mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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671eef9b9d
Original commit message from CVS: reverting error patch before making a branch.
412 lines
12 KiB
C
412 lines
12 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000 Wim Taymans <wtay@chello.be>
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*
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* gstafsrc.c:
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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 "gstafsrc.h"
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/* elementfactory information */
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static GstElementDetails afsrc_details = {
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"Audiofile Src",
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"Source/Audio",
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"LGPL",
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"Read audio files from disk using libaudiofile",
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VERSION,
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"Thomas <thomas@apestaart.org>",
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"(C) 2001"
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};
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/* AFSrc signals and args */
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enum {
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/* FILL ME */
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SIGNAL_HANDOFF,
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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_LOCATION
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};
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/* added a src factory function to force audio/raw MIME type */
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/* I think the caps can be broader, we need to change that somehow */
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GST_PAD_TEMPLATE_FACTORY (afsrc_src_factory,
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"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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"audiofile_src",
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"audio/x-raw-int",
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"endianness", GST_PROPS_INT (G_BYTE_ORDER),
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"signed", GST_PROPS_LIST (
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GST_PROPS_BOOLEAN (TRUE),
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GST_PROPS_BOOLEAN (FALSE)
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),
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"width", GST_PROPS_INT_RANGE (8, 16),
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"depth", GST_PROPS_INT_RANGE (8, 16),
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"rate", GST_PROPS_INT_RANGE (4000, 48000), /*FIXME*/
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"channels", GST_PROPS_INT_RANGE (1, 2)
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)
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);
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/* we use an enum for the output type arg */
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#define GST_TYPE_AFSRC_TYPES (gst_afsrc_types_get_type())
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/* FIXME: fix the string ints to be string-converted from the audiofile.h types */
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/* defined but not used
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static GType
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gst_afsrc_types_get_type (void)
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{
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static GType afsrc_types_type = 0;
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static GEnumValue afsrc_types[] = {
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{AF_FILE_RAWDATA, "0", "raw PCM"},
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{AF_FILE_AIFFC, "1", "AIFFC"},
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{AF_FILE_AIFF, "2", "AIFF"},
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{AF_FILE_NEXTSND, "3", "Next/SND"},
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{AF_FILE_WAVE, "4", "Wave"},
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{0, NULL, NULL},
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};
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if (!afsrc_types_type)
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{
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afsrc_types_type = g_enum_register_static ("GstAudiosrcTypes", afsrc_types);
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}
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return afsrc_types_type;
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}
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*/
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static void gst_afsrc_class_init (GstAFSrcClass *klass);
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static void gst_afsrc_init (GstAFSrc *afsrc);
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static gboolean gst_afsrc_open_file (GstAFSrc *src);
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static void gst_afsrc_close_file (GstAFSrc *src);
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static GstBuffer* gst_afsrc_get (GstPad *pad);
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static void gst_afsrc_set_property (GObject *object, guint prop_id,
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const GValue *value, GParamSpec *pspec);
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static void gst_afsrc_get_property (GObject *object, guint prop_id,
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GValue *value, GParamSpec *pspec);
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static GstElementStateReturn gst_afsrc_change_state (GstElement *element);
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static GstElementClass *parent_class = NULL;
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static guint gst_afsrc_signals[LAST_SIGNAL] = { 0 };
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GType
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gst_afsrc_get_type (void)
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{
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static GType afsrc_type = 0;
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if (!afsrc_type) {
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static const GTypeInfo afsrc_info = {
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sizeof (GstAFSrcClass), NULL,
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NULL,
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(GClassInitFunc) gst_afsrc_class_init,
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NULL,
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NULL,
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sizeof (GstAFSrc),
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0,
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(GInstanceInitFunc) gst_afsrc_init,
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};
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afsrc_type = g_type_register_static (GST_TYPE_ELEMENT, "GstAFSrc", &afsrc_info, 0);
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}
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return afsrc_type;
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}
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static void
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gst_afsrc_class_init (GstAFSrcClass *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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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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gst_element_class_install_std_props (
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GST_ELEMENT_CLASS (klass),
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"location", ARG_LOCATION, G_PARAM_READWRITE,
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NULL);
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gst_afsrc_signals[SIGNAL_HANDOFF] =
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g_signal_new ("handoff", G_TYPE_FROM_CLASS(klass), G_SIGNAL_RUN_LAST,
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G_STRUCT_OFFSET (GstAFSrcClass, handoff), NULL, NULL,
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g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0);
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gobject_class->set_property = gst_afsrc_set_property;
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gobject_class->get_property = gst_afsrc_get_property;
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gstelement_class->change_state = gst_afsrc_change_state;
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}
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static void
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gst_afsrc_init (GstAFSrc *afsrc)
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{
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/* no need for a template, caps are set based on file, right ? */
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afsrc->srcpad = gst_pad_new_from_template (afsrc_src_factory (), "src");
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gst_element_add_pad (GST_ELEMENT (afsrc), afsrc->srcpad);
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gst_pad_set_get_function (afsrc->srcpad, gst_afsrc_get);
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afsrc->bytes_per_read = 4096;
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afsrc->curoffset = 0;
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afsrc->seq = 0;
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afsrc->filename = NULL;
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afsrc->file = NULL;
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/* default values, should never be needed */
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afsrc->channels = 2;
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afsrc->width = 16;
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afsrc->rate = 44100;
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afsrc->type = AF_FILE_WAVE;
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afsrc->endianness_data = 1234;
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afsrc->endianness_wanted = 1234;
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afsrc->framestamp = 0;
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}
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static GstBuffer *
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gst_afsrc_get (GstPad *pad)
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{
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GstAFSrc *src;
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GstBuffer *buf;
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glong readbytes, readframes;
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glong frameCount;
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g_return_val_if_fail (pad != NULL, NULL);
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src = GST_AFSRC (gst_pad_get_parent (pad));
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buf = gst_buffer_new ();
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g_return_val_if_fail (buf, NULL);
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GST_BUFFER_DATA (buf) = (gpointer) g_malloc (src->bytes_per_read);
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/* calculate frameCount to read based on file info */
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frameCount = src->bytes_per_read / (src->channels * src->width / 8);
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/* g_print ("DEBUG: gstafsrc: going to read %ld frames\n", frameCount); */
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readframes = afReadFrames (src->file, AF_DEFAULT_TRACK, GST_BUFFER_DATA (buf),
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frameCount);
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readbytes = readframes * (src->channels * src->width / 8);
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if (readbytes == 0) {
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gst_element_set_eos (GST_ELEMENT (src));
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return GST_BUFFER (gst_event_new (GST_EVENT_EOS));
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}
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GST_BUFFER_SIZE (buf) = readbytes;
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GST_BUFFER_OFFSET (buf) = src->curoffset;
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src->curoffset += readbytes;
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src->framestamp += gst_audio_frame_length (src->srcpad, buf);
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GST_BUFFER_TIMESTAMP (buf) = src->framestamp * 1E9
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/ gst_audio_frame_rate (src->srcpad);
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printf ("DEBUG: afsrc: timestamp set on output buffer: %f sec\n",
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GST_BUFFER_TIMESTAMP (buf) / 1E9);
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/* g_print("DEBUG: gstafsrc: pushed buffer of %ld bytes\n", readbytes); */
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return buf;
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}
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static void
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gst_afsrc_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
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{
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GstAFSrc *src;
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/* it's not null if we got it, but it might not be ours */
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src = GST_AFSRC (object);
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switch (prop_id) {
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case ARG_LOCATION:
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if (src->filename)
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g_free (src->filename);
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src->filename = g_strdup (g_value_get_string (value));
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break;
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default:
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break;
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}
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}
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static void
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gst_afsrc_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
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{
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GstAFSrc *src;
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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_AFSRC (object));
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src = GST_AFSRC (object);
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switch (prop_id) {
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case ARG_LOCATION:
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g_value_set_string (value, src->filename);
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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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gboolean
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gst_afsrc_plugin_init (GModule *module, GstPlugin *plugin)
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{
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GstElementFactory *factory;
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factory = gst_element_factory_new ("afsrc", GST_TYPE_AFSRC,
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&afsrc_details);
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g_return_val_if_fail (factory != NULL, FALSE);
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gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (afsrc_src_factory));
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gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
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/* load audio support library */
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if (!gst_library_load ("gstaudio"))
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return FALSE;
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return TRUE;
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}
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/* this is where we open the audiofile */
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static gboolean
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gst_afsrc_open_file (GstAFSrc *src)
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{
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g_return_val_if_fail (!GST_FLAG_IS_SET (src, GST_AFSRC_OPEN), FALSE);
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/* open the file */
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src->file = afOpenFile (src->filename, "r", AF_NULL_FILESETUP);
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if (src->file == AF_NULL_FILEHANDLE)
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{
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g_print ("ERROR: gstafsrc: Could not open file %s for reading\n",
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src->filename);
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gst_element_error (GST_ELEMENT (src), g_strconcat ("opening file \"",
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src->filename, "\"", NULL));
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return FALSE;
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}
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/* get the audiofile audio parameters */
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{
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int sampleFormat, sampleWidth;
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src->channels = afGetChannels (src->file, AF_DEFAULT_TRACK);
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afGetSampleFormat (src->file, AF_DEFAULT_TRACK,
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&sampleFormat, &sampleWidth);
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switch (sampleFormat)
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{
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case AF_SAMPFMT_TWOSCOMP:
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src->is_signed = TRUE;
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break;
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case AF_SAMPFMT_UNSIGNED:
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src->is_signed = FALSE;
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break;
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case AF_SAMPFMT_FLOAT:
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case AF_SAMPFMT_DOUBLE:
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GST_DEBUG (
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"ERROR: float data not supported yet !\n");
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}
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src->rate = (guint) afGetRate (src->file, AF_DEFAULT_TRACK);
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src->width = sampleWidth;
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GST_DEBUG (
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"input file: %d channels, %d width, %d rate, signed %s\n",
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src->channels, src->width, src->rate,
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src->is_signed ? "yes" : "no");
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}
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/* set caps on src */
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/*FIXME: add all the possible formats, especially float ! */
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gst_pad_try_set_caps (src->srcpad,
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GST_CAPS_NEW (
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"af_src",
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"audio/x-raw-int",
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"endianness", GST_PROPS_INT (G_BYTE_ORDER), /*FIXME */
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"signed", GST_PROPS_BOOLEAN (src->is_signed),
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"width", GST_PROPS_INT (src->width),
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"depth", GST_PROPS_INT (src->width),
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"rate", GST_PROPS_INT (src->rate),
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"channels", GST_PROPS_INT (src->channels)
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)
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);
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GST_FLAG_SET (src, GST_AFSRC_OPEN);
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return TRUE;
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}
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static void
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gst_afsrc_close_file (GstAFSrc *src)
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{
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/* g_print ("DEBUG: closing srcfile...\n"); */
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g_return_if_fail (GST_FLAG_IS_SET (src, GST_AFSRC_OPEN));
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/* g_print ("DEBUG: past flag test\n"); */
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/* if (fclose (src->file) != 0) */
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if (afCloseFile (src->file) != 0)
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{
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g_print ("WARNING: afsrc: oops, error closing !\n");
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perror ("close");
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gst_element_error (GST_ELEMENT (src), g_strconcat("closing file \"", src->filename, "\"", NULL));
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}
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else {
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GST_FLAG_UNSET (src, GST_AFSRC_OPEN);
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}
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}
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static GstElementStateReturn
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gst_afsrc_change_state (GstElement *element)
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{
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g_return_val_if_fail (GST_IS_AFSRC (element), GST_STATE_FAILURE);
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/* if going to NULL then close the file */
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if (GST_STATE_PENDING (element) == GST_STATE_NULL)
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{
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/* printf ("DEBUG: afsrc state change: null pending\n"); */
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if (GST_FLAG_IS_SET (element, GST_AFSRC_OPEN))
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{
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/* g_print ("DEBUG: trying to close the src file\n"); */
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gst_afsrc_close_file (GST_AFSRC (element));
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}
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}
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else if (GST_STATE_PENDING (element) == GST_STATE_READY)
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{
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/* g_print ("DEBUG: afsrc: ready state pending. This shouldn't happen at the *end* of a stream\n"); */
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if (!GST_FLAG_IS_SET (element, GST_AFSRC_OPEN))
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{
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/* g_print ("DEBUG: GST_AFSRC_OPEN not set\n"); */
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if (!gst_afsrc_open_file (GST_AFSRC (element)))
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{
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/* g_print ("DEBUG: element tries to open file\n"); */
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return GST_STATE_FAILURE;
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}
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}
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}
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if (GST_ELEMENT_CLASS (parent_class)->change_state)
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return GST_ELEMENT_CLASS (parent_class)->change_state (element);
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return GST_STATE_SUCCESS;
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}
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