mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
7448ede81d
Original commit message from CVS: * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps_with_data): Do actually fix invalid RIFF fmt header values for alaw and mulaw audio instead of just saying so. * gst/wavparse/gstwavparse.c: (gst_wavparse_fmt): Give gst_riff_create_audio_caps_with_data() a chance to fix up broken format header fields before extracting any parameters from the header. (fixes #167633)
1037 lines
29 KiB
C
1037 lines
29 KiB
C
/* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */
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/* GStreamer
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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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "gstwavparse.h"
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#include "gst/riff/riff-ids.h"
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#include "gst/riff/riff-media.h"
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#ifndef G_MAXUINT32
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#define G_MAXUINT32 0xffffffff
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#endif
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GST_DEBUG_CATEGORY (wavparse_debug);
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#define GST_CAT_DEFAULT (wavparse_debug)
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static void gst_wavparse_base_init (gpointer g_class);
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static void gst_wavparse_class_init (GstWavParseClass * klass);
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static void gst_wavparse_init (GstWavParse * wavparse);
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static GstElementStateReturn gst_wavparse_change_state (GstElement * element);
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static const GstFormat *gst_wavparse_get_formats (GstPad * pad);
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static const GstQueryType *gst_wavparse_get_query_types (GstPad * pad);
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static gboolean gst_wavparse_pad_query (GstPad * pad,
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GstQueryType type, GstFormat * format, gint64 * value);
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static gboolean gst_wavparse_pad_convert (GstPad * pad,
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GstFormat src_format,
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gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
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static void gst_wavparse_loop (GstElement * element);
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static const GstEventMask *gst_wavparse_get_event_masks (GstPad * pad);
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static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event);
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static void gst_wavparse_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstStaticPadTemplate sink_template_factory =
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GST_STATIC_PAD_TEMPLATE ("wavparse_sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-wav")
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);
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static GstStaticPadTemplate src_template_factory =
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GST_STATIC_PAD_TEMPLATE ("wavparse_src",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES, /* FIXME: spider */
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"endianness = (int) little_endian, "
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"signed = (boolean) { true, false }, "
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"width = (int) { 8, 16 }, "
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"depth = (int) { 8, 16 }, "
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"rate = (int) [ 8000, 48000 ], "
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"channels = (int) [ 1, 2 ]; "
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"audio/mpeg, "
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"mpegversion = (int) 1, "
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"layer = (int) [ 1, 3 ], "
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"rate = (int) [ 8000, 48000 ], "
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"channels = (int) [ 1, 2 ]; "
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"audio/x-alaw, "
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"rate = (int) [ 8000, 48000 ], "
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"channels = (int) [ 1, 2 ]; "
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"audio/x-mulaw, "
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"rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ];"
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"audio/x-adpcm, "
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"layout = (string) microsoft, "
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"block_align = (int) [ 1, 8192 ], "
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"rate = (int) [ 8000, 48000 ], "
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"channels = (int) [ 1, 2 ]; "
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"audio/x-adpcm, "
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"layout = (string) dvi, "
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"block_align = (int) [ 1, 8192 ], "
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"rate = (int) [ 8000, 48000 ], " "channels = (int) [ 1, 2 ];"
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"audio/x-vnd.sony.atrac3")
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);
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/* WavParse signals and args */
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enum
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{
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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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{
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PROP_0
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};
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static GstElementClass *parent_class = NULL;
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/*static guint gst_wavparse_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_wavparse_get_type (void)
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{
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static GType wavparse_type = 0;
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if (!wavparse_type) {
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static const GTypeInfo wavparse_info = {
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sizeof (GstWavParseClass),
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gst_wavparse_base_init,
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NULL,
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(GClassInitFunc) gst_wavparse_class_init,
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NULL,
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NULL,
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sizeof (GstWavParse),
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0,
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(GInstanceInitFunc) gst_wavparse_init,
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};
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wavparse_type =
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g_type_register_static (GST_TYPE_RIFF_READ, "GstWavParse",
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&wavparse_info, 0);
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}
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return wavparse_type;
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}
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static void
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gst_wavparse_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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static GstElementDetails gst_wavparse_details =
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GST_ELEMENT_DETAILS (".wav demuxer",
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"Codec/Demuxer/Audio",
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"Parse a .wav file into raw audio",
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"Erik Walthinsen <omega@cse.ogi.edu>");
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gst_element_class_set_details (element_class, &gst_wavparse_details);
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/* register src pads */
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template_factory));
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}
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static void
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gst_wavparse_class_init (GstWavParseClass * klass)
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{
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GstElementClass *gstelement_class;
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GObjectClass *object_class;
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gstelement_class = (GstElementClass *) klass;
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object_class = (GObjectClass *) klass;
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parent_class = g_type_class_ref (GST_TYPE_RIFF_READ);
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object_class->get_property = gst_wavparse_get_property;
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gstelement_class->change_state = gst_wavparse_change_state;
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GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
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}
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static void
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gst_wavparse_init (GstWavParse * wavparse)
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{
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/* sink */
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wavparse->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&sink_template_factory), "sink");
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gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->sinkpad);
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GST_RIFF_READ (wavparse)->sinkpad = wavparse->sinkpad;
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gst_pad_set_formats_function (wavparse->sinkpad, gst_wavparse_get_formats);
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gst_pad_set_convert_function (wavparse->sinkpad, gst_wavparse_pad_convert);
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gst_pad_set_query_type_function (wavparse->sinkpad,
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gst_wavparse_get_query_types);
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gst_pad_set_query_function (wavparse->sinkpad, gst_wavparse_pad_query);
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#if 0
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/* source */
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wavparse->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&src_template_factory), "src");
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gst_pad_use_explicit_caps (wavparse->srcpad);
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gst_element_add_pad (GST_ELEMENT (wavparse), wavparse->srcpad);
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gst_pad_set_formats_function (wavparse->srcpad, gst_wavparse_get_formats);
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gst_pad_set_convert_function (wavparse->srcpad, gst_wavparse_pad_convert);
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gst_pad_set_query_type_function (wavparse->srcpad,
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gst_wavparse_get_query_types);
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gst_pad_set_query_function (wavparse->srcpad, gst_wavparse_pad_query);
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gst_pad_set_event_function (wavparse->srcpad, gst_wavparse_srcpad_event);
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gst_pad_set_event_mask_function (wavparse->srcpad,
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gst_wavparse_get_event_masks);
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#endif
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gst_element_set_loop_function (GST_ELEMENT (wavparse), gst_wavparse_loop);
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wavparse->state = GST_WAVPARSE_START;
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/* These will all be set correctly in the fmt chunk */
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wavparse->depth = 0;
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wavparse->rate = 0;
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wavparse->width = 0;
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wavparse->channels = 0;
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wavparse->seek_pending = FALSE;
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wavparse->seek_offset = 0;
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}
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static void
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gst_wavparse_destroy_sourcepad (GstWavParse * wavparse)
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{
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if (wavparse->srcpad) {
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gst_element_remove_pad (GST_ELEMENT (wavparse), wavparse->srcpad);
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wavparse->srcpad = NULL;
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}
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}
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static void
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gst_wavparse_create_sourcepad (GstWavParse * wavparse)
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{
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gst_wavparse_destroy_sourcepad (wavparse);
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/* source */
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wavparse->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&src_template_factory), "src");
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gst_pad_use_explicit_caps (wavparse->srcpad);
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gst_pad_set_formats_function (wavparse->srcpad, gst_wavparse_get_formats);
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gst_pad_set_convert_function (wavparse->srcpad, gst_wavparse_pad_convert);
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gst_pad_set_query_type_function (wavparse->srcpad,
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gst_wavparse_get_query_types);
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gst_pad_set_query_function (wavparse->srcpad, gst_wavparse_pad_query);
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gst_pad_set_event_function (wavparse->srcpad, gst_wavparse_srcpad_event);
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gst_pad_set_event_mask_function (wavparse->srcpad,
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gst_wavparse_get_event_masks);
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}
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static void
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gst_wavparse_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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GstWavParse *wavparse;
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wavparse = GST_WAVPARSE (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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#if 0
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static void
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gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
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{
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guint32 got_bytes;
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GstByteStream *bs = wavparse->bs;
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gst_riff_chunk *temp_chunk, chunk;
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guint8 *tempdata;
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struct _gst_riff_labl labl, *temp_labl;
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struct _gst_riff_ltxt ltxt, *temp_ltxt;
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struct _gst_riff_note note, *temp_note;
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char *label_name;
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GstProps *props;
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GstPropsEntry *entry;
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GstCaps *new_caps;
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GList *caps = NULL;
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props = wavparse->metadata->properties;
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while (len > 0) {
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got_bytes =
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gst_bytestream_peek_bytes (bs, &tempdata, sizeof (gst_riff_chunk));
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if (got_bytes != sizeof (gst_riff_chunk)) {
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return;
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}
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temp_chunk = (gst_riff_chunk *) tempdata;
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chunk.id = GUINT32_FROM_LE (temp_chunk->id);
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chunk.size = GUINT32_FROM_LE (temp_chunk->size);
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if (chunk.size == 0) {
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gst_bytestream_flush (bs, sizeof (gst_riff_chunk));
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len -= sizeof (gst_riff_chunk);
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continue;
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}
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switch (chunk.id) {
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case GST_RIFF_adtl_labl:
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got_bytes =
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gst_bytestream_peek_bytes (bs, &tempdata,
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sizeof (struct _gst_riff_labl));
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if (got_bytes != sizeof (struct _gst_riff_labl)) {
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return;
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}
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temp_labl = (struct _gst_riff_labl *) tempdata;
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labl.id = GUINT32_FROM_LE (temp_labl->id);
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labl.size = GUINT32_FROM_LE (temp_labl->size);
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labl.identifier = GUINT32_FROM_LE (temp_labl->identifier);
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gst_bytestream_flush (bs, sizeof (struct _gst_riff_labl));
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len -= sizeof (struct _gst_riff_labl);
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got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, labl.size - 4);
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if (got_bytes != labl.size - 4) {
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return;
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}
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label_name = (char *) tempdata;
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gst_bytestream_flush (bs, ((labl.size - 4) + 1) & ~1);
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len -= (((labl.size - 4) + 1) & ~1);
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new_caps = gst_caps_new ("label",
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"application/x-gst-metadata",
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gst_props_new ("identifier", G_TYPE_INT (labl.identifier),
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"name", G_TYPE_STRING (label_name), NULL));
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if (gst_props_get (props, "labels", &caps, NULL)) {
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caps = g_list_append (caps, new_caps);
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} else {
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caps = g_list_append (NULL, new_caps);
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entry = gst_props_entry_new ("labels", GST_PROPS_GLIST (caps));
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gst_props_add_entry (props, entry);
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}
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break;
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case GST_RIFF_adtl_ltxt:
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got_bytes =
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gst_bytestream_peek_bytes (bs, &tempdata,
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sizeof (struct _gst_riff_ltxt));
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if (got_bytes != sizeof (struct _gst_riff_ltxt)) {
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return;
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}
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temp_ltxt = (struct _gst_riff_ltxt *) tempdata;
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ltxt.id = GUINT32_FROM_LE (temp_ltxt->id);
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ltxt.size = GUINT32_FROM_LE (temp_ltxt->size);
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ltxt.identifier = GUINT32_FROM_LE (temp_ltxt->identifier);
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ltxt.length = GUINT32_FROM_LE (temp_ltxt->length);
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ltxt.purpose = GUINT32_FROM_LE (temp_ltxt->purpose);
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ltxt.country = GUINT16_FROM_LE (temp_ltxt->country);
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ltxt.language = GUINT16_FROM_LE (temp_ltxt->language);
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ltxt.dialect = GUINT16_FROM_LE (temp_ltxt->dialect);
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ltxt.codepage = GUINT16_FROM_LE (temp_ltxt->codepage);
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gst_bytestream_flush (bs, sizeof (struct _gst_riff_ltxt));
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len -= sizeof (struct _gst_riff_ltxt);
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if (ltxt.size - 20 > 0) {
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got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, ltxt.size - 20);
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if (got_bytes != ltxt.size - 20) {
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return;
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}
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gst_bytestream_flush (bs, ((ltxt.size - 20) + 1) & ~1);
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len -= (((ltxt.size - 20) + 1) & ~1);
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label_name = (char *) tempdata;
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} else {
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label_name = "";
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}
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new_caps = gst_caps_new ("ltxt",
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"application/x-gst-metadata",
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gst_props_new ("identifier", G_TYPE_INT (ltxt.identifier),
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"name", G_TYPE_STRING (label_name),
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"length", G_TYPE_INT (ltxt.length), NULL));
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if (gst_props_get (props, "ltxts", &caps, NULL)) {
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caps = g_list_append (caps, new_caps);
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} else {
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caps = g_list_append (NULL, new_caps);
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entry = gst_props_entry_new ("ltxts", GST_PROPS_GLIST (caps));
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gst_props_add_entry (props, entry);
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}
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break;
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case GST_RIFF_adtl_note:
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got_bytes =
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gst_bytestream_peek_bytes (bs, &tempdata,
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sizeof (struct _gst_riff_note));
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if (got_bytes != sizeof (struct _gst_riff_note)) {
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return;
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}
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temp_note = (struct _gst_riff_note *) tempdata;
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note.id = GUINT32_FROM_LE (temp_note->id);
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note.size = GUINT32_FROM_LE (temp_note->size);
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note.identifier = GUINT32_FROM_LE (temp_note->identifier);
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gst_bytestream_flush (bs, sizeof (struct _gst_riff_note));
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len -= sizeof (struct _gst_riff_note);
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got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, note.size - 4);
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if (got_bytes != note.size - 4) {
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return;
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}
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gst_bytestream_flush (bs, ((note.size - 4) + 1) & ~1);
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len -= (((note.size - 4) + 1) & ~1);
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label_name = (char *) tempdata;
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new_caps = gst_caps_new ("note",
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"application/x-gst-metadata",
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gst_props_new ("identifier", G_TYPE_INT (note.identifier),
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"name", G_TYPE_STRING (label_name), NULL));
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if (gst_props_get (props, "notes", &caps, NULL)) {
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caps = g_list_append (caps, new_caps);
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} else {
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caps = g_list_append (NULL, new_caps);
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entry = gst_props_entry_new ("notes", GST_PROPS_GLIST (caps));
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gst_props_add_entry (props, entry);
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}
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break;
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default:
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g_print ("Unknown chunk: " GST_FOURCC_FORMAT "\n",
|
|
GST_FOURCC_ARGS (chunk.id));
|
|
return;
|
|
}
|
|
}
|
|
|
|
g_object_notify (G_OBJECT (wavparse), "metadata");
|
|
}
|
|
#endif
|
|
|
|
#if 0
|
|
static void
|
|
gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
|
|
{
|
|
guint32 got_bytes;
|
|
GstByteStream *bs = wavparse->bs;
|
|
struct _gst_riff_cue *temp_cue, cue;
|
|
struct _gst_riff_cuepoints *points;
|
|
guint8 *tempdata;
|
|
int i;
|
|
GList *cues = NULL;
|
|
GstPropsEntry *entry;
|
|
|
|
while (len > 0) {
|
|
int required;
|
|
|
|
got_bytes =
|
|
gst_bytestream_peek_bytes (bs, &tempdata,
|
|
sizeof (struct _gst_riff_cue));
|
|
temp_cue = (struct _gst_riff_cue *) tempdata;
|
|
|
|
/* fixup for our big endian friends */
|
|
cue.id = GUINT32_FROM_LE (temp_cue->id);
|
|
cue.size = GUINT32_FROM_LE (temp_cue->size);
|
|
cue.cuepoints = GUINT32_FROM_LE (temp_cue->cuepoints);
|
|
|
|
gst_bytestream_flush (bs, sizeof (struct _gst_riff_cue));
|
|
if (got_bytes != sizeof (struct _gst_riff_cue)) {
|
|
return;
|
|
}
|
|
|
|
len -= sizeof (struct _gst_riff_cue);
|
|
|
|
/* -4 because cue.size contains the cuepoints size
|
|
and we've already flushed that out of the system */
|
|
required = cue.size - 4;
|
|
got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, required);
|
|
gst_bytestream_flush (bs, ((required) + 1) & ~1);
|
|
if (got_bytes != required) {
|
|
return;
|
|
}
|
|
|
|
len -= (((cue.size - 4) + 1) & ~1);
|
|
|
|
/* now we have an array of struct _gst_riff_cuepoints in tempdata */
|
|
points = (struct _gst_riff_cuepoints *) tempdata;
|
|
|
|
for (i = 0; i < cue.cuepoints; i++) {
|
|
GstCaps *caps;
|
|
|
|
caps = gst_caps_new ("cues",
|
|
"application/x-gst-metadata",
|
|
gst_props_new ("identifier", G_TYPE_INT (points[i].identifier),
|
|
"position", G_TYPE_INT (points[i].offset), NULL));
|
|
cues = g_list_append (cues, caps);
|
|
}
|
|
|
|
entry = gst_props_entry_new ("cues", GST_PROPS_GLIST (cues));
|
|
gst_props_add_entry (wavparse->metadata->properties, entry);
|
|
}
|
|
|
|
g_object_notify (G_OBJECT (wavparse), "metadata");
|
|
}
|
|
#endif
|
|
|
|
static gboolean
|
|
gst_wavparse_stream_init (GstWavParse * wav)
|
|
{
|
|
GstRiffRead *riff = GST_RIFF_READ (wav);
|
|
guint32 doctype;
|
|
|
|
if (!gst_riff_read_header (riff, &doctype)) {
|
|
GST_WARNING_OBJECT (wav, "could not read header");
|
|
return FALSE;
|
|
}
|
|
|
|
if (doctype != GST_RIFF_RIFF_WAVE) {
|
|
GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL), (NULL));
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* Read 'fmt ' header */
|
|
static gboolean
|
|
gst_wavparse_fmt (GstWavParse * wav)
|
|
{
|
|
GstRiffRead *riff = GST_RIFF_READ (wav);
|
|
gst_riff_strf_auds *header = NULL;
|
|
GstCaps *caps;
|
|
|
|
if (!gst_riff_read_strf_auds (riff, &header)) {
|
|
g_warning ("Not fmt");
|
|
return FALSE;
|
|
}
|
|
|
|
/* Note: gst_riff_create_audio_caps might nedd to fix values in
|
|
* the header header depending on the format, so call it first */
|
|
caps = gst_riff_create_audio_caps (header->format, NULL, header, NULL);
|
|
|
|
wav->format = header->format;
|
|
wav->rate = header->rate;
|
|
wav->channels = header->channels;
|
|
wav->blockalign = header->blockalign;
|
|
wav->width = (header->blockalign * 8) / header->channels;
|
|
wav->depth = header->size;
|
|
wav->bps = header->av_bps;
|
|
|
|
g_free (header);
|
|
|
|
if (caps) {
|
|
gst_wavparse_create_sourcepad (wav);
|
|
gst_pad_set_explicit_caps (wav->srcpad, caps);
|
|
gst_caps_free (caps);
|
|
gst_element_add_pad (GST_ELEMENT (wav), wav->srcpad);
|
|
gst_element_no_more_pads (GST_ELEMENT (wav));
|
|
GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels);
|
|
} else {
|
|
GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wavparse_other (GstWavParse * wav)
|
|
{
|
|
GstRiffRead *riff = GST_RIFF_READ (wav);
|
|
guint32 tag, length;
|
|
|
|
if (!gst_riff_peek_head (riff, &tag, &length, NULL)) {
|
|
GST_WARNING_OBJECT (wav, "could not peek head");
|
|
return FALSE;
|
|
}
|
|
GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %d", tag,
|
|
(gchar *) & tag, length);
|
|
|
|
switch (tag) {
|
|
case GST_RIFF_TAG_LIST:
|
|
if (!(tag = gst_riff_peek_list (riff))) {
|
|
GST_WARNING_OBJECT (wav, "could not peek list");
|
|
return FALSE;
|
|
}
|
|
|
|
switch (tag) {
|
|
case GST_RIFF_LIST_INFO:
|
|
if (!gst_riff_read_list (riff, &tag) || !gst_riff_read_info (riff)) {
|
|
GST_WARNING_OBJECT (wav, "could not read list");
|
|
return FALSE;
|
|
}
|
|
break;
|
|
|
|
case GST_RIFF_LIST_adtl:
|
|
if (!gst_riff_read_skip (riff)) {
|
|
GST_WARNING_OBJECT (wav, "could not read skip");
|
|
return FALSE;
|
|
}
|
|
break;
|
|
|
|
default:
|
|
GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag,
|
|
(gchar *) & tag);
|
|
if (!gst_riff_read_skip (riff)) {
|
|
GST_WARNING_OBJECT (wav, "could not read skip");
|
|
return FALSE;
|
|
}
|
|
break;
|
|
}
|
|
|
|
break;
|
|
|
|
case GST_RIFF_TAG_data:
|
|
if (!gst_bytestream_flush (riff->bs, 8)) {
|
|
GST_WARNING_OBJECT (wav, "could not flush 8 bytes");
|
|
return FALSE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (wav, "switching to data mode");
|
|
wav->state = GST_WAVPARSE_DATA;
|
|
wav->datastart = gst_bytestream_tell (riff->bs);
|
|
if (length == 0) {
|
|
guint64 file_length;
|
|
|
|
/* length is 0, data probably stretches to the end
|
|
* of file */
|
|
GST_DEBUG_OBJECT (wav, "length is 0 trying to find length");
|
|
/* get length of file */
|
|
file_length = gst_bytestream_length (riff->bs);
|
|
if (file_length == -1) {
|
|
GST_DEBUG_OBJECT (wav,
|
|
"could not get file length, assuming data to eof");
|
|
/* could not get length, assuming till eof */
|
|
length = G_MAXUINT32;
|
|
}
|
|
if (file_length > G_MAXUINT32) {
|
|
GST_DEBUG_OBJECT (wav, "file length %lld, clipping to 32 bits");
|
|
/* could not get length, assuming till eof */
|
|
length = G_MAXUINT32;
|
|
} else {
|
|
GST_DEBUG_OBJECT (wav, "file length %lld, datalength", file_length,
|
|
length);
|
|
/* substract offset of datastart from length */
|
|
length = file_length - wav->datastart;
|
|
GST_DEBUG_OBJECT (wav, "datalength %lld", length);
|
|
}
|
|
}
|
|
wav->dataleft = wav->datasize = (guint64) length;
|
|
break;
|
|
|
|
case GST_RIFF_TAG_cue:
|
|
if (!gst_riff_read_skip (riff)) {
|
|
GST_WARNING_OBJECT (wav, "could not read skip");
|
|
return FALSE;
|
|
}
|
|
break;
|
|
|
|
default:
|
|
GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag, (gchar *) & tag);
|
|
if (!gst_riff_read_skip (riff))
|
|
return FALSE;
|
|
break;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wavparse_handle_seek (GstWavParse * wav)
|
|
{
|
|
GstRiffRead *riff = GST_RIFF_READ (wav);
|
|
GstEvent *event = NULL;
|
|
guint32 remaining;
|
|
guint8 *data;
|
|
|
|
if (!gst_bytestream_seek (riff->bs, wav->seek_offset + wav->datastart,
|
|
GST_SEEK_METHOD_SET))
|
|
return FALSE;
|
|
|
|
/* wait for discont */
|
|
while (!event) {
|
|
if (gst_bytestream_peek_bytes (riff->bs, &data, 1)) {
|
|
GST_WARNING ("Unexpected data after seek - this means seek failed");
|
|
return FALSE;
|
|
}
|
|
|
|
/* get the discont event and return */
|
|
gst_bytestream_get_status (riff->bs, &remaining, &event);
|
|
if (!event) {
|
|
GST_WARNING ("No discontinuity event after seek - seek failed");
|
|
return FALSE;
|
|
} else if (GST_EVENT_TYPE (event) != GST_EVENT_DISCONTINUOUS) {
|
|
GstEventType type = GST_EVENT_TYPE (event);
|
|
|
|
gst_pad_event_default (riff->sinkpad, event);
|
|
if (type == GST_EVENT_EOS)
|
|
return FALSE;
|
|
|
|
event = NULL;
|
|
}
|
|
}
|
|
|
|
wav->dataleft = wav->datasize - wav->seek_offset;
|
|
|
|
gst_event_unref (event);
|
|
event = gst_event_new_discontinuous (FALSE,
|
|
GST_FORMAT_BYTES, wav->seek_offset,
|
|
GST_FORMAT_TIME, GST_SECOND * wav->seek_offset / wav->bps,
|
|
GST_FORMAT_UNDEFINED);
|
|
gst_pad_event_default (wav->sinkpad, event);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
#define MAX_BUFFER_SIZE 4096
|
|
|
|
static void
|
|
gst_wavparse_loop (GstElement * element)
|
|
{
|
|
GstWavParse *wav = GST_WAVPARSE (element);
|
|
GstRiffRead *riff = GST_RIFF_READ (wav);
|
|
|
|
if (wav->state == GST_WAVPARSE_DATA) {
|
|
/* seek handling */
|
|
if (wav->seek_pending) {
|
|
gst_wavparse_handle_seek (wav);
|
|
wav->seek_pending = FALSE;
|
|
}
|
|
|
|
if (wav->dataleft > 0) {
|
|
guint32 got_bytes, desired;
|
|
GstBuffer *buf = NULL;
|
|
|
|
desired = MIN (wav->dataleft, MAX_BUFFER_SIZE);
|
|
if (!(buf = gst_riff_read_element_data (riff, desired, &got_bytes))) {
|
|
GST_WARNING_OBJECT (wav, "trying to read %d bytes failed", desired);
|
|
return;
|
|
}
|
|
GST_DEBUG_OBJECT (wav, "read %d bytes, got %d bytes", desired, got_bytes);
|
|
|
|
GST_BUFFER_TIMESTAMP (buf) = GST_SECOND *
|
|
(wav->datasize - wav->dataleft) / wav->bps;
|
|
GST_BUFFER_DURATION (buf) = GST_SECOND * got_bytes / wav->bps;
|
|
|
|
gst_pad_push (wav->srcpad, GST_DATA (buf));
|
|
|
|
wav->byteoffset += got_bytes;
|
|
if (got_bytes < wav->dataleft) {
|
|
wav->dataleft -= got_bytes;
|
|
return;
|
|
} else {
|
|
wav->dataleft = 0;
|
|
wav->state = GST_WAVPARSE_OTHER;
|
|
}
|
|
} else {
|
|
wav->state = GST_WAVPARSE_OTHER;
|
|
}
|
|
}
|
|
|
|
switch (wav->state) {
|
|
case GST_WAVPARSE_START:
|
|
if (!gst_wavparse_stream_init (wav)) {
|
|
return;
|
|
}
|
|
|
|
wav->state = GST_WAVPARSE_FMT;
|
|
/* fall-through */
|
|
|
|
case GST_WAVPARSE_FMT:
|
|
if (!gst_wavparse_fmt (wav)) {
|
|
return;
|
|
}
|
|
|
|
wav->state = GST_WAVPARSE_OTHER;
|
|
/* fall-through */
|
|
|
|
case GST_WAVPARSE_OTHER:
|
|
if (!gst_wavparse_other (wav)) {
|
|
return;
|
|
}
|
|
|
|
break;
|
|
|
|
case GST_WAVPARSE_DATA:
|
|
|
|
default:
|
|
g_assert_not_reached ();
|
|
}
|
|
}
|
|
|
|
/* convert and query stuff */
|
|
static const GstFormat *
|
|
gst_wavparse_get_formats (GstPad * pad)
|
|
{
|
|
static GstFormat formats[] = {
|
|
GST_FORMAT_TIME,
|
|
GST_FORMAT_BYTES,
|
|
GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
|
|
0
|
|
};
|
|
|
|
return formats;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wavparse_pad_convert (GstPad * pad,
|
|
GstFormat src_format, gint64 src_value,
|
|
GstFormat * dest_format, gint64 * dest_value)
|
|
{
|
|
guint bytes_per_sample, byterate;
|
|
GstWavParse *wavparse;
|
|
|
|
wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
|
|
|
|
bytes_per_sample = wavparse->channels * wavparse->width / 8;
|
|
if (bytes_per_sample == 0) {
|
|
GST_DEBUG ("bytes_per_sample 0, probably an mp3 - channels %d, width %d",
|
|
wavparse->channels, wavparse->width);
|
|
return FALSE;
|
|
}
|
|
byterate = wavparse->bps;
|
|
if (byterate == 0) {
|
|
g_warning ("byterate is 0, internal error\n");
|
|
return FALSE;
|
|
}
|
|
GST_DEBUG ("bytes per sample: %d", bytes_per_sample);
|
|
|
|
switch (src_format) {
|
|
case GST_FORMAT_BYTES:
|
|
switch (*dest_format) {
|
|
case GST_FORMAT_DEFAULT:
|
|
*dest_value = src_value / bytes_per_sample;
|
|
break;
|
|
case GST_FORMAT_TIME:
|
|
*dest_value = src_value * GST_SECOND / byterate;
|
|
break;
|
|
default:
|
|
return FALSE;
|
|
}
|
|
*dest_value -= *dest_value % bytes_per_sample;
|
|
break;
|
|
|
|
case GST_FORMAT_DEFAULT:
|
|
switch (*dest_format) {
|
|
case GST_FORMAT_BYTES:
|
|
*dest_value = src_value * bytes_per_sample;
|
|
break;
|
|
case GST_FORMAT_TIME:
|
|
*dest_value = src_value * GST_SECOND / wavparse->rate;
|
|
break;
|
|
default:
|
|
return FALSE;
|
|
}
|
|
break;
|
|
|
|
case GST_FORMAT_TIME:
|
|
switch (*dest_format) {
|
|
case GST_FORMAT_BYTES:
|
|
/* make sure we end up on a sample boundary */
|
|
*dest_value =
|
|
(src_value * wavparse->rate / GST_SECOND) * wavparse->blockalign;
|
|
break;
|
|
case GST_FORMAT_DEFAULT:
|
|
*dest_value = src_value * wavparse->rate / GST_SECOND;
|
|
break;
|
|
default:
|
|
return FALSE;
|
|
}
|
|
break;
|
|
|
|
default:
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static const GstQueryType *
|
|
gst_wavparse_get_query_types (GstPad * pad)
|
|
{
|
|
static const GstQueryType types[] = {
|
|
GST_QUERY_TOTAL,
|
|
GST_QUERY_POSITION,
|
|
0
|
|
};
|
|
|
|
return types;
|
|
}
|
|
|
|
/* handle queries for location and length in requested format */
|
|
static gboolean
|
|
gst_wavparse_pad_query (GstPad * pad, GstQueryType type,
|
|
GstFormat * format, gint64 * value)
|
|
{
|
|
gint64 bytevalue;
|
|
GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (pad));
|
|
|
|
/* only if we know */
|
|
if (wav->state != GST_WAVPARSE_DATA)
|
|
return FALSE;
|
|
|
|
switch (type) {
|
|
case GST_QUERY_POSITION:
|
|
bytevalue = wav->datasize - wav->dataleft;
|
|
break;
|
|
case GST_QUERY_TOTAL:
|
|
bytevalue = wav->datasize;
|
|
break;
|
|
default:
|
|
return FALSE;
|
|
}
|
|
|
|
if (*format == GST_FORMAT_BYTES) {
|
|
*value = bytevalue;
|
|
return TRUE;
|
|
}
|
|
|
|
return gst_pad_convert (wav->sinkpad, GST_FORMAT_BYTES,
|
|
bytevalue, format, value);
|
|
}
|
|
|
|
static const GstEventMask *
|
|
gst_wavparse_get_event_masks (GstPad * pad)
|
|
{
|
|
static const GstEventMask gst_wavparse_src_event_masks[] = {
|
|
{GST_EVENT_SEEK, GST_SEEK_METHOD_SET | GST_SEEK_FLAG_FLUSH},
|
|
{0,}
|
|
};
|
|
|
|
return gst_wavparse_src_event_masks;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstWavParse *wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
|
|
gboolean res = FALSE;
|
|
|
|
GST_DEBUG ("event %d", GST_EVENT_TYPE (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
{
|
|
gint64 byteoffset;
|
|
GstFormat format;
|
|
|
|
/* bring format to samples for the peer element, */
|
|
format = GST_FORMAT_BYTES;
|
|
res = gst_pad_convert (pad,
|
|
GST_EVENT_SEEK_FORMAT (event),
|
|
GST_EVENT_SEEK_OFFSET (event), &format, &byteoffset);
|
|
|
|
if (res) {
|
|
/* ok, seek worked, update our state */
|
|
wavparse->seek_offset = byteoffset;
|
|
wavparse->seek_pending = TRUE;
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
gst_event_unref (event);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstElementStateReturn
|
|
gst_wavparse_change_state (GstElement * element)
|
|
{
|
|
GstWavParse *wav = GST_WAVPARSE (element);
|
|
|
|
switch (GST_STATE_TRANSITION (element)) {
|
|
case GST_STATE_NULL_TO_READY:
|
|
break;
|
|
|
|
case GST_STATE_READY_TO_PAUSED:
|
|
wav->state = GST_WAVPARSE_START;
|
|
break;
|
|
|
|
case GST_STATE_PAUSED_TO_PLAYING:
|
|
break;
|
|
|
|
case GST_STATE_PLAYING_TO_PAUSED:
|
|
break;
|
|
|
|
case GST_STATE_PAUSED_TO_READY:
|
|
gst_wavparse_destroy_sourcepad (wav);
|
|
wav->state = GST_WAVPARSE_START;
|
|
|
|
wav->width = 0;
|
|
wav->depth = 0;
|
|
wav->rate = 0;
|
|
wav->channels = 0;
|
|
|
|
wav->seek_pending = FALSE;
|
|
wav->seek_offset = 0;
|
|
break;
|
|
|
|
case GST_STATE_READY_TO_NULL:
|
|
break;
|
|
}
|
|
|
|
if (GST_ELEMENT_CLASS (parent_class)->change_state)
|
|
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
|
|
|
|
return GST_STATE_SUCCESS;
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
if (!gst_library_load ("riff")) {
|
|
return FALSE;
|
|
}
|
|
|
|
return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
|
|
GST_TYPE_WAVPARSE);
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"wavparse",
|
|
"Parse a .wav file into raw audio",
|
|
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE, GST_ORIGIN)
|