mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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ab8b696754
Original commit message from CVS: Convert elements to use gst_pad_use_explicit_caps() where appropriate.
524 lines
15 KiB
C
524 lines
15 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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* gstafparse.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 <string.h>
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#include "gstafparse.h"
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/* elementfactory information */
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static GstElementDetails afparse_details = {
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"Audiofile Parse",
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"Codec/Parser/Audio",
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"Audiofile parser for audio/raw",
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"Steve Baker <stevebaker_org@yahoo.co.uk>",
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};
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/* AFParse 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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};
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/* added a src factory function to force audio/raw MIME type */
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static GstStaticPadTemplate afparse_src_factory =
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GST_STATIC_PAD_TEMPLATE (
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"src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (
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"audio/x-raw-int, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ], "
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"endianness = (int) BYTE_ORDER, "
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"width = (int) { 8, 16 }, "
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"depth = (int) { 8, 16 }, "
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"signed = (boolean) { true, false }, "
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"buffer-frames = (int) [ 1, MAX ]"
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)
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);
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static GstStaticPadTemplate afparse_sink_factory =
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GST_STATIC_PAD_TEMPLATE (
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"sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (
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"audio/x-aiff; "
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"audio/x-wav; "
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"audio/x-au"
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)
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);
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static void gst_afparse_base_init (gpointer g_class);
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static void gst_afparse_class_init(GstAFParseClass *klass);
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static void gst_afparse_init (GstAFParse *afparse);
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static gboolean gst_afparse_open_file(GstAFParse *afparse);
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static void gst_afparse_close_file(GstAFParse *afparse);
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static void gst_afparse_loop(GstElement *element);
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static void gst_afparse_set_property(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec);
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static void gst_afparse_get_property(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
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static ssize_t gst_afparse_vf_read (AFvirtualfile *vfile, void *data, size_t nbytes);
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static long gst_afparse_vf_length (AFvirtualfile *vfile);
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static ssize_t gst_afparse_vf_write (AFvirtualfile *vfile, const void *data, size_t nbytes);
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static void gst_afparse_vf_destroy(AFvirtualfile *vfile);
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static long gst_afparse_vf_seek (AFvirtualfile *vfile, long offset, int is_relative);
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static long gst_afparse_vf_tell (AFvirtualfile *vfile);
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GType
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gst_afparse_get_type (void)
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{
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static GType afparse_type = 0;
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if (!afparse_type) {
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static const GTypeInfo afparse_info = {
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sizeof (GstAFParseClass),
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gst_afparse_base_init,
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NULL,
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(GClassInitFunc) gst_afparse_class_init,
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NULL,
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NULL,
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sizeof (GstAFParse),
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0,
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(GInstanceInitFunc) gst_afparse_init,
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};
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afparse_type = g_type_register_static (GST_TYPE_ELEMENT, "GstAFParse", &afparse_info, 0);
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}
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return afparse_type;
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}
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static void
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gst_afparse_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 (&afparse_src_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&afparse_sink_factory));
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gst_element_class_set_details (element_class, &afparse_details);
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}
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static void
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gst_afparse_class_init (GstAFParseClass *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_afparse_set_property;
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gobject_class->get_property = gst_afparse_get_property;
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}
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static void
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gst_afparse_init (GstAFParse *afparse)
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{
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afparse->srcpad = gst_pad_new_from_template (
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gst_element_get_pad_template (GST_ELEMENT (afparse), "src"), "src");
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gst_pad_use_explicit_caps (afparse->srcpad);
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gst_element_add_pad (GST_ELEMENT (afparse), afparse->srcpad);
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afparse->sinkpad = gst_pad_new_from_template (
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gst_element_get_pad_template (GST_ELEMENT (afparse), "sink"), "sink");
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gst_element_add_pad (GST_ELEMENT (afparse), afparse->sinkpad);
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gst_element_set_loop_function (GST_ELEMENT (afparse), gst_afparse_loop);
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afparse->vfile = af_virtual_file_new();
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afparse->vfile->closure = NULL;
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afparse->vfile->read = gst_afparse_vf_read;
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afparse->vfile->length = gst_afparse_vf_length;
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afparse->vfile->write = gst_afparse_vf_write;
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afparse->vfile->destroy = gst_afparse_vf_destroy;
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afparse->vfile->seek = gst_afparse_vf_seek;
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afparse->vfile->tell = gst_afparse_vf_tell;
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afparse->frames_per_read = 1024;
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afparse->curoffset = 0;
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afparse->seq = 0;
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afparse->file = NULL;
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/* default values, should never be needed */
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afparse->channels = 2;
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afparse->width = 16;
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afparse->rate = 44100;
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afparse->type = AF_FILE_WAVE;
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afparse->endianness_data = 1234;
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afparse->endianness_wanted = 1234;
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afparse->timestamp = 0LL;
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}
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static void
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gst_afparse_loop(GstElement *element)
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{
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GstAFParse *afparse;
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GstBuffer *buf;
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gint numframes = 0, frames_to_bytes, frames_per_read, bytes_per_read;
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guint8 *data;
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gboolean bypass_afread = TRUE;
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GstByteStream *bs;
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int s_format, v_format, s_width, v_width;
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afparse = GST_AFPARSE(element);
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afparse->vfile->closure = bs = gst_bytestream_new (afparse->sinkpad);
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/* just stop if we cannot open the file */
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if (!gst_afparse_open_file (afparse)){
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gst_bytestream_destroy ((GstByteStream *) afparse->vfile->closure);
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gst_pad_push (afparse->srcpad, GST_DATA(gst_event_new (GST_EVENT_EOS)));
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gst_element_set_eos (GST_ELEMENT (afparse));
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return;
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}
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/* if audiofile changes the data in any way, we have to access
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* the audio data via afReadFrames. Otherwise we can just access
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* the data directly. */
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afGetSampleFormat(afparse->file, AF_DEFAULT_TRACK, &s_format, &s_width);
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afGetVirtualSampleFormat(afparse->file, AF_DEFAULT_TRACK, &v_format, &v_width);
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if (afGetCompression != AF_COMPRESSION_NONE ||
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afGetByteOrder(afparse->file, AF_DEFAULT_TRACK) != afGetVirtualByteOrder(afparse->file, AF_DEFAULT_TRACK) ||
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s_format != v_format ||
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s_width != v_width) {
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bypass_afread = FALSE;
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}
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if (bypass_afread){
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GST_DEBUG("will bypass afReadFrames\n");
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}
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frames_to_bytes = afparse->channels * afparse->width / 8;
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frames_per_read = afparse->frames_per_read;
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bytes_per_read = frames_per_read * frames_to_bytes;
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afSeekFrame(afparse->file, AF_DEFAULT_TRACK, 0);
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if (bypass_afread){
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GstEvent *event = NULL;
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guint32 waiting;
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guint32 got_bytes;
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do {
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got_bytes = gst_bytestream_read (bs, &buf, bytes_per_read);
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if (got_bytes == 0) {
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/* we need to check for an event. */
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gst_bytestream_get_status (bs, &waiting, &event);
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if (event && GST_EVENT_TYPE(event) == GST_EVENT_EOS) {
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gst_pad_push (afparse->srcpad,
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GST_DATA (gst_event_new (GST_EVENT_EOS)));
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gst_element_set_eos (GST_ELEMENT (afparse));
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break;
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}
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}
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else {
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GST_BUFFER_TIMESTAMP(buf) = afparse->timestamp;
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gst_pad_push (afparse->srcpad, GST_DATA (buf));
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if (got_bytes != bytes_per_read){
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/* this shouldn't happen very often */
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/* FIXME calculate the timestamps based on the fewer bytes received */
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}
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else {
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afparse->timestamp += frames_per_read * 1E9 / afparse->rate;
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}
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}
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}
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while (TRUE);
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}
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else {
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do {
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buf = gst_buffer_new_and_alloc (bytes_per_read);
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GST_BUFFER_TIMESTAMP(buf) = afparse->timestamp;
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data = GST_BUFFER_DATA(buf);
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numframes = afReadFrames (afparse->file, AF_DEFAULT_TRACK, data, frames_per_read);
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/* events are handled in gst_afparse_vf_read so if there are no
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* frames it must be EOS */
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if (numframes < 1){
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gst_buffer_unref(buf);
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gst_pad_push (afparse->srcpad, GST_DATA(gst_event_new (GST_EVENT_EOS)));
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gst_element_set_eos (GST_ELEMENT (afparse));
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break;
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}
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GST_BUFFER_SIZE(buf) = numframes * frames_to_bytes;
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gst_pad_push (afparse->srcpad, GST_DATA (buf));
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afparse->timestamp += numframes * 1E9 / afparse->rate;
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}
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while (TRUE);
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}
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gst_afparse_close_file (afparse);
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gst_bytestream_destroy ((GstByteStream*) afparse->vfile->closure);
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}
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static void
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gst_afparse_set_property (GObject *object, guint prop_id,
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const GValue *value, GParamSpec *pspec)
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{
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GstAFParse *afparse;
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/* it's not null if we got it, but it might not be ours */
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afparse = GST_AFPARSE (object);
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switch (prop_id) {
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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_afparse_get_property (GObject *object, guint prop_id,
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GValue *value, GParamSpec *pspec)
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{
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GstAFParse *afparse;
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g_return_if_fail (GST_IS_AFPARSE (object));
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afparse = GST_AFPARSE (object);
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switch (prop_id) {
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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_afparse_plugin_init (GstPlugin *plugin)
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{
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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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if (!gst_element_register (plugin, "afparse", GST_RANK_NONE, GST_TYPE_AFPARSE))
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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_afparse_open_file (GstAFParse *afparse)
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{
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g_return_val_if_fail (!GST_FLAG_IS_SET (afparse, GST_AFPARSE_OPEN), FALSE);
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/* open the file */
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GST_DEBUG("opening vfile %p\n", afparse->vfile);
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afparse->file = afOpenVirtualFile (afparse->vfile, "r", AF_NULL_FILESETUP);
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if (afparse->file == AF_NULL_FILEHANDLE)
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{
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/* this should never happen */
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g_warning ("ERROR: gstafparse: Could not open virtual file for reading\n");
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return FALSE;
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}
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GST_DEBUG("vfile opened\n");
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/* get the audiofile audio parameters */
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{
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int sampleFormat, sampleWidth;
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afparse->channels = afGetChannels (afparse->file, AF_DEFAULT_TRACK);
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afGetSampleFormat (afparse->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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afparse->is_signed = TRUE;
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break;
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case AF_SAMPFMT_UNSIGNED:
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afparse->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 ("ERROR: float data not supported yet !\n");
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}
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afparse->rate = (guint) afGetRate (afparse->file, AF_DEFAULT_TRACK);
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afparse->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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afparse->channels, afparse->width, afparse->rate,
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afparse->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_set_explicit_caps (afparse->srcpad,
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gst_caps_new_simple (
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"audio/x-raw-int",
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"signed", G_TYPE_BOOLEAN, afparse->is_signed,
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"width", G_TYPE_INT, afparse->width,
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"depth", G_TYPE_INT, afparse->width,
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"rate", G_TYPE_INT, afparse->rate,
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"channels", G_TYPE_INT, afparse->channels,
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NULL));
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GST_FLAG_SET (afparse, GST_AFPARSE_OPEN);
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return TRUE;
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}
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static void
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gst_afparse_close_file (GstAFParse *afparse)
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{
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g_return_if_fail (GST_FLAG_IS_SET (afparse, GST_AFPARSE_OPEN));
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if (afCloseFile (afparse->file) != 0)
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{
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g_warning ("afparse: oops, error closing !\n");
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}
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else {
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GST_FLAG_UNSET (afparse, GST_AFPARSE_OPEN);
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}
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}
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static ssize_t
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gst_afparse_vf_read (AFvirtualfile *vfile, void *data, size_t nbytes)
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{
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GstByteStream *bs = (GstByteStream*)vfile->closure;
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guint8 *bytes = NULL;
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GstEvent *event = NULL;
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guint32 waiting;
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guint32 got_bytes;
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/*gchar *debug_str;*/
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got_bytes = gst_bytestream_peek_bytes(bs, &bytes, nbytes);
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while (got_bytes != nbytes){
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/* handle events */
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gst_bytestream_get_status (bs, &waiting, &event);
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/* FIXME this event handling isn't right yet */
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if (!event){
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/*g_print("no event found with %u bytes\n", got_bytes);*/
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return 0;
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}
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switch (GST_EVENT_TYPE(event)) {
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case GST_EVENT_EOS:
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return 0;
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case GST_EVENT_FLUSH:
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GST_DEBUG("flush");
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break;
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case GST_EVENT_DISCONTINUOUS:
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GST_DEBUG("seek done");
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got_bytes = gst_bytestream_peek_bytes(bs, &bytes, nbytes);
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break;
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default:
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g_warning("unknown event %d", GST_EVENT_TYPE(event));
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got_bytes = gst_bytestream_peek_bytes(bs, &bytes, nbytes);
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}
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}
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memcpy(data, bytes, got_bytes);
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gst_bytestream_flush_fast(bs, got_bytes);
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/* debug_str = g_strndup((gchar*)bytes, got_bytes);
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g_print("read %u bytes: %s\n", got_bytes, debug_str);
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*/
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return got_bytes;
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}
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static long
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gst_afparse_vf_seek (AFvirtualfile *vfile, long offset, int is_relative)
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{
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GstByteStream *bs = (GstByteStream*)vfile->closure;
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GstSeekType method;
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guint64 current_offset = gst_bytestream_tell(bs);
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if (!is_relative){
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if ((guint64)offset == current_offset) {
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/* this seems to happen before every read - bad audiofile */
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return offset;
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}
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method = GST_SEEK_METHOD_SET;
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}
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else {
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if (offset == 0) return current_offset;
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method = GST_SEEK_METHOD_CUR;
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}
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if (gst_bytestream_seek(bs, (gint64)offset, method)){
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GST_DEBUG("doing seek to %d", (gint)offset);
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return offset;
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}
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return 0;
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}
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|
|
|
static long
|
|
gst_afparse_vf_length (AFvirtualfile *vfile)
|
|
{
|
|
GstByteStream *bs = (GstByteStream*)vfile->closure;
|
|
guint64 length;
|
|
|
|
length = gst_bytestream_length(bs);
|
|
GST_DEBUG("doing length: %" G_GUINT64_FORMAT, length);
|
|
return length;
|
|
}
|
|
|
|
static ssize_t
|
|
gst_afparse_vf_write (AFvirtualfile *vfile, const void *data, size_t nbytes)
|
|
{
|
|
/* GstByteStream *bs = (GstByteStream*)vfile->closure;*/
|
|
g_warning("shouldn't write to a readonly pad");
|
|
return 0;
|
|
}
|
|
|
|
static void
|
|
gst_afparse_vf_destroy(AFvirtualfile *vfile)
|
|
{
|
|
/* GstByteStream *bs = (GstByteStream*)vfile->closure;*/
|
|
|
|
GST_DEBUG("doing destroy");
|
|
}
|
|
|
|
static long
|
|
gst_afparse_vf_tell (AFvirtualfile *vfile)
|
|
{
|
|
GstByteStream *bs = (GstByteStream*)vfile->closure;
|
|
guint64 offset;
|
|
|
|
offset = gst_bytestream_tell(bs);
|
|
GST_DEBUG("doing tell: %" G_GUINT64_FORMAT, offset);
|
|
return offset;
|
|
}
|
|
|