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gstreamer/gst/mpegaudioparse/gstmpegaudioparse.c
Ronald S. Bultje 0498722c8a Whaaaaaaaaaaaaaaaaaaaaaack 
Original commit message from CVS:
Whaaaaaaaaaaaaaaaaaaaaaack
2003-11-02 21:10:18 +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 */
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#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",
"Erik Walthinsen <omega@cse.ogi.edu>"
};
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_base_init (GstMPEGAudioParseClass *klass);
static void gst_mp3parse_init (GstMPEGAudioParse *mp3parse);
static void gst_mp3parse_chain (GstPad *pad,GstData *_data);
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),
(GBaseInitFunc)gst_mp3parse_base_init,
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 guint mp3types_bitrates[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 guint mp3types_freqs[3][3] =
{ {44100, 48000, 32000},
{22050, 24000, 16000},
{11025, 12000, 8000}};
static inline guint
mp3_type_frame_length_from_header (guint32 header, guint *put_layer,
guint *put_channels, guint *put_bitrate,
guint *put_samplerate)
{
guint length;
gulong mode, samplerate, bitrate, layer, version, channels;
/* we don't need extension, copyright, original or
* emphasis for the frame length */
header >>= 6;
/* mode */
mode = header & 0x3;
header >>= 3;
/* padding */
length = header & 0x1;
header >>= 1;
/* sampling frequency */
samplerate = header & 0x3;
if (samplerate == 3)
return 0;
header >>= 2;
/* bitrate index */
bitrate = header & 0xF;
if (bitrate == 15 || bitrate == 0)
return 0;
/* ignore error correction, too */
header >>= 5;
/* layer */
layer = 4 - (header & 0x3);
if (layer == 4)
return 0;
header >>= 2;
/* version */
version = header & 0x3;
if (version == 1)
return 0;
/* lookup */
channels = (mode == 3) ? 1 : 2;
bitrate = mp3types_bitrates[version == 3 ? 0 : 1][layer - 1][bitrate];
samplerate = mp3types_freqs[version > 0 ? version - 1 : 0][samplerate];
/* calculating */
if (layer == 1) {
length = ((12000 * bitrate / samplerate) + length) * 4;
} else {
length += ((layer == 3 && version == 0) ? 144000 : 72000) * bitrate / samplerate;
}
GST_DEBUG ("Calculated mp3 frame length of %u bytes", length);
GST_DEBUG ("samplerate = %lu - bitrate = %lu - layer = %lu - version = %lu"
" - channels = %lu",
samplerate, bitrate, layer, version, channels);
if (put_layer)
*put_layer = layer;
if (put_channels)
*put_channels = channels;
if (put_bitrate)
*put_bitrate = bitrate;
if (put_samplerate)
*put_samplerate = samplerate;
return length;
}
/**
* The chance that random data is identified as a valid mp3 header is 63 / 2^18
* (0.024%) per try. This makes the function for calculating false positives
* 1 - (1 - ((63 / 2 ^18) ^ GST_MP3_TYPEFIND_MIN_HEADERS)) ^ buffersize)
* This has the following probabilities of false positives:
* bufsize MIN_HEADERS
* (bytes) 1 2 3 4
* 4096 62.6% 0.02% 0% 0%
* 16384 98% 0.09% 0% 0%
* 1 MiB 100% 5.88% 0% 0%
* 1 GiB 100% 100% 1.44% 0%
* 1 TiB 100% 100% 100% 0.35%
* This means that the current choice (3 headers by most of the time 4096 byte
* buffers is pretty safe for now.
*
* The max. size of each frame is 1440 bytes, which means that for N frames
* to be detected, we need 1440 * GST_MP3_TYPEFIND_MIN_HEADERS + 3 of data.
* Assuming we step into the stream right after the frame header, this
* means we need 1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3 bytes
* of data (5762) to always detect any mp3.
*/
/* increase this value when this function finds too many false positives */
#define GST_MP3_TYPEFIND_MIN_HEADERS 3
#define GST_MP3_TYPEFIND_MIN_DATA (1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3)
static GstCaps *
mp3_caps_create (guint layer, guint channels,
guint bitrate, guint samplerate)
{
GstCaps *new;
g_assert (layer);
g_assert (samplerate);
g_assert (bitrate);
g_assert (channels);
new = GST_CAPS_NEW ("mp3_type_find",
"audio/mpeg",
"mpegversion", GST_PROPS_INT (1),
"layer", GST_PROPS_INT (layer),
/*"bitrate", GST_PROPS_INT (bitrate),*/
"rate", GST_PROPS_INT (samplerate),
"channels", GST_PROPS_INT (channels));
return new;
}
static void
gst_mp3parse_base_init (GstMPEGAudioParseClass *klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class, sink_temp);
gst_element_class_add_pad_template (element_class, src_temp);
gst_element_class_set_details (element_class, &mp3parse_details);
}
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, GstData *_data)
{
GstBuffer *buf = GST_BUFFER (_data);
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 );
/* mask the bits which are allowed to differ between frames */
#define HDRMASK ~((0xF << 12) /* bitrate */ | \
(0x1 << 9) /* padding */ | \
(0x3 << 4)) /*mode extension*/
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,GST_DATA (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 long
bpf_from_header (GstMPEGAudioParse *parse, unsigned long header)
{
guint bitrate, layer, rate, channels, length;
if (!(length = mp3_type_frame_length_from_header (header, &layer,
&channels,
&bitrate, &rate))) {
return 0;
}
if (channels != parse->channels ||
rate != parse->rate ||
layer != parse->layer ||
bitrate != parse->bit_rate) {
GstCaps *caps = mp3_caps_create (layer, channels, bitrate, 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;
parse->bit_rate = bitrate;
}
return length;
}
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 (GstPlugin *plugin)
{
sink_temp = mp3_sink_factory ();
src_temp = mp3_src_factory ();
return gst_element_register (plugin, "mp3parse",
GST_RANK_NONE, GST_TYPE_MP3PARSE);
}
GST_PLUGIN_DEFINE (
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"mpegaudioparse",
"MPEG-1 layer 1/2/3 audio parser",
plugin_init,
VERSION,
"LGPL",
GST_COPYRIGHT,
GST_PACKAGE,
GST_ORIGIN
)
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