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Original commit message from CVS: * configure.ac: * gst/cutter/Makefile.am: * gst/cutter/gstcutter.c: (gst_cutter_class_init), (gst_cutter_init), (gst_cutter_message_new), (gst_cutter_chain), (gst_cutter_set_property), (gst_cutter_get_property), (plugin_init), (gst_cutter_get_caps): port cutter * gst/level/gstlevel.c: fix up plugin details
402 lines
13 KiB
C
402 lines
13 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) 2002,2003,2005
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* Thomas Vander Stichele <thomas at apestaart dot 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include "gstcutter.h"
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#include "math.h"
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GST_DEBUG_CATEGORY (cutter_debug);
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#define GST_CAT_DEFAULT cutter_debug
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#define CUTTER_DEFAULT_THRESHOLD_LEVEL 0.1
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#define CUTTER_DEFAULT_THRESHOLD_LENGTH (500 * GST_MSECOND)
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#define CUTTER_DEFAULT_PRE_LENGTH (200 * GST_MSECOND)
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static GstElementDetails cutter_details = {
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"Cutter",
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"Filter/Editor/Audio",
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"Audio Cutter to split audio into non-silent bits",
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"Thomas <thomas@apestaart.org>",
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};
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static GstStaticPadTemplate cutter_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; "
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GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)
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);
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static GstStaticPadTemplate cutter_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; "
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GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)
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);
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enum
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{
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PROP_0,
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PROP_THRESHOLD,
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PROP_THRESHOLD_DB,
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PROP_RUN_LENGTH,
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PROP_PRE_LENGTH,
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PROP_LEAKY
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};
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GST_BOILERPLATE (GstCutter, gst_cutter, GstElement, GST_TYPE_ELEMENT);
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static void gst_cutter_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_cutter_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstFlowReturn gst_cutter_chain (GstPad * pad, GstBuffer * buffer);
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static double inline gst_cutter_16bit_ms (gint16 * data, guint numsamples);
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static double inline gst_cutter_8bit_ms (gint8 * data, guint numsamples);
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void gst_cutter_get_caps (GstPad * pad, GstCutter * filter);
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static void
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gst_cutter_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&cutter_src_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&cutter_sink_factory));
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gst_element_class_set_details (element_class, &cutter_details);
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}
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static void
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gst_cutter_class_init (GstCutterClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gobject_class->set_property = gst_cutter_set_property;
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gobject_class->get_property = gst_cutter_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_THRESHOLD,
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g_param_spec_double ("threshold", "Threshold",
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"Volume threshold before trigger",
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-G_MAXDOUBLE, G_MAXDOUBLE, 0.0, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_THRESHOLD_DB,
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g_param_spec_double ("threshold-dB", "Threshold (dB)",
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"Volume threshold before trigger (in dB)",
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-G_MAXDOUBLE, G_MAXDOUBLE, 0.0, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_RUN_LENGTH,
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g_param_spec_uint64 ("run-length", "Run length",
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"Length of drop below threshold before cut_stop (in nanoseconds)",
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0, G_MAXUINT64, 0, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PRE_LENGTH,
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g_param_spec_uint64 ("pre-length", "Pre-recording buffer length",
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"Length of pre-recording buffer (in nanoseconds)",
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0, G_MAXUINT64, 0, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LEAKY,
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g_param_spec_boolean ("leaky", "Leaky",
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"do we leak buffers when below threshold ?",
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FALSE, G_PARAM_READWRITE));
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GST_DEBUG_CATEGORY_INIT (cutter_debug, "cutter", 0, "Audio cutting");
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}
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static void
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gst_cutter_init (GstCutter * filter, GstCutterClass * g_class)
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{
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filter->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&cutter_sink_factory), "sink");
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filter->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&cutter_src_factory), "src");
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filter->threshold_level = CUTTER_DEFAULT_THRESHOLD_LEVEL;
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filter->threshold_length = CUTTER_DEFAULT_THRESHOLD_LENGTH;
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filter->silent_run_length = 0 * GST_SECOND;
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filter->silent = TRUE;
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filter->pre_length = CUTTER_DEFAULT_PRE_LENGTH;
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filter->pre_run_length = 0 * GST_SECOND;
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filter->pre_buffer = NULL;
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filter->leaky = FALSE;
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gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad);
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gst_pad_set_chain_function (filter->sinkpad, gst_cutter_chain);
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gst_pad_use_fixed_caps (filter->sinkpad);
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gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad);
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gst_pad_use_fixed_caps (filter->srcpad);
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}
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static GstMessage *
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gst_cutter_message_new (GstCutter * c, gboolean above, GstClockTime timestamp)
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{
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GstStructure *s;
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GValue v = { 0, };
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g_value_init (&v, GST_TYPE_LIST);
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s = gst_structure_new ("cutter",
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"above", G_TYPE_BOOLEAN, above,
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"timestamp", GST_TYPE_CLOCK_TIME, timestamp, NULL);
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return gst_message_new_element (GST_OBJECT (c), s);
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}
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static GstFlowReturn
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gst_cutter_chain (GstPad * pad, GstBuffer * buf)
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{
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GstCutter *filter;
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gint16 *in_data;
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double RMS = 0.0; /* RMS of signal in buffer */
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double ms = 0.0; /* mean square value of buffer */
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static gboolean silent_prev = FALSE; /* previous value of silent */
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GstBuffer *prebuf; /* pointer to a prebuffer element */
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g_return_val_if_fail (pad != NULL, GST_FLOW_ERROR);
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g_return_val_if_fail (GST_IS_PAD (pad), GST_FLOW_ERROR);
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g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
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filter = GST_CUTTER (GST_OBJECT_PARENT (pad));
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g_return_val_if_fail (filter != NULL, GST_FLOW_ERROR);
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g_return_val_if_fail (GST_IS_CUTTER (filter), GST_FLOW_ERROR);
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if (gst_audio_is_buffer_framed (pad, buf) == FALSE) {
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g_warning ("audio buffer is not framed !\n");
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return GST_FLOW_ERROR;
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}
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if (!filter->have_caps)
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gst_cutter_get_caps (pad, filter);
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in_data = (gint16 *) GST_BUFFER_DATA (buf);
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GST_LOG_OBJECT (filter, "length of prerec buffer: %" GST_TIME_FORMAT,
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GST_TIME_ARGS (filter->pre_run_length));
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/* calculate mean square value on buffer */
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switch (filter->width) {
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case 16:
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ms = gst_cutter_16bit_ms (in_data, GST_BUFFER_SIZE (buf) / 2);
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break;
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case 8:
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ms = gst_cutter_8bit_ms ((gint8 *) in_data, GST_BUFFER_SIZE (buf));
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break;
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default:
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/* this shouldn't happen */
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g_warning ("no mean square function for width %d\n", filter->width);
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break;
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}
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silent_prev = filter->silent;
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RMS = sqrt (ms) / (double) filter->max_sample;
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/* if RMS below threshold, add buffer length to silent run length count
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* if not, reset
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*/
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GST_LOG_OBJECT (filter, "buffer stats: ms %f, RMS %f, audio length %f",
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ms, RMS, gst_audio_duration_from_pad_buffer (filter->sinkpad, buf));
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if (RMS < filter->threshold_level)
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filter->silent_run_length +=
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gst_audio_duration_from_pad_buffer (filter->sinkpad, buf);
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else {
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filter->silent_run_length = 0 * GST_SECOND;
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filter->silent = FALSE;
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}
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if (filter->silent_run_length > filter->threshold_length)
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/* it has been silent long enough, flag it */
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filter->silent = TRUE;
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/* has the silent status changed ? if so, send right signal
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* and, if from silent -> not silent, flush pre_record buffer
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*/
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if (filter->silent != silent_prev) {
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if (filter->silent) {
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GstMessage *m =
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gst_cutter_message_new (filter, FALSE, GST_BUFFER_TIMESTAMP (buf));
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GST_DEBUG_OBJECT (filter, "signaling CUT_STOP");
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gst_element_post_message (GST_ELEMENT (filter), m);
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} else {
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gint count = 0;
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GstMessage *m =
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gst_cutter_message_new (filter, TRUE, GST_BUFFER_TIMESTAMP (buf));
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GST_DEBUG_OBJECT (filter, "signaling CUT_START");
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gst_element_post_message (GST_ELEMENT (filter), m);
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/* first of all, flush current buffer */
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GST_DEBUG_OBJECT (filter, "flushing buffer of length %" GST_TIME_FORMAT,
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GST_TIME_ARGS (filter->pre_run_length));
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while (filter->pre_buffer) {
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prebuf = (g_list_first (filter->pre_buffer))->data;
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filter->pre_buffer = g_list_remove (filter->pre_buffer, prebuf);
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gst_pad_push (filter->srcpad, prebuf);
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++count;
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}
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GST_DEBUG_OBJECT (filter, "flushed %d buffers", count);
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filter->pre_run_length = 0 * GST_SECOND;
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}
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}
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/* now check if we have to send the new buffer to the internal buffer cache
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* or to the srcpad */
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if (filter->silent) {
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/* we ref it before putting it in the pre_buffer */
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/* FIXME: we shouldn't probably do this, because the buffer
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* arrives reffed already; the plugin should just push it
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* or unref it to make it disappear */
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/*
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gst_buffer_ref (buf);
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*/
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filter->pre_buffer = g_list_append (filter->pre_buffer, buf);
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filter->pre_run_length +=
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gst_audio_duration_from_pad_buffer (filter->sinkpad, buf);
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while (filter->pre_run_length > filter->pre_length) {
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prebuf = (g_list_first (filter->pre_buffer))->data;
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g_assert (GST_IS_BUFFER (prebuf));
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filter->pre_buffer = g_list_remove (filter->pre_buffer, prebuf);
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filter->pre_run_length -=
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gst_audio_duration_from_pad_buffer (filter->sinkpad, prebuf);
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/* only pass buffers if we don't leak */
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if (!filter->leaky)
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gst_pad_push (filter->srcpad, prebuf);
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}
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} else
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gst_pad_push (filter->srcpad, buf);
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return GST_FLOW_OK;
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}
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static double inline
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gst_cutter_16bit_ms (gint16 * data, guint num_samples)
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#include "filter.func"
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static double inline gst_cutter_8bit_ms (gint8 * data, guint num_samples)
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#include "filter.func"
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static void
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gst_cutter_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstCutter *filter;
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g_return_if_fail (GST_IS_CUTTER (object));
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filter = GST_CUTTER (object);
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switch (prop_id) {
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case PROP_THRESHOLD:
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/* set the level */
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filter->threshold_level = g_value_get_double (value);
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GST_DEBUG ("DEBUG: set threshold level to %f", filter->threshold_level);
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break;
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case PROP_THRESHOLD_DB:
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/* set the level given in dB
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* value in dB = 20 * log (value)
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* values in dB < 0 result in values between 0 and 1
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*/
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filter->threshold_level = pow (10, g_value_get_double (value) / 20);
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GST_DEBUG ("DEBUG: set threshold level to %f", filter->threshold_level);
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break;
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case PROP_RUN_LENGTH:
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/* set the minimum length of the silent run required */
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filter->threshold_length = g_value_get_uint64 (value);
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break;
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case PROP_PRE_LENGTH:
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/* set the length of the pre-record block */
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filter->pre_length = g_value_get_uint64 (value);
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break;
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case PROP_LEAKY:
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/* set if the pre-record buffer is leaky or not */
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filter->leaky = g_value_get_boolean (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_cutter_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstCutter *filter;
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g_return_if_fail (GST_IS_CUTTER (object));
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filter = GST_CUTTER (object);
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switch (prop_id) {
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case PROP_RUN_LENGTH:
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g_value_set_uint64 (value, filter->threshold_length);
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break;
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case PROP_THRESHOLD:
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g_value_set_double (value, filter->threshold_level);
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break;
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case PROP_THRESHOLD_DB:
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g_value_set_double (value, 20 * log (filter->threshold_level));
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break;
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case PROP_PRE_LENGTH:
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g_value_set_uint64 (value, filter->pre_length);
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break;
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case PROP_LEAKY:
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g_value_set_boolean (value, filter->leaky);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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if (!gst_element_register (plugin, "cutter", GST_RANK_NONE, GST_TYPE_CUTTER))
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return FALSE;
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return TRUE;
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}
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GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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"cutter",
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"Audio Cutter to split audio into non-silent bits",
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plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
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void
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gst_cutter_get_caps (GstPad * pad, GstCutter * filter)
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{
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const GstCaps *caps = NULL;
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GstStructure *structure;
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caps = GST_PAD_CAPS (pad);
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/* FIXME : Please change this to a better warning method ! */
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g_assert (caps != NULL);
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (structure, "width", &filter->width);
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filter->max_sample = gst_audio_highest_sample_value (pad);
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filter->have_caps = TRUE;
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}
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