MAX ], " \
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"channels = (int) [ 1, 2 ]"
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#elif defined(SOUNDTOUCH_INTEGER_SAMPLES)
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#define ALLOWED_CAPS \
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"audio/x-raw, " \
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"format = (string) " GST_AUDIO_NE (S16) ", " \
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"rate = (int) [ 8000, MAX ], " \
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"channels = (int) [ 1, 2 ]"
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#else
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#error "Only integer or float samples are supported"
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#endif
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#define gst_bpm_detect_parent_class parent_class
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G_DEFINE_TYPE_WITH_PRIVATE (GstBPMDetect, gst_bpm_detect, GST_TYPE_AUDIO_FILTER);
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GST_ELEMENT_REGISTER_DEFINE (bpmdetect, "bpmdetect", GST_RANK_NONE,
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GST_TYPE_BPM_DETECT);
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static void gst_bpm_detect_finalize (GObject * object);
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static gboolean gst_bpm_detect_stop (GstBaseTransform * trans);
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static gboolean gst_bpm_detect_event (GstBaseTransform * trans,
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GstEvent * event);
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static GstFlowReturn gst_bpm_detect_transform_ip (GstBaseTransform * trans,
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GstBuffer * in);
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static gboolean gst_bpm_detect_setup (GstAudioFilter * filter,
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const GstAudioInfo * info);
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static void
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gst_bpm_detect_class_init (GstBPMDetectClass * klass)
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{
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GstCaps *caps;
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstBaseTransformClass *trans_class = GST_BASE_TRANSFORM_CLASS (klass);
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GstAudioFilterClass *filter_class = GST_AUDIO_FILTER_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (gst_bpm_detect_debug, "bpm_detect", 0,
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"audio bpm detection element");
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gobject_class->finalize = gst_bpm_detect_finalize;
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gst_element_class_set_static_metadata (element_class, "BPM Detector",
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"Filter/Analyzer/Audio", "Detect the BPM of an audio stream",
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"Sebastian Dröge <slomo@circular-chaos.org>");
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caps = gst_caps_from_string (ALLOWED_CAPS);
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gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
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caps);
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gst_caps_unref (caps);
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trans_class->stop = GST_DEBUG_FUNCPTR (gst_bpm_detect_stop);
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trans_class->sink_event = GST_DEBUG_FUNCPTR (gst_bpm_detect_event);
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trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_bpm_detect_transform_ip);
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trans_class->passthrough_on_same_caps = TRUE;
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filter_class->setup = GST_DEBUG_FUNCPTR (gst_bpm_detect_setup);
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}
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static void
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gst_bpm_detect_init (GstBPMDetect * bpm_detect)
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{
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bpm_detect->priv =
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(GstBPMDetectPrivate *) gst_bpm_detect_get_instance_private (bpm_detect);
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bpm_detect->priv->detect = NULL;
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bpm_detect->bpm = 0.0;
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}
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static void
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gst_bpm_detect_finalize (GObject * object)
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{
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GstBPMDetect *bpm_detect = GST_BPM_DETECT (object);
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if (bpm_detect->priv->detect) {
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delete bpm_detect->priv->detect;
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bpm_detect->priv->detect = NULL;
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}
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_bpm_detect_stop (GstBaseTransform * trans)
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{
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GstBPMDetect *bpm_detect = GST_BPM_DETECT (trans);
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if (bpm_detect->priv->detect) {
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delete bpm_detect->priv->detect;
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bpm_detect->priv->detect = NULL;
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}
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bpm_detect->bpm = 0.0;
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return TRUE;
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}
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static gboolean
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gst_bpm_detect_event (GstBaseTransform * trans, GstEvent * event)
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{
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GstBPMDetect *bpm_detect = GST_BPM_DETECT (trans);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_FLUSH_STOP:
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case GST_EVENT_EOS:
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case GST_EVENT_SEGMENT:
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if (bpm_detect->priv->detect) {
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delete bpm_detect->priv->detect;
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bpm_detect->priv->detect = NULL;
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}
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bpm_detect->bpm = 0.0;
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break;
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default:
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break;
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}
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return GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (trans, event);
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}
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static gboolean
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gst_bpm_detect_setup (GstAudioFilter * filter, const GstAudioInfo * info)
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{
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GstBPMDetect *bpm_detect = GST_BPM_DETECT (filter);
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if (bpm_detect->priv->detect) {
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delete bpm_detect->priv->detect;
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bpm_detect->priv->detect = NULL;
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}
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return TRUE;
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}
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static GstFlowReturn
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gst_bpm_detect_transform_ip (GstBaseTransform * trans, GstBuffer * in)
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{
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GstBPMDetect *bpm_detect = GST_BPM_DETECT (trans);
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GstAudioFilter *filter = GST_AUDIO_FILTER (trans);
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gint nsamples;
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gfloat bpm;
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GstMapInfo info;
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if (G_UNLIKELY (!bpm_detect->priv->detect)) {
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if (GST_AUDIO_INFO_FORMAT (&filter->info) == GST_AUDIO_FORMAT_UNKNOWN) {
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GST_ERROR_OBJECT (bpm_detect, "No channels or rate set yet");
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return GST_FLOW_ERROR;
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}
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#ifdef HAVE_SOUNDTOUCH_1_4
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bpm_detect->priv->detect =
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new soundtouch::BPMDetect (GST_AUDIO_INFO_CHANNELS (&filter->info),
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GST_AUDIO_INFO_RATE (&filter->info));
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#else
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bpm_detect->priv->detect =
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new BPMDetect (GST_AUDIO_INFO_CHANNELS (&filter->info),
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GST_AUDIO_INFO_RATE (&filter->info));
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#endif
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}
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gst_buffer_map (in, &info, GST_MAP_READ);
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nsamples = info.size / (GST_AUDIO_INFO_BPF (&filter->info) * GST_AUDIO_INFO_CHANNELS (&filter->info));
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/* For stereo BPMDetect->inputSamples() does downmixing into the input
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* data but our buffer data shouldn't be modified.
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*/
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if (GST_AUDIO_INFO_CHANNELS (&filter->info) == 1) {
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soundtouch::SAMPLETYPE *inbuf = (soundtouch::SAMPLETYPE *) info.data;
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while (nsamples > 0) {
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bpm_detect->priv->detect->inputSamples (inbuf, MIN (nsamples, 2048));
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nsamples -= 2048;
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inbuf += 2048;
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}
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} else {
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soundtouch::SAMPLETYPE *inbuf, *intmp, data[2 * 2048];
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inbuf = (soundtouch::SAMPLETYPE *) info.data;
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intmp = data;
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while (nsamples > 0) {
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memcpy (intmp, inbuf, sizeof (soundtouch::SAMPLETYPE) * 2 * MIN (nsamples, 2048));
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bpm_detect->priv->detect->inputSamples (intmp, MIN (nsamples, 2048));
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nsamples -= 2048;
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inbuf += 2048 * 2;
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}
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}
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gst_buffer_unmap (in, &info);
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bpm = bpm_detect->priv->detect->getBpm ();
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if (bpm >= 1.0 && fabs (bpm_detect->bpm - bpm) >= 1.0) {
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GstTagList *tags = gst_tag_list_new_empty ();
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gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE_ALL, GST_TAG_BEATS_PER_MINUTE,
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bpm, (void *) NULL);
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gst_pad_push_event (trans->srcpad, gst_event_new_tag (tags));
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GST_INFO_OBJECT (bpm_detect, "Detected BPM: %lf", bpm);
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bpm_detect->bpm = bpm;
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}
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return GST_FLOW_OK;
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}
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