mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-12 03:16:33 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/mpegaudioparse/gstmpegaudioparse.c
Stéphane Loeuillet a74cdc5f34 first batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc 
Original commit message from CVS:

first batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
2004-05-21 22:39:29 +00:00

595 lines
18 KiB
C
Raw Blame History

/* 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/Audio",
"Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
"Erik Walthinsen <omega@cse.ogi.edu>"
};
static GstStaticPadTemplate mp3_src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, "
"mpegversion = (int) 1, "
"layer = (int) [ 1, 3 ], "
"rate = (int) [ 8000, 48000], " "channels = (int) [ 1, 2 ]")
);
static GstStaticPadTemplate mp3_sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1")
);
/* GstMPEGAudioParse signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_SKIP,
ARG_BIT_RATE
/* FILL ME */
};
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, channels, padding;
gint lsf, mpg25;
if (header & (1 << 20)) {
lsf = (header & (1 << 19)) ? 0 : 1;
mpg25 = 0;
} else {
lsf = 1;
mpg25 = 1;
}
mode = (header >> 6) & 0x3;
channels = (mode == 3) ? 1 : 2;
samplerate = (header >> 10) & 0x3;
samplerate = mp3types_freqs[lsf + mpg25][samplerate];
layer = 4 - ((header >> 17) & 0x3);
bitrate = (header >> 12) & 0xF;
bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
if (bitrate == 0)
return 0;
padding = (header >> 9) & 0x1;
switch (layer) {
case 1:
length = (bitrate * 12) / samplerate + 4 * padding;
break;
case 2:
length = (bitrate * 144) / samplerate + padding;
break;
default:
case 3:
length = (bitrate * 144) / (samplerate << lsf) + padding;
break;
}
GST_DEBUG ("Calculated mp3 frame length of %u bytes", length);
GST_DEBUG ("samplerate = %lu, bitrate = %lu, layer = %lu, channels = %lu",
samplerate, bitrate, layer, 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_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 1,
"layer", G_TYPE_INT, layer,
"rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);
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,
gst_static_pad_template_get (&mp3_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&mp3_src_template));
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 (gst_static_pad_template_get
(&mp3_sink_template), "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 (gst_static_pad_template_get
(&mp3_src_template), "src");
gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad);
gst_pad_use_explicit_caps (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) {
GstBuffer *newbuf;
newbuf = gst_buffer_merge (mp3parse->partialbuf, buf);
/* and the one we received.. */
gst_buffer_unref (buf);
gst_buffer_unref (mp3parse->partialbuf);
mp3parse->partialbuf = newbuf;
} 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 (; ((offset < size - 4) && (data[offset] != 0xff)); offset++)
skipped++;
if (skipped && !mp3parse->in_flush) {
GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes", offset, skipped);
}
/* construct the header word */
header = GST_READ_UINT32_BE (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 = GST_READ_UINT32_BE (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 {
guint bitrate, layer, rate, channels;
if (!mp3_type_frame_length_from_header (header, &layer,
&channels, &bitrate, &rate)) {
g_error ("Header failed internal error");
}
if (channels != mp3parse->channels ||
rate != mp3parse->rate ||
layer != mp3parse->layer || bitrate != mp3parse->bit_rate) {
GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate);
gst_pad_set_explicit_caps (mp3parse->srcpad, caps);
mp3parse->channels = channels;
mp3parse->layer = layer;
mp3parse->rate = rate;
mp3parse->bit_rate = bitrate;
}
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;
}
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)
{
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_PACKAGE, GST_ORIGIN)
Reference in a new issue View git blame Copy permalink