mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-24 02:31:03 +00:00
1e418567fa
Original commit message from CVS: made changes everywhere to accomodate for the headers being in <gst/(lib)/...> we'll need to conclude this fast because we will also need to change stuff in core real soon for the libs in order to fix everything and I can't do it right now because I disabled all of the plugins here ;)
359 lines
9.6 KiB
C
359 lines
9.6 KiB
C
/* Gnome-Streamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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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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#include <string.h>
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#include <math.h>
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//#define DEBUG_ENABLED
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#include <gstaudioscale.h>
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#include <gst/audio/audio.h>
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#include <gst/resample/resample.h>
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/* elementfactory information */
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static GstElementDetails audioscale_details = {
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"Audio scaler",
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"Filter/Audio/Scaler",
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"Resizes audio",
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VERSION,
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"Wim Taymans <wim.taymans@chello.be>",
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"(C) 2000",
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};
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/* Audioscale signals and args */
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enum {
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/* FILL ME */
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LAST_SIGNAL
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};
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enum {
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ARG_0,
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ARG_FREQUENCY,
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ARG_FILTERLEN,
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ARG_METHOD,
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/* FILL ME */
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};
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static GstPadTemplate *
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sink_template (void)
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{
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static GstPadTemplate *template = NULL;
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if (!template) {
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template = gst_padtemplate_new ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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gst_caps_new
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("audioscale_sink",
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"audio/raw", GST_AUDIO_INT_PAD_TEMPLATE_PROPS), NULL);
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}
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return template;
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}
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static GstPadTemplate *
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src_template (void)
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{
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static GstPadTemplate *template = NULL;
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if (!template) {
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template = gst_padtemplate_new ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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gst_caps_new
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("audioscale_src",
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"audio/raw", GST_AUDIO_INT_PAD_TEMPLATE_PROPS), NULL);
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}
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return template;
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}
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#define GST_TYPE_AUDIOSCALE_METHOD (gst_audioscale_method_get_type())
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static GType
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gst_audioscale_method_get_type (void)
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{
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static GType audioscale_method_type = 0;
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static GEnumValue audioscale_methods[] = {
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{ GST_AUDIOSCALE_NEAREST, "0", "Nearest" },
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{ GST_AUDIOSCALE_BILINEAR, "1", "Bilinear" },
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{ GST_AUDIOSCALE_SINC, "2", "Sinc" },
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{ 0, NULL, NULL },
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};
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if(!audioscale_method_type){
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audioscale_method_type = g_enum_register_static("GstAudioscaleMethod",
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audioscale_methods);
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}
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return audioscale_method_type;
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}
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static void gst_audioscale_class_init (AudioscaleClass *klass);
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static void gst_audioscale_init (Audioscale *audioscale);
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static void gst_audioscale_chain (GstPad *pad, GstBuffer *buf);
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static void gst_audioscale_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audioscale_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstElementClass *parent_class = NULL;
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//static guint gst_audioscale_signals[LAST_SIGNAL] = { 0 };
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GType
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audioscale_get_type (void)
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{
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static GType audioscale_type = 0;
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if (!audioscale_type) {
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static const GTypeInfo audioscale_info = {
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sizeof(AudioscaleClass), NULL,
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NULL,
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(GClassInitFunc)gst_audioscale_class_init,
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NULL,
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NULL,
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sizeof(Audioscale),
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0,
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(GInstanceInitFunc)gst_audioscale_init,
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};
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audioscale_type = g_type_register_static(GST_TYPE_ELEMENT, "Audioscale", &audioscale_info, 0);
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}
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return audioscale_type;
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}
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static void
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gst_audioscale_class_init (AudioscaleClass *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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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FREQUENCY,
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g_param_spec_int("frequency","frequency","frequency",
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G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); // CHECKME
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FILTERLEN,
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g_param_spec_int ("filter_length", "filter_length", "filter_length",
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G_MININT, G_MAXINT, 0, G_PARAM_READWRITE)); // CHECKME
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_METHOD,
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g_param_spec_int ("method", "method", "method",
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G_MININT, G_MAXINT, 0, G_PARAM_READWRITE)); // CHECKME
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parent_class = g_type_class_ref(GST_TYPE_ELEMENT);
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gobject_class->set_property = gst_audioscale_set_property;
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gobject_class->get_property = gst_audioscale_get_property;
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}
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static GstPadNegotiateReturn
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audioscale_negotiate_src (GstPad * pad, GstCaps ** caps, gpointer * data)
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{
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Audioscale *audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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g_print ("audioscale_negotiate_src\n");
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if (*caps == NULL)
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return GST_PAD_NEGOTIATE_FAIL;
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*caps = gst_caps_copy_on_write (*caps);
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gst_caps_set (*caps, "rate", GST_PROPS_INT_RANGE (8000, 48000));
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return gst_pad_negotiate_proxy (pad, audioscale->sinkpad, caps);
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}
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static GstPadNegotiateReturn
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audioscale_negotiate_sink (GstPad * pad, GstCaps ** caps, gpointer * data)
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{
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Audioscale *audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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g_print ("audioscale_negotiate_sink\n");
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if (*caps == NULL)
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return GST_PAD_NEGOTIATE_FAIL;
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*caps = gst_caps_copy_on_write (*caps);
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gst_caps_set (*caps, "rate", GST_PROPS_INT (audioscale->targetfrequency));
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return gst_pad_negotiate_proxy (pad, audioscale->srcpad, caps);
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}
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static void
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gst_audioscale_newcaps (GstPad * pad, GstCaps * caps)
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{
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Audioscale *audioscale;
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resample_t *r;
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audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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r = audioscale->resample;
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r->i_rate = gst_caps_get_int (caps, "rate");
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r->channels = gst_caps_get_int (caps, "channels");
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resample_reinit(r);
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//g_print("audioscale: unsupported scaling method %d\n", audioscale->method);
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}
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static void *
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gst_audioscale_get_buffer (void *priv, unsigned int size)
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{
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Audioscale * audioscale = priv;
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audioscale->outbuf = gst_buffer_new();
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GST_BUFFER_SIZE(audioscale->outbuf) = size;
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GST_BUFFER_DATA(audioscale->outbuf) = g_malloc(size);
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return GST_BUFFER_DATA(audioscale->outbuf);
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}
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static void
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gst_audioscale_init (Audioscale *audioscale)
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{
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resample_t *r;
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audioscale->sinkpad = gst_pad_new_from_template (GST_PADTEMPLATE_GET (sink_template), "sink");
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gst_pad_set_negotiate_function (audioscale->sinkpad, audioscale_negotiate_sink);
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gst_element_add_pad(GST_ELEMENT(audioscale),audioscale->sinkpad);
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gst_pad_set_chain_function(audioscale->sinkpad,gst_audioscale_chain);
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gst_pad_set_newcaps_function (audioscale->sinkpad, gst_audioscale_newcaps);
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audioscale->srcpad = gst_pad_new_from_template (GST_PADTEMPLATE_GET (src_template), "src");
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gst_pad_set_negotiate_function (audioscale->srcpad, audioscale_negotiate_src);
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gst_element_add_pad(GST_ELEMENT(audioscale),audioscale->srcpad);
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r = g_new0(resample_t,1);
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audioscale->resample = r;
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r->priv = audioscale;
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r->get_buffer = gst_audioscale_get_buffer;
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r->method = RESAMPLE_SINC;
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r->channels = 0;
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r->filter_length = 16;
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r->i_rate = -1;
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r->o_rate = -1;
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//r->verbose = 1;
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resample_init(r);
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}
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static void
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gst_audioscale_chain (GstPad *pad, GstBuffer *buf)
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{
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Audioscale *audioscale;
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guchar *data;
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gulong size;
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g_return_if_fail(pad != NULL);
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g_return_if_fail(GST_IS_PAD(pad));
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g_return_if_fail(buf != NULL);
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audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
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data = GST_BUFFER_DATA(buf);
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size = GST_BUFFER_SIZE(buf);
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GST_DEBUG (0,
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"gst_audioscale_chain: got buffer of %ld bytes in '%s'\n",
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size, gst_element_get_name (GST_ELEMENT (audioscale)));
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resample_scale (audioscale->resample, data, size);
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gst_pad_push (audioscale->srcpad, audioscale->outbuf);
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gst_buffer_unref (buf);
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}
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static void
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gst_audioscale_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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Audioscale *src;
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resample_t *r;
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/* it's not null if we got it, but it might not be ours */
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g_return_if_fail(GST_IS_AUDIOSCALE(object));
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src = GST_AUDIOSCALE(object);
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r = src->resample;
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switch (prop_id) {
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case ARG_FREQUENCY:
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src->targetfrequency = g_value_get_int (value);
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r->o_rate = src->targetfrequency;
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break;
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case ARG_FILTERLEN:
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r->filter_length = g_value_get_int (value);
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g_print ("new filter length %d\n", r->filter_length);
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break;
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case ARG_METHOD:
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r->method = g_value_get_int (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_audioscale_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
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{
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Audioscale *src;
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resample_t *r;
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/* it's not null if we got it, but it might not be ours */
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g_return_if_fail(GST_IS_AUDIOSCALE(object));
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src = GST_AUDIOSCALE(object);
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r = src->resample;
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switch (prop_id) {
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case ARG_FREQUENCY:
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g_value_set_int (value, src->targetfrequency);
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break;
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case ARG_FILTERLEN:
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g_value_set_int (value, r->filter_length);
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break;
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case ARG_METHOD:
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g_value_set_int (value, r->method);
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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 (GModule *module, GstPlugin *plugin)
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{
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GstElementFactory *factory;
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/* create an elementfactory for the audioscale element */
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factory = gst_elementfactory_new ("audioscale", GST_TYPE_AUDIOSCALE, &audioscale_details);
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g_return_val_if_fail(factory != NULL, FALSE);
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gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
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return TRUE;
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}
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GstPluginDesc plugin_desc = {
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GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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"audioscale",
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plugin_init
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};
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