mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-19 18:40:31 +00:00
496 lines
15 KiB
C
496 lines
15 KiB
C
/* 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 */
|
|
#ifdef HAVE_CONFIG_H
|
|
#include "config.h"
|
|
#endif
|
|
#include <gstmpegaudioparse.h>
|
|
|
|
|
|
/* elementfactory information */
|
|
static GstElementDetails mp3parse_details = {
|
|
"MPEG1 Audio Parser",
|
|
"Codec/Parser",
|
|
"LGPL",
|
|
"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/mpeg",
|
|
gst_props_new (
|
|
"mpegversion", GST_PROPS_INT (1),
|
|
"layer", GST_PROPS_INT_RANGE (1, 3),
|
|
"rate", GST_PROPS_INT_RANGE (8000, 48000),
|
|
"channels", GST_PROPS_INT_RANGE (1, 2),
|
|
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/mpeg",
|
|
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_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 GstElementStateReturn
|
|
gst_mp3parse_change_state (GstElement *element);
|
|
|
|
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("bitrate","Bitrate","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;
|
|
|
|
gstelement_class->change_state = gst_mp3parse_change_state;
|
|
}
|
|
|
|
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);
|
|
gst_element_set_loop_function (GST_ELEMENT(mp3parse),NULL);
|
|
|
|
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;
|
|
|
|
mp3parse->rate = mp3parse->channels = mp3parse->layer = -1;
|
|
}
|
|
|
|
static void
|
|
gst_mp3parse_chain (GstPad *pad, GstBuffer *buf)
|
|
{
|
|
GstMPEGAudioParse *mp3parse;
|
|
guchar *data;
|
|
glong size,offset = 0;
|
|
guint32 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 ("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_merge(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 ("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 ("mp3parse: **** now at %ld skipped %d bytes",offset,skipped);
|
|
}
|
|
/* construct the header word */
|
|
header = GUINT32_FROM_BE(*((guint32 *)(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 ) {
|
|
guint32 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 = GUINT32_FROM_BE(*((guint32 *)(data+offset+bpf)));
|
|
GST_DEBUG ("mp3parse: header=%08X, header2=%08X, bpf=%d", (unsigned int)header, (unsigned int)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 ("mp3parse: next header doesn't match (header=%08X, header2=%08X, bpf=%d)", (unsigned int)header, (unsigned int)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 ("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 ("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_BUFFER_DURATION(outbuf) = 8 * (GST_SECOND/1000) * GST_BUFFER_SIZE(outbuf) / mp3parse->bit_rate;
|
|
|
|
if (GST_PAD_CAPS (mp3parse->srcpad) != NULL) {
|
|
gst_pad_push(mp3parse->srcpad,outbuf);
|
|
} else {
|
|
GST_DEBUG ("No capsnego yet, delaying buffer push");
|
|
gst_buffer_unref (outbuf);
|
|
}
|
|
}
|
|
else {
|
|
GST_DEBUG ("mp3parse: skipping buffer of %d bytes",GST_BUFFER_SIZE(outbuf));
|
|
gst_buffer_unref(outbuf);
|
|
mp3parse->skip--;
|
|
}
|
|
}
|
|
} else {
|
|
offset++;
|
|
if (!mp3parse->in_flush) GST_DEBUG ("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 ("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,mode;
|
|
long bpf;
|
|
gint channels, rate;
|
|
|
|
/*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;
|
|
mode = (header >> 6) & 0x3;
|
|
|
|
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;
|
|
}
|
|
|
|
channels = (mode == 3) ? 1 : 2;
|
|
rate = mp3parse_freqs[samplerate_index];
|
|
if (channels != parse->channels ||
|
|
rate != parse->rate ||
|
|
layer != parse->layer) {
|
|
GstCaps *caps = GST_CAPS_NEW ("mp3parse_src",
|
|
"audio/mpeg",
|
|
"mpegversion", GST_PROPS_INT (1),
|
|
"layer", GST_PROPS_INT (layer),
|
|
"channels", GST_PROPS_INT (channels),
|
|
"rate", GST_PROPS_INT (rate));
|
|
if (gst_pad_try_set_caps(parse->srcpad, caps) <= 0) {
|
|
gst_element_error (GST_ELEMENT (parse),
|
|
"mp3parse: failed to negotiate format with next element");
|
|
}
|
|
|
|
parse->channels = channels;
|
|
parse->layer = layer;
|
|
parse->rate = rate;
|
|
}
|
|
|
|
/*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 ("checking mp3 header 0x%08lx",head);
|
|
/* if it's not a valid sync */
|
|
if ((head & 0xffe00000) != 0xffe00000) {
|
|
GST_DEBUG ("invalid sync");return FALSE; }
|
|
/* if it's an invalid MPEG version */
|
|
if (((head >> 19) & 3) == 0x1) {
|
|
GST_DEBUG ("invalid MPEG version");return FALSE; }
|
|
/* if it's an invalid layer */
|
|
if (!((head >> 17) & 3)) {
|
|
GST_DEBUG ("invalid layer");return FALSE; }
|
|
/* if it's an invalid bitrate */
|
|
if (((head >> 12) & 0xf) == 0x0) {
|
|
GST_DEBUG ("invalid bitrate");return FALSE; }
|
|
if (((head >> 12) & 0xf) == 0xf) {
|
|
GST_DEBUG ("invalid bitrate");return FALSE; }
|
|
/* if it's an invalid samplerate */
|
|
if (((head >> 10) & 0x3) == 0x3) {
|
|
GST_DEBUG ("invalid samplerate");return FALSE; }
|
|
if ((head & 0xffff0000) == 0xfffe0000) {
|
|
GST_DEBUG ("invalid sync");return FALSE; }
|
|
if (head & 0x00000002) {
|
|
GST_DEBUG ("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 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;
|
|
}
|
|
|
|
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
|
|
};
|