mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-08 18:39:54 +00:00
95011fd7e8
Original commit message from CVS: New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs
511 lines
16 KiB
C
511 lines
16 KiB
C
/* 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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/*#define GST_DEBUG_ENABLED */
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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 <gstmpegaudioparse.h>
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/* elementfactory information */
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static GstElementDetails mp3parse_details = {
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"MPEG1 Audio Parser",
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"Codec/Parser",
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"LGPL",
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"Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
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VERSION,
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"Erik Walthinsen <omega@cse.ogi.edu>",
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"(C) 1999",
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};
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static GstPadTemplate*
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mp3_src_factory (void)
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{
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return
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gst_pad_template_new (
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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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"mp3parse_src",
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"audio/x-mp3",
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gst_props_new (
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"layer", GST_PROPS_INT_RANGE (1, 3),
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"rate", GST_PROPS_INT_RANGE (8000, 48000),
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"channels", GST_PROPS_INT_RANGE (1, 2),
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NULL)),
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NULL);
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}
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static GstPadTemplate*
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mp3_sink_factory (void)
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{
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return
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gst_pad_template_new (
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"sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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gst_caps_new (
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"mp3parse_sink",
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"audio/x-mp3",
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NULL),
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NULL);
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};
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/* GstMPEGAudioParse signals and args */
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enum {
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/* FILL ME */
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LAST_SIGNAL
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};
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enum {
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ARG_0,
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ARG_SKIP,
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ARG_BIT_RATE,
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/* FILL ME */
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};
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static GstPadTemplate *sink_temp, *src_temp;
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static void gst_mp3parse_class_init (GstMPEGAudioParseClass *klass);
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static void gst_mp3parse_init (GstMPEGAudioParse *mp3parse);
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static void gst_mp3parse_chain (GstPad *pad,GstBuffer *buf);
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static long bpf_from_header (GstMPEGAudioParse *parse, unsigned long header);
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static int head_check (unsigned long head);
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static void gst_mp3parse_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec);
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static void gst_mp3parse_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
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static GstElementStateReturn
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gst_mp3parse_change_state (GstElement *element);
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static GstElementClass *parent_class = NULL;
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/*static guint gst_mp3parse_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_mp3parse_get_type(void) {
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static GType mp3parse_type = 0;
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if (!mp3parse_type) {
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static const GTypeInfo mp3parse_info = {
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sizeof(GstMPEGAudioParseClass), NULL,
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NULL,
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(GClassInitFunc)gst_mp3parse_class_init,
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NULL,
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NULL,
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sizeof(GstMPEGAudioParse),
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0,
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(GInstanceInitFunc)gst_mp3parse_init,
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};
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mp3parse_type = g_type_register_static(GST_TYPE_ELEMENT, "GstMPEGAudioParse", &mp3parse_info, 0);
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}
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return mp3parse_type;
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}
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static void
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gst_mp3parse_class_init (GstMPEGAudioParseClass *klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass*)klass;
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gstelement_class = (GstElementClass*)klass;
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_SKIP,
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g_param_spec_int("skip","skip","skip",
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G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); /* CHECKME */
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_BIT_RATE,
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g_param_spec_int("bitrate","Bitrate","Bit Rate",
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G_MININT,G_MAXINT,0,G_PARAM_READABLE)); /* CHECKME */
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parent_class = g_type_class_ref(GST_TYPE_ELEMENT);
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gobject_class->set_property = gst_mp3parse_set_property;
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gobject_class->get_property = gst_mp3parse_get_property;
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gstelement_class->change_state = gst_mp3parse_change_state;
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}
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static void
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gst_mp3parse_init (GstMPEGAudioParse *mp3parse)
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{
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mp3parse->sinkpad = gst_pad_new_from_template(sink_temp, "sink");
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gst_element_add_pad(GST_ELEMENT(mp3parse),mp3parse->sinkpad);
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gst_pad_set_chain_function(mp3parse->sinkpad,gst_mp3parse_chain);
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gst_element_set_loop_function (GST_ELEMENT(mp3parse),NULL);
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mp3parse->srcpad = gst_pad_new_from_template(src_temp, "src");
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gst_element_add_pad(GST_ELEMENT(mp3parse),mp3parse->srcpad);
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/*gst_pad_set_type_id(mp3parse->srcpad, mp3frametype); */
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mp3parse->partialbuf = NULL;
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mp3parse->skip = 0;
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mp3parse->in_flush = FALSE;
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mp3parse->rate = mp3parse->channels = mp3parse->layer = -1;
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}
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static guint32
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gst_mp3parse_next_header (guchar *buf,guint32 len,guint32 start)
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{
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guint32 offset = start;
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int f = 0;
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while (offset < (len - 4)) {
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if (buf[offset] == 0xff)
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f = 1;
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else if (f && ((buf[offset] >> 4) == 0x0f))
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return offset - 1;
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else
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f = 0;
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offset++;
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}
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return -1;
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}
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static void
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gst_mp3parse_chain (GstPad *pad, GstBuffer *buf)
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{
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GstMPEGAudioParse *mp3parse;
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guchar *data;
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glong size,offset = 0;
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guint32 header;
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int bpf;
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GstBuffer *outbuf;
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guint64 last_ts;
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g_return_if_fail(pad != NULL);
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g_return_if_fail(GST_IS_PAD(pad));
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g_return_if_fail(buf != NULL);
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/* g_return_if_fail(GST_IS_BUFFER(buf)); */
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mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
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GST_DEBUG ("mp3parse: received buffer of %d bytes",GST_BUFFER_SIZE(buf));
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last_ts = GST_BUFFER_TIMESTAMP(buf);
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/* FIXME, do flush */
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/*
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if (mp3parse->partialbuf) {
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gst_buffer_unref(mp3parse->partialbuf);
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mp3parse->partialbuf = NULL;
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}
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mp3parse->in_flush = TRUE;
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*/
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/* if we have something left from the previous frame */
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if (mp3parse->partialbuf) {
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mp3parse->partialbuf = gst_buffer_merge(mp3parse->partialbuf, buf);
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/* and the one we received.. */
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gst_buffer_unref(buf);
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}
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else {
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mp3parse->partialbuf = buf;
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}
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size = GST_BUFFER_SIZE(mp3parse->partialbuf);
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data = GST_BUFFER_DATA(mp3parse->partialbuf);
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/* while we still have bytes left -4 for the header */
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while (offset < size-4) {
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int skipped = 0;
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GST_DEBUG ("mp3parse: offset %ld, size %ld ",offset, size);
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/* search for a possible start byte */
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for (;((data[offset] != 0xff) && (offset < size));offset++) skipped++;
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if (skipped && !mp3parse->in_flush) {
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GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes",offset,skipped);
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}
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/* construct the header word */
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header = GUINT32_FROM_BE(*((guint32 *)(data+offset)));
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/* if it's a valid header, go ahead and send off the frame */
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if (head_check(header)) {
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/* calculate the bpf of the frame */
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bpf = bpf_from_header(mp3parse, header);
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/********************************************************************************
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* robust seek support
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* - This performs additional frame validation if the in_flush flag is set
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* (indicating a discontinuous stream).
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* - The current frame header is not accepted as valid unless the NEXT frame
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* header has the same values for most fields. This significantly increases
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* the probability that we aren't processing random data.
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* - It is not clear if this is sufficient for robust seeking of Layer III
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* streams which utilize the concept of a "bit reservoir" by borrow bitrate
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* from previous frames. In this case, seeking may be more complicated because
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* the frames are not independently coded.
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********************************************************************************/
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if ( mp3parse->in_flush ) {
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guint32 header2;
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if ((size-offset)<(bpf+4)) { if (mp3parse->in_flush) break; } /* wait until we have the the entire current frame as well as the next frame header */
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header2 = GUINT32_FROM_BE(*((guint32 *)(data+offset+bpf)));
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GST_DEBUG ("mp3parse: header=%08X, header2=%08X, bpf=%d", (unsigned int)header, (unsigned int)header2, bpf );
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#define HDRMASK ~( (0xF<<12)/*bitrate*/ | (1<<9)/*padding*/ | (3<<4)/*mode extension*/ ) /* mask the bits which are allowed to differ between frames */
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if ( (header2&HDRMASK) != (header&HDRMASK) ) { /* require 2 matching headers in a row */
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GST_DEBUG ("mp3parse: next header doesn't match (header=%08X, header2=%08X, bpf=%d)", (unsigned int)header, (unsigned int)header2, bpf );
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offset++; /* This frame is invalid. Start looking for a valid frame at the next position in the stream */
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continue;
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}
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}
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/* if we don't have the whole frame... */
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if ((size - offset) < bpf) {
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GST_DEBUG ("mp3parse: partial buffer needed %ld < %d ",(size-offset), bpf);
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break;
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} else {
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outbuf = gst_buffer_create_sub(mp3parse->partialbuf,offset,bpf);
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offset += bpf;
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if (mp3parse->skip == 0) {
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GST_DEBUG ("mp3parse: pushing buffer of %d bytes",GST_BUFFER_SIZE(outbuf));
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if (mp3parse->in_flush) {
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/* FIXME do some sort of flush event */
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mp3parse->in_flush = FALSE;
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}
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GST_BUFFER_TIMESTAMP(outbuf) = last_ts;
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if (GST_PAD_CAPS (mp3parse->srcpad) != NULL) {
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gst_pad_push(mp3parse->srcpad,outbuf);
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} else {
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GST_DEBUG ("No capsnego yet, delaying buffer push");
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gst_buffer_unref (outbuf);
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}
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}
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else {
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GST_DEBUG ("mp3parse: skipping buffer of %d bytes",GST_BUFFER_SIZE(outbuf));
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gst_buffer_unref(outbuf);
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mp3parse->skip--;
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}
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}
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} else {
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offset++;
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if (!mp3parse->in_flush) GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)");
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}
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}
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/* if we have processed this block and there are still */
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/* bytes left not in a partial block, copy them over. */
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if (size-offset > 0) {
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glong remainder = (size - offset);
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GST_DEBUG ("mp3parse: partial buffer needed %ld for trailing bytes",remainder);
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outbuf = gst_buffer_create_sub(mp3parse->partialbuf,offset,remainder);
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gst_buffer_unref(mp3parse->partialbuf);
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mp3parse->partialbuf = outbuf;
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}
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else {
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gst_buffer_unref(mp3parse->partialbuf);
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mp3parse->partialbuf = NULL;
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}
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}
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static int mp3parse_tabsel[2][3][16] =
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{ { {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448, },
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{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, },
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{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, } },
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{ {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256, },
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{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, },
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{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, } },
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};
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static long mp3parse_freqs[9] =
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{44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000};
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static long
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bpf_from_header (GstMPEGAudioParse *parse, unsigned long header)
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{
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int layer_index,layer,lsf,samplerate_index,padding,mode;
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long bpf;
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gint channels, rate;
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/*mpegver = (header >> 19) & 0x3; // don't need this for bpf */
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layer_index = (header >> 17) & 0x3;
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layer = 4 - layer_index;
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lsf = (header & (1 << 20)) ? ((header & (1 << 19)) ? 0 : 1) : 1;
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parse->bit_rate = mp3parse_tabsel[lsf][layer - 1][((header >> 12) & 0xf)];
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samplerate_index = (header >> 10) & 0x3;
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padding = (header >> 9) & 0x1;
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mode = (header >> 6) & 0x3;
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if (layer == 1) {
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bpf = parse->bit_rate * 12000;
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bpf /= mp3parse_freqs[samplerate_index];
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bpf = ((bpf + padding) << 2);
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} else {
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bpf = parse->bit_rate * 144000;
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bpf /= mp3parse_freqs[samplerate_index];
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bpf += padding;
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}
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channels = (mode == 3) ? 1 : 2;
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rate = mp3parse_freqs[samplerate_index];
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if (channels != parse->channels ||
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rate != parse->rate ||
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layer != parse->layer) {
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GstCaps *caps = GST_CAPS_NEW ("mp3parse_src",
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"audio/mpeg",
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"layer", GST_PROPS_INT (layer),
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"channels", GST_PROPS_INT (channels),
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"rate", GST_PROPS_INT (rate));
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if (gst_pad_try_set_caps(parse->srcpad, caps) <= 0) {
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gst_element_error (GST_ELEMENT (parse),
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"mp3parse: failed to negotiate format with next element");
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}
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parse->channels = channels;
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parse->layer = layer;
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parse->rate = rate;
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}
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/*g_print("%08x: layer %d lsf %d bitrate %d samplerate_index %d padding %d - bpf %d\n", */
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/*header,layer,lsf,bitrate,samplerate_index,padding,bpf); */
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return bpf;
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}
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static gboolean
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head_check (unsigned long head)
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{
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GST_DEBUG ("checking mp3 header 0x%08lx",head);
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/* if it's not a valid sync */
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if ((head & 0xffe00000) != 0xffe00000) {
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GST_DEBUG ("invalid sync");return FALSE; }
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/* if it's an invalid MPEG version */
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if (((head >> 19) & 3) == 0x1) {
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GST_DEBUG ("invalid MPEG version");return FALSE; }
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/* if it's an invalid layer */
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if (!((head >> 17) & 3)) {
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GST_DEBUG ("invalid layer");return FALSE; }
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/* if it's an invalid bitrate */
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if (((head >> 12) & 0xf) == 0x0) {
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GST_DEBUG ("invalid bitrate");return FALSE; }
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if (((head >> 12) & 0xf) == 0xf) {
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GST_DEBUG ("invalid bitrate");return FALSE; }
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/* if it's an invalid samplerate */
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if (((head >> 10) & 0x3) == 0x3) {
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GST_DEBUG ("invalid samplerate");return FALSE; }
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if ((head & 0xffff0000) == 0xfffe0000) {
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GST_DEBUG ("invalid sync");return FALSE; }
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if (head & 0x00000002) {
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GST_DEBUG ("invalid emphasis");return FALSE; }
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return TRUE;
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}
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static void
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gst_mp3parse_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
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{
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GstMPEGAudioParse *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_MP3PARSE(object));
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src = GST_MP3PARSE(object);
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switch (prop_id) {
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case ARG_SKIP:
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src->skip = g_value_get_int (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_mp3parse_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
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{
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GstMPEGAudioParse *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_MP3PARSE(object));
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src = GST_MP3PARSE(object);
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switch (prop_id) {
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case ARG_SKIP:
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g_value_set_int (value, src->skip);
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break;
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case ARG_BIT_RATE:
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g_value_set_int (value, src->bit_rate * 1000);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstElementStateReturn
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gst_mp3parse_change_state (GstElement *element)
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{
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GstMPEGAudioParse *src;
|
|
|
|
g_return_val_if_fail(GST_IS_MP3PARSE(element), GST_STATE_FAILURE);
|
|
src = GST_MP3PARSE(element);
|
|
|
|
switch (GST_STATE_TRANSITION (element)) {
|
|
case GST_STATE_PAUSED_TO_READY:
|
|
src->channels = -1; src->rate = -1; src->layer = -1;
|
|
break;
|
|
default:
|
|
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 (GModule *module, GstPlugin *plugin)
|
|
{
|
|
GstElementFactory *factory;
|
|
|
|
/* create an elementfactory for the mp3parse element */
|
|
factory = gst_element_factory_new ("mp3parse",
|
|
GST_TYPE_MP3PARSE,
|
|
&mp3parse_details);
|
|
g_return_val_if_fail (factory != NULL, FALSE);
|
|
|
|
sink_temp = mp3_sink_factory ();
|
|
gst_element_factory_add_pad_template (factory, sink_temp);
|
|
|
|
src_temp = mp3_src_factory ();
|
|
gst_element_factory_add_pad_template (factory, src_temp);
|
|
|
|
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
GstPluginDesc plugin_desc = {
|
|
GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"mp3parse",
|
|
plugin_init
|
|
};
|