2002-03-20 21:45:03 +00:00
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/* GStreamer
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2001-12-22 23:26:48 +00:00
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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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2002-03-19 04:10:05 +00:00
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/*#define GST_DEBUG_ENABLED */
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2003-06-29 19:46:09 +00:00
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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2001-12-22 23:26:48 +00:00
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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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2002-04-20 21:42:51 +00:00
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"MPEG1 Audio Parser",
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2003-11-16 22:02:04 +00:00
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"Codec/Parser/Audio",
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2002-04-20 21:42:51 +00:00
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"Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
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2003-11-02 21:10:18 +00:00
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"Erik Walthinsen <omega@cse.ogi.edu>"
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2001-12-22 23:26:48 +00:00
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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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2002-04-11 20:42:25 +00:00
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gst_pad_template_new (
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2001-12-22 23:26:48 +00:00
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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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2003-07-29 19:57:11 +00:00
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"audio/mpeg",
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2001-12-22 23:26:48 +00:00
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gst_props_new (
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2003-07-29 19:57:11 +00:00
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"mpegversion", GST_PROPS_INT (1),
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2001-12-22 23:26:48 +00:00
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"layer", GST_PROPS_INT_RANGE (1, 3),
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2003-07-06 20:49:50 +00:00
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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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2001-12-22 23:26:48 +00:00
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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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2002-04-11 20:42:25 +00:00
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gst_pad_template_new (
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2001-12-22 23:26:48 +00:00
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"sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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configure.ac: Improve mpeg2enc detection. This is for distributions that do ship mjpegtools, but without mpeg2enc. Al...
Original commit message from CVS:
2003-12-21 Ronald Bultje <rbultje@ronald.bitfreak.net>
* configure.ac:
Improve mpeg2enc detection. This is for distributions that do
ship mjpegtools, but without mpeg2enc. Also does object check
for might there ever be ABI incompatibility.
* ext/mpeg2enc/gstmpeg2enc.cc:
Add Andrew as second maintainer (he's helping me), and also add
an error if no caps was set. This happens if I pull before capsnego
and that's something I should solve sometime else.
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup):
Fix time parsing.
* gst/matroska/matroska-mux.c: (gst_matroska_mux_audio_pad_link),
(gst_matroska_mux_track_header):
Add caps to templates.
* gst/mpegaudioparse/gstmpegaudioparse.c: (mp3_sink_factory):
Add mpegversion=1 to prevent confusion with MPEG/AAC.
* gst/mpegstream/gstmpegdemux.c:
Remove layer since it causes warnings about unfixed caps.
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
Fix obvious typo (we error out if caps were set, we should of
course error out if *no* caps were set).
* sys/oss/gstosselement.c: (gst_osselement_convert):
Fix format conversion, we confused bits/bytes.
* sys/oss/gstosselement.h:
Improve documentation for 'bps'.
* sys/v4l/TODO:
Remove stuff about plugins that need removing - this was done
ages ago.
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_init),
(gst_v4lmjpegsrc_src_convert), (gst_v4lmjpegsrc_src_query):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_src_convert),
(gst_v4lsrc_src_query):
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init),
(gst_v4l2src_src_convert), (gst_v4l2src_src_query):
Add get_query_types(), get_formats() and query() functions.
2003-12-21 18:44:33 +00:00
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GST_CAPS_NEW (
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2001-12-22 23:26:48 +00:00
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"mp3parse_sink",
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2003-07-29 19:57:11 +00:00
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"audio/mpeg",
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configure.ac: Improve mpeg2enc detection. This is for distributions that do ship mjpegtools, but without mpeg2enc. Al...
Original commit message from CVS:
2003-12-21 Ronald Bultje <rbultje@ronald.bitfreak.net>
* configure.ac:
Improve mpeg2enc detection. This is for distributions that do
ship mjpegtools, but without mpeg2enc. Also does object check
for might there ever be ABI incompatibility.
* ext/mpeg2enc/gstmpeg2enc.cc:
Add Andrew as second maintainer (he's helping me), and also add
an error if no caps was set. This happens if I pull before capsnego
and that's something I should solve sometime else.
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup):
Fix time parsing.
* gst/matroska/matroska-mux.c: (gst_matroska_mux_audio_pad_link),
(gst_matroska_mux_track_header):
Add caps to templates.
* gst/mpegaudioparse/gstmpegaudioparse.c: (mp3_sink_factory):
Add mpegversion=1 to prevent confusion with MPEG/AAC.
* gst/mpegstream/gstmpegdemux.c:
Remove layer since it causes warnings about unfixed caps.
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
Fix obvious typo (we error out if caps were set, we should of
course error out if *no* caps were set).
* sys/oss/gstosselement.c: (gst_osselement_convert):
Fix format conversion, we confused bits/bytes.
* sys/oss/gstosselement.h:
Improve documentation for 'bps'.
* sys/v4l/TODO:
Remove stuff about plugins that need removing - this was done
ages ago.
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_init),
(gst_v4lmjpegsrc_src_convert), (gst_v4lmjpegsrc_src_query):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_src_convert),
(gst_v4lsrc_src_query):
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init),
(gst_v4l2src_src_convert), (gst_v4l2src_src_query):
Add get_query_types(), get_formats() and query() functions.
2003-12-21 18:44:33 +00:00
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"mpegversion", GST_PROPS_INT (1)
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)
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);
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2001-12-22 23:26:48 +00:00
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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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2003-11-02 21:10:18 +00:00
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static void gst_mp3parse_base_init (GstMPEGAudioParseClass *klass);
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2001-12-22 23:26:48 +00:00
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static void gst_mp3parse_init (GstMPEGAudioParse *mp3parse);
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2003-10-08 16:08:10 +00:00
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static void gst_mp3parse_chain (GstPad *pad,GstData *_data);
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2001-12-22 23:26:48 +00:00
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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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2003-07-06 20:49:50 +00:00
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static GstElementStateReturn
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gst_mp3parse_change_state (GstElement *element);
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2001-12-22 23:26:48 +00:00
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static GstElementClass *parent_class = NULL;
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2002-03-19 04:10:05 +00:00
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/*static guint gst_mp3parse_signals[LAST_SIGNAL] = { 0 }; */
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2001-12-22 23:26:48 +00:00
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GType
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2001-12-23 06:56:55 +00:00
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gst_mp3parse_get_type(void) {
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2001-12-22 23:26:48 +00:00
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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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2003-11-02 21:10:18 +00:00
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sizeof(GstMPEGAudioParseClass),
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(GBaseInitFunc)gst_mp3parse_base_init,
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2001-12-22 23:26:48 +00:00
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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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2003-10-01 13:14:51 +00:00
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mp3parse_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstMPEGAudioParse",
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&mp3parse_info, 0);
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2001-12-22 23:26:48 +00:00
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}
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return mp3parse_type;
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}
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2003-10-01 13:14:51 +00:00
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static guint mp3types_bitrates[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 guint mp3types_freqs[3][3] =
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{ {44100, 48000, 32000},
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{22050, 24000, 16000},
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{11025, 12000, 8000}};
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static inline guint
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mp3_type_frame_length_from_header (guint32 header, guint *put_layer,
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guint *put_channels, guint *put_bitrate,
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guint *put_samplerate)
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{
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guint length;
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gulong mode, samplerate, bitrate, layer, version, channels;
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/* we don't need extension, copyright, original or
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* emphasis for the frame length */
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header >>= 6;
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/* mode */
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mode = header & 0x3;
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header >>= 3;
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/* padding */
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length = header & 0x1;
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header >>= 1;
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/* sampling frequency */
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samplerate = header & 0x3;
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if (samplerate == 3)
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return 0;
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header >>= 2;
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/* bitrate index */
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bitrate = header & 0xF;
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if (bitrate == 15 || bitrate == 0)
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return 0;
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/* ignore error correction, too */
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header >>= 5;
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/* layer */
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layer = 4 - (header & 0x3);
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if (layer == 4)
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return 0;
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header >>= 2;
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/* version */
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version = header & 0x3;
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if (version == 1)
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return 0;
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/* lookup */
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channels = (mode == 3) ? 1 : 2;
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bitrate = mp3types_bitrates[version == 3 ? 0 : 1][layer - 1][bitrate];
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samplerate = mp3types_freqs[version > 0 ? version - 1 : 0][samplerate];
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/* calculating */
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if (layer == 1) {
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length = ((12000 * bitrate / samplerate) + length) * 4;
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} else {
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length += ((layer == 3 && version == 0) ? 144000 : 72000) * bitrate / samplerate;
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}
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GST_DEBUG ("Calculated mp3 frame length of %u bytes", length);
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GST_DEBUG ("samplerate = %lu - bitrate = %lu - layer = %lu - version = %lu"
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" - channels = %lu",
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samplerate, bitrate, layer, version, channels);
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if (put_layer)
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*put_layer = layer;
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if (put_channels)
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*put_channels = channels;
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if (put_bitrate)
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*put_bitrate = bitrate;
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if (put_samplerate)
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*put_samplerate = samplerate;
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return length;
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}
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/**
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* The chance that random data is identified as a valid mp3 header is 63 / 2^18
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* (0.024%) per try. This makes the function for calculating false positives
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* 1 - (1 - ((63 / 2 ^18) ^ GST_MP3_TYPEFIND_MIN_HEADERS)) ^ buffersize)
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* This has the following probabilities of false positives:
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* bufsize MIN_HEADERS
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* (bytes) 1 2 3 4
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* 4096 62.6% 0.02% 0% 0%
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* 16384 98% 0.09% 0% 0%
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* 1 MiB 100% 5.88% 0% 0%
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* 1 GiB 100% 100% 1.44% 0%
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* 1 TiB 100% 100% 100% 0.35%
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* This means that the current choice (3 headers by most of the time 4096 byte
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* buffers is pretty safe for now.
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*
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* The max. size of each frame is 1440 bytes, which means that for N frames
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* to be detected, we need 1440 * GST_MP3_TYPEFIND_MIN_HEADERS + 3 of data.
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* Assuming we step into the stream right after the frame header, this
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* means we need 1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3 bytes
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* of data (5762) to always detect any mp3.
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*/
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/* increase this value when this function finds too many false positives */
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#define GST_MP3_TYPEFIND_MIN_HEADERS 3
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#define GST_MP3_TYPEFIND_MIN_DATA (1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3)
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static GstCaps *
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mp3_caps_create (guint layer, guint channels,
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guint bitrate, guint samplerate)
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{
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GstCaps *new;
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g_assert (layer);
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g_assert (samplerate);
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g_assert (bitrate);
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g_assert (channels);
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new = GST_CAPS_NEW ("mp3_type_find",
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"audio/mpeg",
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"mpegversion", GST_PROPS_INT (1),
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"layer", GST_PROPS_INT (layer),
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/*"bitrate", GST_PROPS_INT (bitrate),*/
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"rate", GST_PROPS_INT (samplerate),
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"channels", GST_PROPS_INT (channels));
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return new;
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}
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2003-11-02 21:10:18 +00:00
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static void
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gst_mp3parse_base_init (GstMPEGAudioParseClass *klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class, sink_temp);
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gst_element_class_add_pad_template (element_class, src_temp);
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|
|
gst_element_class_set_details (element_class, &mp3parse_details);
|
|
|
|
}
|
|
|
|
|
2001-12-22 23:26:48 +00:00
|
|
|
static void
|
|
|
|
gst_mp3parse_class_init (GstMPEGAudioParseClass *klass)
|
|
|
|
{
|
|
|
|
GObjectClass *gobject_class;
|
|
|
|
GstElementClass *gstelement_class;
|
|
|
|
|
|
|
|
gobject_class = (GObjectClass*)klass;
|
|
|
|
gstelement_class = (GstElementClass*)klass;
|
|
|
|
|
|
|
|
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_SKIP,
|
|
|
|
g_param_spec_int("skip","skip","skip",
|
2002-03-19 04:10:05 +00:00
|
|
|
G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); /* CHECKME */
|
2001-12-22 23:26:48 +00:00
|
|
|
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_BIT_RATE,
|
2003-07-06 20:49:50 +00:00
|
|
|
g_param_spec_int("bitrate","Bitrate","Bit Rate",
|
2002-03-19 04:10:05 +00:00
|
|
|
G_MININT,G_MAXINT,0,G_PARAM_READABLE)); /* CHECKME */
|
2001-12-22 23:26:48 +00:00
|
|
|
|
|
|
|
parent_class = g_type_class_ref(GST_TYPE_ELEMENT);
|
|
|
|
|
|
|
|
gobject_class->set_property = gst_mp3parse_set_property;
|
|
|
|
gobject_class->get_property = gst_mp3parse_get_property;
|
2003-07-06 20:49:50 +00:00
|
|
|
|
|
|
|
gstelement_class->change_state = gst_mp3parse_change_state;
|
2001-12-22 23:26:48 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
gst_mp3parse_init (GstMPEGAudioParse *mp3parse)
|
|
|
|
{
|
|
|
|
mp3parse->sinkpad = gst_pad_new_from_template(sink_temp, "sink");
|
|
|
|
gst_element_add_pad(GST_ELEMENT(mp3parse),mp3parse->sinkpad);
|
|
|
|
|
|
|
|
gst_pad_set_chain_function(mp3parse->sinkpad,gst_mp3parse_chain);
|
2002-04-06 03:40:15 +00:00
|
|
|
gst_element_set_loop_function (GST_ELEMENT(mp3parse),NULL);
|
2001-12-22 23:26:48 +00:00
|
|
|
|
|
|
|
mp3parse->srcpad = gst_pad_new_from_template(src_temp, "src");
|
|
|
|
gst_element_add_pad(GST_ELEMENT(mp3parse),mp3parse->srcpad);
|
2002-03-19 04:10:05 +00:00
|
|
|
/*gst_pad_set_type_id(mp3parse->srcpad, mp3frametype); */
|
2001-12-22 23:26:48 +00:00
|
|
|
|
|
|
|
mp3parse->partialbuf = NULL;
|
|
|
|
mp3parse->skip = 0;
|
|
|
|
mp3parse->in_flush = FALSE;
|
2003-07-06 20:49:50 +00:00
|
|
|
|
|
|
|
mp3parse->rate = mp3parse->channels = mp3parse->layer = -1;
|
2001-12-22 23:26:48 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
2003-10-08 16:08:10 +00:00
|
|
|
gst_mp3parse_chain (GstPad *pad, GstData *_data)
|
2001-12-22 23:26:48 +00:00
|
|
|
{
|
2003-10-08 16:08:10 +00:00
|
|
|
GstBuffer *buf = GST_BUFFER (_data);
|
2001-12-22 23:26:48 +00:00
|
|
|
GstMPEGAudioParse *mp3parse;
|
|
|
|
guchar *data;
|
|
|
|
glong size,offset = 0;
|
2003-04-07 18:43:25 +00:00
|
|
|
guint32 header;
|
2001-12-22 23:26:48 +00:00
|
|
|
int bpf;
|
|
|
|
GstBuffer *outbuf;
|
|
|
|
guint64 last_ts;
|
|
|
|
|
|
|
|
g_return_if_fail(pad != NULL);
|
|
|
|
g_return_if_fail(GST_IS_PAD(pad));
|
|
|
|
g_return_if_fail(buf != NULL);
|
2002-03-19 04:10:05 +00:00
|
|
|
/* g_return_if_fail(GST_IS_BUFFER(buf)); */
|
2001-12-22 23:26:48 +00:00
|
|
|
|
|
|
|
mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
|
|
|
|
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("mp3parse: received buffer of %d bytes",GST_BUFFER_SIZE(buf));
|
2001-12-22 23:26:48 +00:00
|
|
|
|
|
|
|
last_ts = GST_BUFFER_TIMESTAMP(buf);
|
|
|
|
|
2001-12-28 20:24:41 +00:00
|
|
|
/* FIXME, do flush */
|
|
|
|
/*
|
2001-12-22 23:26:48 +00:00
|
|
|
if (mp3parse->partialbuf) {
|
|
|
|
gst_buffer_unref(mp3parse->partialbuf);
|
|
|
|
mp3parse->partialbuf = NULL;
|
|
|
|
}
|
|
|
|
mp3parse->in_flush = TRUE;
|
2001-12-28 20:24:41 +00:00
|
|
|
*/
|
2001-12-22 23:26:48 +00:00
|
|
|
|
2002-03-19 04:10:05 +00:00
|
|
|
/* if we have something left from the previous frame */
|
2001-12-22 23:26:48 +00:00
|
|
|
if (mp3parse->partialbuf) {
|
|
|
|
|
2002-07-08 19:43:33 +00:00
|
|
|
mp3parse->partialbuf = gst_buffer_merge(mp3parse->partialbuf, buf);
|
2002-03-19 04:10:05 +00:00
|
|
|
/* and the one we received.. */
|
2001-12-22 23:26:48 +00:00
|
|
|
gst_buffer_unref(buf);
|
|
|
|
}
|
|
|
|
else {
|
|
|
|
mp3parse->partialbuf = buf;
|
|
|
|
}
|
|
|
|
|
|
|
|
size = GST_BUFFER_SIZE(mp3parse->partialbuf);
|
|
|
|
data = GST_BUFFER_DATA(mp3parse->partialbuf);
|
|
|
|
|
2002-03-19 04:10:05 +00:00
|
|
|
/* while we still have bytes left -4 for the header */
|
2001-12-22 23:26:48 +00:00
|
|
|
while (offset < size-4) {
|
|
|
|
int skipped = 0;
|
|
|
|
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("mp3parse: offset %ld, size %ld ",offset, size);
|
2001-12-22 23:26:48 +00:00
|
|
|
|
2002-03-19 04:10:05 +00:00
|
|
|
/* search for a possible start byte */
|
2001-12-22 23:26:48 +00:00
|
|
|
for (;((data[offset] != 0xff) && (offset < size));offset++) skipped++;
|
|
|
|
if (skipped && !mp3parse->in_flush) {
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes",offset,skipped);
|
2001-12-22 23:26:48 +00:00
|
|
|
}
|
2002-03-19 04:10:05 +00:00
|
|
|
/* construct the header word */
|
2003-04-07 18:43:25 +00:00
|
|
|
header = GUINT32_FROM_BE(*((guint32 *)(data+offset)));
|
2002-03-19 04:10:05 +00:00
|
|
|
/* if it's a valid header, go ahead and send off the frame */
|
2001-12-22 23:26:48 +00:00
|
|
|
if (head_check(header)) {
|
2002-03-19 04:10:05 +00:00
|
|
|
/* calculate the bpf of the frame */
|
2001-12-22 23:26:48 +00:00
|
|
|
bpf = bpf_from_header(mp3parse, header);
|
|
|
|
|
|
|
|
/********************************************************************************
|
|
|
|
* robust seek support
|
|
|
|
* - This performs additional frame validation if the in_flush flag is set
|
|
|
|
* (indicating a discontinuous stream).
|
|
|
|
* - The current frame header is not accepted as valid unless the NEXT frame
|
|
|
|
* header has the same values for most fields. This significantly increases
|
|
|
|
* the probability that we aren't processing random data.
|
|
|
|
* - It is not clear if this is sufficient for robust seeking of Layer III
|
|
|
|
* streams which utilize the concept of a "bit reservoir" by borrow bitrate
|
|
|
|
* from previous frames. In this case, seeking may be more complicated because
|
|
|
|
* the frames are not independently coded.
|
|
|
|
********************************************************************************/
|
|
|
|
if ( mp3parse->in_flush ) {
|
2003-04-07 18:43:25 +00:00
|
|
|
guint32 header2;
|
2001-12-22 23:26:48 +00:00
|
|
|
|
2002-03-19 04:10:05 +00:00
|
|
|
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 */
|
2001-12-22 23:26:48 +00:00
|
|
|
|
2003-04-07 18:43:25 +00:00
|
|
|
header2 = GUINT32_FROM_BE(*((guint32 *)(data+offset+bpf)));
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("mp3parse: header=%08X, header2=%08X, bpf=%d", (unsigned int)header, (unsigned int)header2, bpf );
|
2001-12-22 23:26:48 +00:00
|
|
|
|
2003-10-01 13:14:51 +00:00
|
|
|
/* mask the bits which are allowed to differ between frames */
|
|
|
|
#define HDRMASK ~((0xF << 12) /* bitrate */ | \
|
|
|
|
(0x1 << 9) /* padding */ | \
|
|
|
|
(0x3 << 4)) /*mode extension*/
|
2001-12-22 23:26:48 +00:00
|
|
|
|
2002-03-19 04:10:05 +00:00
|
|
|
if ( (header2&HDRMASK) != (header&HDRMASK) ) { /* require 2 matching headers in a row */
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("mp3parse: next header doesn't match (header=%08X, header2=%08X, bpf=%d)", (unsigned int)header, (unsigned int)header2, bpf );
|
2002-03-19 04:10:05 +00:00
|
|
|
offset++; /* This frame is invalid. Start looking for a valid frame at the next position in the stream */
|
2001-12-22 23:26:48 +00:00
|
|
|
continue;
|
|
|
|
}
|
|
|
|
|
|
|
|
}
|
|
|
|
|
2002-03-19 04:10:05 +00:00
|
|
|
/* if we don't have the whole frame... */
|
2001-12-22 23:26:48 +00:00
|
|
|
if ((size - offset) < bpf) {
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("mp3parse: partial buffer needed %ld < %d ",(size-offset), bpf);
|
2001-12-22 23:26:48 +00:00
|
|
|
break;
|
|
|
|
} else {
|
|
|
|
|
|
|
|
outbuf = gst_buffer_create_sub(mp3parse->partialbuf,offset,bpf);
|
|
|
|
|
|
|
|
offset += bpf;
|
|
|
|
if (mp3parse->skip == 0) {
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("mp3parse: pushing buffer of %d bytes",GST_BUFFER_SIZE(outbuf));
|
2001-12-22 23:26:48 +00:00
|
|
|
if (mp3parse->in_flush) {
|
2001-12-28 20:24:41 +00:00
|
|
|
/* FIXME do some sort of flush event */
|
2001-12-22 23:26:48 +00:00
|
|
|
mp3parse->in_flush = FALSE;
|
|
|
|
}
|
|
|
|
GST_BUFFER_TIMESTAMP(outbuf) = last_ts;
|
2003-07-29 19:57:11 +00:00
|
|
|
GST_BUFFER_DURATION(outbuf) = 8 * (GST_SECOND/1000) * GST_BUFFER_SIZE(outbuf) / mp3parse->bit_rate;
|
2003-07-06 20:49:50 +00:00
|
|
|
|
|
|
|
if (GST_PAD_CAPS (mp3parse->srcpad) != NULL) {
|
2003-10-08 16:08:10 +00:00
|
|
|
gst_pad_push(mp3parse->srcpad,GST_DATA (outbuf));
|
2003-07-06 20:49:50 +00:00
|
|
|
} else {
|
|
|
|
GST_DEBUG ("No capsnego yet, delaying buffer push");
|
|
|
|
gst_buffer_unref (outbuf);
|
|
|
|
}
|
2001-12-22 23:26:48 +00:00
|
|
|
}
|
|
|
|
else {
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("mp3parse: skipping buffer of %d bytes",GST_BUFFER_SIZE(outbuf));
|
2001-12-22 23:26:48 +00:00
|
|
|
gst_buffer_unref(outbuf);
|
|
|
|
mp3parse->skip--;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
} else {
|
|
|
|
offset++;
|
2003-06-29 19:46:09 +00:00
|
|
|
if (!mp3parse->in_flush) GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)");
|
2001-12-22 23:26:48 +00:00
|
|
|
}
|
|
|
|
}
|
2002-03-19 04:10:05 +00:00
|
|
|
/* if we have processed this block and there are still */
|
|
|
|
/* bytes left not in a partial block, copy them over. */
|
2001-12-22 23:26:48 +00:00
|
|
|
if (size-offset > 0) {
|
|
|
|
glong remainder = (size - offset);
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("mp3parse: partial buffer needed %ld for trailing bytes",remainder);
|
2001-12-22 23:26:48 +00:00
|
|
|
|
|
|
|
outbuf = gst_buffer_create_sub(mp3parse->partialbuf,offset,remainder);
|
|
|
|
gst_buffer_unref(mp3parse->partialbuf);
|
|
|
|
mp3parse->partialbuf = outbuf;
|
|
|
|
}
|
|
|
|
else {
|
|
|
|
gst_buffer_unref(mp3parse->partialbuf);
|
|
|
|
mp3parse->partialbuf = NULL;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
static long
|
|
|
|
bpf_from_header (GstMPEGAudioParse *parse, unsigned long header)
|
|
|
|
{
|
2003-10-01 13:14:51 +00:00
|
|
|
guint bitrate, layer, rate, channels, length;
|
2001-12-22 23:26:48 +00:00
|
|
|
|
2003-10-01 13:14:51 +00:00
|
|
|
if (!(length = mp3_type_frame_length_from_header (header, &layer,
|
|
|
|
&channels,
|
|
|
|
&bitrate, &rate))) {
|
|
|
|
return 0;
|
2001-12-22 23:26:48 +00:00
|
|
|
}
|
|
|
|
|
2003-07-06 20:49:50 +00:00
|
|
|
if (channels != parse->channels ||
|
|
|
|
rate != parse->rate ||
|
2003-10-01 13:14:51 +00:00
|
|
|
layer != parse->layer ||
|
|
|
|
bitrate != parse->bit_rate) {
|
|
|
|
GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate);
|
|
|
|
|
2003-07-06 20:49:50 +00:00
|
|
|
if (gst_pad_try_set_caps(parse->srcpad, caps) <= 0) {
|
2003-09-16 10:00:00 +00:00
|
|
|
gst_element_error (GST_ELEMENT (parse),
|
|
|
|
"mp3parse: failed to negotiate format with next element");
|
2003-07-06 20:49:50 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
parse->channels = channels;
|
|
|
|
parse->layer = layer;
|
|
|
|
parse->rate = rate;
|
2003-10-01 13:14:51 +00:00
|
|
|
parse->bit_rate = bitrate;
|
2003-07-06 20:49:50 +00:00
|
|
|
}
|
|
|
|
|
2003-10-01 13:14:51 +00:00
|
|
|
return length;
|
2001-12-22 23:26:48 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
static gboolean
|
|
|
|
head_check (unsigned long head)
|
|
|
|
{
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("checking mp3 header 0x%08lx",head);
|
2001-12-22 23:26:48 +00:00
|
|
|
/* if it's not a valid sync */
|
|
|
|
if ((head & 0xffe00000) != 0xffe00000) {
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("invalid sync");return FALSE; }
|
2001-12-22 23:26:48 +00:00
|
|
|
/* if it's an invalid MPEG version */
|
|
|
|
if (((head >> 19) & 3) == 0x1) {
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("invalid MPEG version");return FALSE; }
|
2001-12-22 23:26:48 +00:00
|
|
|
/* if it's an invalid layer */
|
|
|
|
if (!((head >> 17) & 3)) {
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("invalid layer");return FALSE; }
|
2001-12-22 23:26:48 +00:00
|
|
|
/* if it's an invalid bitrate */
|
|
|
|
if (((head >> 12) & 0xf) == 0x0) {
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("invalid bitrate");return FALSE; }
|
2001-12-22 23:26:48 +00:00
|
|
|
if (((head >> 12) & 0xf) == 0xf) {
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("invalid bitrate");return FALSE; }
|
2001-12-22 23:26:48 +00:00
|
|
|
/* if it's an invalid samplerate */
|
|
|
|
if (((head >> 10) & 0x3) == 0x3) {
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("invalid samplerate");return FALSE; }
|
2001-12-22 23:26:48 +00:00
|
|
|
if ((head & 0xffff0000) == 0xfffe0000) {
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("invalid sync");return FALSE; }
|
2001-12-22 23:26:48 +00:00
|
|
|
if (head & 0x00000002) {
|
2003-06-29 19:46:09 +00:00
|
|
|
GST_DEBUG ("invalid emphasis");return FALSE; }
|
2001-12-22 23:26:48 +00:00
|
|
|
|
|
|
|
return TRUE;
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
gst_mp3parse_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
|
|
|
|
{
|
|
|
|
GstMPEGAudioParse *src;
|
|
|
|
|
|
|
|
/* it's not null if we got it, but it might not be ours */
|
|
|
|
g_return_if_fail(GST_IS_MP3PARSE(object));
|
|
|
|
src = GST_MP3PARSE(object);
|
|
|
|
|
|
|
|
switch (prop_id) {
|
|
|
|
case ARG_SKIP:
|
|
|
|
src->skip = g_value_get_int (value);
|
|
|
|
break;
|
|
|
|
default:
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
static void
|
|
|
|
gst_mp3parse_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
|
|
|
|
{
|
|
|
|
GstMPEGAudioParse *src;
|
|
|
|
|
|
|
|
/* it's not null if we got it, but it might not be ours */
|
|
|
|
g_return_if_fail(GST_IS_MP3PARSE(object));
|
|
|
|
src = GST_MP3PARSE(object);
|
|
|
|
|
|
|
|
switch (prop_id) {
|
|
|
|
case ARG_SKIP:
|
|
|
|
g_value_set_int (value, src->skip);
|
|
|
|
break;
|
|
|
|
case ARG_BIT_RATE:
|
|
|
|
g_value_set_int (value, src->bit_rate * 1000);
|
|
|
|
break;
|
|
|
|
default:
|
|
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
|
|
break;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2003-07-06 20:49:50 +00:00
|
|
|
static GstElementStateReturn
|
|
|
|
gst_mp3parse_change_state (GstElement *element)
|
|
|
|
{
|
|
|
|
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;
|
|
|
|
}
|
|
|
|
|
2001-12-22 23:26:48 +00:00
|
|
|
static gboolean
|
2003-11-02 21:10:18 +00:00
|
|
|
plugin_init (GstPlugin *plugin)
|
2001-12-22 23:26:48 +00:00
|
|
|
{
|
|
|
|
sink_temp = mp3_sink_factory ();
|
|
|
|
src_temp = mp3_src_factory ();
|
|
|
|
|
2003-11-02 21:10:18 +00:00
|
|
|
return gst_element_register (plugin, "mp3parse",
|
|
|
|
GST_RANK_NONE, GST_TYPE_MP3PARSE);
|
2001-12-22 23:26:48 +00:00
|
|
|
}
|
|
|
|
|
2003-11-02 21:10:18 +00:00
|
|
|
GST_PLUGIN_DEFINE (
|
2001-12-22 23:26:48 +00:00
|
|
|
GST_VERSION_MAJOR,
|
|
|
|
GST_VERSION_MINOR,
|
2003-11-02 21:10:18 +00:00
|
|
|
"mpegaudioparse",
|
|
|
|
"MPEG-1 layer 1/2/3 audio parser",
|
|
|
|
plugin_init,
|
|
|
|
VERSION,
|
|
|
|
"LGPL",
|
|
|
|
GST_PACKAGE,
|
|
|
|
GST_ORIGIN
|
|
|
|
)
|