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gstreamer/gst/mpegaudioparse/gstmpegaudioparse.c
Andy Wingo 6863dd9240 a hack to work around intltool's brokenness a current check for mpeg2dec details->klass reorganizations an element br... 
Original commit message from CVS:
* a hack to work around intltool's brokenness
* a current check for mpeg2dec
* details->klass reorganizations
* an element browser that uses details->klass
* separated cdxa parse out from the avi directory
2002-04-20 21:42:51 +00:00

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*#define GST_DEBUG_ENABLED */
#include <gstmpegaudioparse.h>
/* elementfactory information */
static GstElementDetails mp3parse_details = {
"MPEG1 Audio Parser",
"Codec/Parser",
"Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
VERSION,
"Erik Walthinsen <omega@cse.ogi.edu>",
"(C) 1999",
};
static GstPadTemplate*
mp3_src_factory (void)
{
return
gst_pad_template_new (
"src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
gst_caps_new (
"mp3parse_src",
"audio/mp3",
/*
gst_props_new (
"layer", GST_PROPS_INT_RANGE (1, 3),
"bitrate", GST_PROPS_INT_RANGE (8, 320),
"framed", GST_PROPS_BOOLEAN (TRUE),
*/
NULL),
NULL);
}
static GstPadTemplate*
mp3_sink_factory (void)
{
return
gst_pad_template_new (
"sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
gst_caps_new (
"mp3parse_sink",
"audio/mp3",
NULL),
NULL);
};
/* GstMPEGAudioParse signals and args */
enum {
/* FILL ME */
LAST_SIGNAL
};
enum {
ARG_0,
ARG_SKIP,
ARG_BIT_RATE,
/* FILL ME */
};
static GstPadTemplate *sink_temp, *src_temp;
static void gst_mp3parse_class_init (GstMPEGAudioParseClass *klass);
static void gst_mp3parse_init (GstMPEGAudioParse *mp3parse);
static void gst_mp3parse_loop (GstElement *element);
static void gst_mp3parse_chain (GstPad *pad,GstBuffer *buf);
static long bpf_from_header (GstMPEGAudioParse *parse, unsigned long header);
static int head_check (unsigned long head);
static void gst_mp3parse_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec);
static void gst_mp3parse_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
static GstElementClass *parent_class = NULL;
/*static guint gst_mp3parse_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_mp3parse_get_type(void) {
static GType mp3parse_type = 0;
if (!mp3parse_type) {
static const GTypeInfo mp3parse_info = {
sizeof(GstMPEGAudioParseClass), NULL,
NULL,
(GClassInitFunc)gst_mp3parse_class_init,
NULL,
NULL,
sizeof(GstMPEGAudioParse),
0,
(GInstanceInitFunc)gst_mp3parse_init,
};
mp3parse_type = g_type_register_static(GST_TYPE_ELEMENT, "GstMPEGAudioParse", &mp3parse_info, 0);
}
return mp3parse_type;
}
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",
G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); /* CHECKME */
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_BIT_RATE,
g_param_spec_int("bit_rate","bit_rate","bit_rate",
G_MININT,G_MAXINT,0,G_PARAM_READABLE)); /* CHECKME */
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;
}
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_element_set_loop_function (GST_ELEMENT(mp3parse),gst_mp3parse_loop);
#if 1 /* set this to one to use the old chaining code */
gst_pad_set_chain_function(mp3parse->sinkpad,gst_mp3parse_chain);
gst_element_set_loop_function (GST_ELEMENT(mp3parse),NULL);
#endif
mp3parse->srcpad = gst_pad_new_from_template(src_temp, "src");
gst_element_add_pad(GST_ELEMENT(mp3parse),mp3parse->srcpad);
/*gst_pad_set_type_id(mp3parse->srcpad, mp3frametype); */
mp3parse->partialbuf = NULL;
mp3parse->skip = 0;
mp3parse->in_flush = FALSE;
}
static guint32
gst_mp3parse_next_header (guchar *buf,guint32 len,guint32 start)
{
guint32 offset = start;
int f = 0;
while (offset < (len - 4)) {
fprintf(stderr,"%02x ",buf[offset]);
if (buf[offset] == 0xff)
f = 1;
else if (f && ((buf[offset] >> 4) == 0x0f))
return offset - 1;
else
f = 0;
offset++;
}
return -1;
}
static void
gst_mp3parse_loop (GstElement *element)
{
GstMPEGAudioParse *parse = GST_MP3PARSE(element);
GstBuffer *inbuf, *outbuf;
guint32 size, offset;
guchar *data;
guint32 start;
guint32 header;
gint bpf;
while (1) {
/* get a new buffer */
inbuf = gst_pad_pull (parse->sinkpad);
size = GST_BUFFER_SIZE (inbuf);
data = GST_BUFFER_DATA (inbuf);
offset = 0;
fprintf(stderr, "have buffer of %d bytes\n",size);
/* loop through it and find all the frames */
while (offset < (size - 4)) {
start = gst_mp3parse_next_header (data,size,offset);
fprintf(stderr, "skipped %d bytes searching for the next header\n",start-offset);
header = GULONG_FROM_BE(*((guint32 *)(data+start)));
fprintf(stderr, "header is 0x%08x\n",header);
/* figure out how big the frame is supposed to be */
bpf = bpf_from_header (parse, header);
/* see if there are enough bytes in this buffer for the whole frame */
if ((start + bpf) <= size) {
outbuf = gst_buffer_create_sub (inbuf,start,bpf);
fprintf(stderr, "sending buffer of %d bytes\n",bpf);
gst_pad_push (parse->srcpad, outbuf);
offset = start + bpf;
/* if not, we have to deal with it somehow */
} else {
fprintf(stderr,"don't have enough data for this frame\n");
break;
}
}
}
}
static void
gst_mp3parse_chain (GstPad *pad, GstBuffer *buf)
{
GstMPEGAudioParse *mp3parse;
guchar *data;
glong size,offset = 0;
unsigned long header;
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);
/* g_return_if_fail(GST_IS_BUFFER(buf)); */
mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
GST_DEBUG (0,"mp3parse: received buffer of %d bytes",GST_BUFFER_SIZE(buf));
last_ts = GST_BUFFER_TIMESTAMP(buf);
/* FIXME, do flush */
/*
if (mp3parse->partialbuf) {
gst_buffer_unref(mp3parse->partialbuf);
mp3parse->partialbuf = NULL;
}
mp3parse->in_flush = TRUE;
*/
/* if we have something left from the previous frame */
if (mp3parse->partialbuf) {
mp3parse->partialbuf = gst_buffer_append(mp3parse->partialbuf, buf);
/* and the one we received.. */
gst_buffer_unref(buf);
}
else {
mp3parse->partialbuf = buf;
}
size = GST_BUFFER_SIZE(mp3parse->partialbuf);
data = GST_BUFFER_DATA(mp3parse->partialbuf);
/* while we still have bytes left -4 for the header */
while (offset < size-4) {
int skipped = 0;
GST_DEBUG (0,"mp3parse: offset %ld, size %ld ",offset, size);
/* search for a possible start byte */
for (;((data[offset] != 0xff) && (offset < size));offset++) skipped++;
if (skipped && !mp3parse->in_flush) {
GST_DEBUG (0,"mp3parse: **** now at %ld skipped %d bytes",offset,skipped);
}
/* construct the header word */
header = GULONG_FROM_BE(*((gulong *)(data+offset)));
/* if it's a valid header, go ahead and send off the frame */
if (head_check(header)) {
/* calculate the bpf of the frame */
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 ) {
unsigned long header2;
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 */
header2 = GULONG_FROM_BE(*((gulong *)(data+offset+bpf)));
GST_DEBUG(0,"mp3parse: header=%08lX, header2=%08lX, bpf=%d", header, header2, bpf );
#define HDRMASK ~( (0xF<<12)/*bitrate*/ | (1<<9)/*padding*/ | (3<<4)/*mode extension*/ ) /* mask the bits which are allowed to differ between frames */
if ( (header2&HDRMASK) != (header&HDRMASK) ) { /* require 2 matching headers in a row */
GST_DEBUG(0,"mp3parse: next header doesn't match (header=%08lX, header2=%08lX, bpf=%d)", header, header2, bpf );
offset++; /* This frame is invalid. Start looking for a valid frame at the next position in the stream */
continue;
}
}
/* if we don't have the whole frame... */
if ((size - offset) < bpf) {
GST_DEBUG (0,"mp3parse: partial buffer needed %ld < %d ",(size-offset), bpf);
break;
} else {
outbuf = gst_buffer_create_sub(mp3parse->partialbuf,offset,bpf);
offset += bpf;
if (mp3parse->skip == 0) {
GST_DEBUG (0,"mp3parse: pushing buffer of %d bytes",GST_BUFFER_SIZE(outbuf));
if (mp3parse->in_flush) {
/* FIXME do some sort of flush event */
mp3parse->in_flush = FALSE;
}
GST_BUFFER_TIMESTAMP(outbuf) = last_ts;
gst_pad_push(mp3parse->srcpad,outbuf);
}
else {
GST_DEBUG (0,"mp3parse: skipping buffer of %d bytes",GST_BUFFER_SIZE(outbuf));
gst_buffer_unref(outbuf);
mp3parse->skip--;
}
}
} else {
offset++;
if (!mp3parse->in_flush) GST_DEBUG (0,"mp3parse: *** wrong header, skipping byte (FIXME?)");
}
}
/* if we have processed this block and there are still */
/* bytes left not in a partial block, copy them over. */
if (size-offset > 0) {
glong remainder = (size - offset);
GST_DEBUG (0,"mp3parse: partial buffer needed %ld for trailing bytes",remainder);
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 int mp3parse_tabsel[2][3][16] =
{ { {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448, },
{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, },
{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, } },
{ {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256, },
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, },
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, } },
};
static long mp3parse_freqs[9] =
{44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000};
static long
bpf_from_header (GstMPEGAudioParse *parse, unsigned long header)
{
int layer_index,layer,lsf,samplerate_index,padding;
long bpf;
/*mpegver = (header >> 19) & 0x3; // don't need this for bpf */
layer_index = (header >> 17) & 0x3;
layer = 4 - layer_index;
lsf = (header & (1 << 20)) ? ((header & (1 << 19)) ? 0 : 1) : 1;
parse->bit_rate = mp3parse_tabsel[lsf][layer - 1][((header >> 12) & 0xf)];
samplerate_index = (header >> 10) & 0x3;
padding = (header >> 9) & 0x1;
if (layer == 1) {
bpf = parse->bit_rate * 12000;
bpf /= mp3parse_freqs[samplerate_index];
bpf = ((bpf + padding) << 2);
} else {
bpf = parse->bit_rate * 144000;
bpf /= mp3parse_freqs[samplerate_index];
bpf += padding;
}
/*g_print("%08x: layer %d lsf %d bitrate %d samplerate_index %d padding %d - bpf %d\n", */
/*header,layer,lsf,bitrate,samplerate_index,padding,bpf); */
return bpf;
}
static gboolean
head_check (unsigned long head)
{
GST_DEBUG (0,"checking mp3 header 0x%08lx",head);
/* if it's not a valid sync */
if ((head & 0xffe00000) != 0xffe00000) {
GST_DEBUG (0,"invalid sync");return FALSE; }
/* if it's an invalid MPEG version */
if (((head >> 19) & 3) == 0x1) {
GST_DEBUG (0,"invalid MPEG version");return FALSE; }
/* if it's an invalid layer */
if (!((head >> 17) & 3)) {
GST_DEBUG (0,"invalid layer");return FALSE; }
/* if it's an invalid bitrate */
if (((head >> 12) & 0xf) == 0x0) {
GST_DEBUG (0,"invalid bitrate");return FALSE; }
if (((head >> 12) & 0xf) == 0xf) {
GST_DEBUG (0,"invalid bitrate");return FALSE; }
/* if it's an invalid samplerate */
if (((head >> 10) & 0x3) == 0x3) {
GST_DEBUG (0,"invalid samplerate");return FALSE; }
if ((head & 0xffff0000) == 0xfffe0000) {
GST_DEBUG (0,"invalid sync");return FALSE; }
if (head & 0x00000002) {
GST_DEBUG (0,"invalid emphasis");return FALSE; }
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;
}
}
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
};
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