gstreamer/gst/mpegaudioparse/gstmpegaudioparse.c

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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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstmpegaudioparse.h"
GST_DEBUG_CATEGORY_STATIC (mp3parse_debug);
#define GST_CAT_DEFAULT mp3parse_debug
/* 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 gboolean gst_mp3parse_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer);
static int head_check (GstMPEGAudioParse * mp3parse, unsigned long head);
static void gst_mp3parse_dispose (GObject * object);
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 GstStateChangeReturn gst_mp3parse_change_state (GstElement * element,
GstStateChange transition);
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 (GstMPEGAudioParse * mp3parse, 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_OBJECT (mp3parse, "Calculated mp3 frame length of %u bytes",
length);
GST_DEBUG_OBJECT (mp3parse, "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;
}
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;
parent_class = g_type_class_peek_parent (klass);
gobject_class->set_property = gst_mp3parse_set_property;
gobject_class->get_property = gst_mp3parse_get_property;
gobject_class->dispose = gst_mp3parse_dispose;
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));
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));
gstelement_class->change_state = gst_mp3parse_change_state;
}
static void
gst_mp3parse_reset (GstMPEGAudioParse * mp3parse)
{
mp3parse->skip = 0;
mp3parse->resyncing = TRUE;
mp3parse->next_ts = -1;
mp3parse->last_ts = -1;
gst_adapter_clear (mp3parse->adapter);
mp3parse->rate = mp3parse->channels = mp3parse->layer = -1;
}
static void
gst_mp3parse_init (GstMPEGAudioParse * mp3parse)
{
mp3parse->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&mp3_sink_template), "sink");
gst_pad_set_event_function (mp3parse->sinkpad, gst_mp3parse_sink_event);
gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain);
gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad);
mp3parse->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&mp3_src_template), "src");
gst_pad_use_fixed_caps (mp3parse->srcpad);
gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad);
mp3parse->adapter = gst_adapter_new ();
gst_mp3parse_reset (mp3parse);
}
static void
gst_mp3parse_dispose (GObject * object)
{
GstMPEGAudioParse *mp3parse = GST_MP3PARSE (object);
if (mp3parse->adapter) {
g_object_unref (mp3parse->adapter);
mp3parse->adapter = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static gboolean
gst_mp3parse_sink_event (GstPad * pad, GstEvent * event)
{
gboolean res;
GstMPEGAudioParse *mp3parse;
mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
{
GstFormat format;
gst_event_parse_new_segment (event, NULL, NULL, &format, NULL, NULL,
NULL);
mp3parse->next_ts = -1;
mp3parse->last_ts = -1;
break;
}
default:
break;
}
res = gst_pad_push_event (mp3parse->srcpad, event);
gst_object_unref (mp3parse);
return res;
}
static GstFlowReturn
gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
{
GstFlowReturn flow = GST_FLOW_OK;
GstMPEGAudioParse *mp3parse;
const guchar *data;
guint32 header;
int bpf;
GstBuffer *outbuf;
GstClockTime timestamp;
guint available;
mp3parse = GST_MP3PARSE (GST_PAD_PARENT (pad));
GST_LOG_OBJECT (mp3parse, "buffer of %d bytes", GST_BUFFER_SIZE (buf));
timestamp = GST_BUFFER_TIMESTAMP (buf);
/* If we don't yet have a next timestamp, and this is valid, use it */
if (!GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts) &&
GST_CLOCK_TIME_IS_VALID (timestamp))
mp3parse->next_ts = timestamp;
gst_adapter_push (mp3parse->adapter, buf);
/* while we still have at least 4 bytes (for the header) available */
while (gst_adapter_available (mp3parse->adapter) >= 4) {
/* search for a possible start byte */
data = gst_adapter_peek (mp3parse->adapter, 4);
if (*data != 0xff) {
/* It'd be nice to make this efficient, but it's ok for now; this is only
* when resyncing
*/
mp3parse->resyncing = TRUE;
gst_adapter_flush (mp3parse->adapter, 1);
continue;
}
available = gst_adapter_available (mp3parse->adapter);
/* construct the header word */
header = GST_READ_UINT32_BE (data);
/* if it's a valid header, go ahead and send off the frame */
if (head_check (mp3parse, header)) {
guint bitrate = 0, layer = 0, rate = 0, channels = 0;
if (!(bpf = mp3_type_frame_length_from_header (mp3parse, header, &layer,
&channels, &bitrate, &rate))) {
g_error ("Header failed internal error");
}
/*************************************************************************
* robust seek support
* - This performs additional frame validation if the resyncing 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 borrowing
* bitrate from previous frames. In this case, seeking may be more
* complicated because the frames are not independently coded.
*************************************************************************/
if (mp3parse->resyncing) {
guint32 header2;
const guint8 *data2;
/* wait until we have the the entire current frame as well as the next
* frame header */
if (available < bpf + 4)
break;
data2 = gst_adapter_peek (mp3parse->adapter, bpf + 4);
header2 = GST_READ_UINT32_BE (data2 + bpf);
GST_DEBUG_OBJECT (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 */
/* require 2 matching headers in a row */
if ((header2 & HDRMASK) != (header & HDRMASK)) {
GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
"(header=%08X, header2=%08X, bpf=%d)",
(unsigned int) header, (unsigned int) header2, bpf);
/* This frame is invalid. Start looking for a valid frame at the
* next position in the stream */
mp3parse->resyncing = TRUE;
gst_adapter_flush (mp3parse->adapter, 1);
continue;
}
}
/* if we don't have the whole frame... */
if (available < bpf) {
GST_DEBUG_OBJECT (mp3parse, "insufficient data available, need "
"%d bytes, have %d", bpf, available);
break;
} else {
if (channels != mp3parse->channels ||
rate != mp3parse->rate ||
layer != mp3parse->layer || bitrate != mp3parse->bit_rate) {
GstCaps *caps;
caps = mp3_caps_create (layer, channels, bitrate, rate);
gst_pad_set_caps (mp3parse->srcpad, caps);
gst_caps_unref (caps);
mp3parse->channels = channels;
mp3parse->layer = layer;
mp3parse->rate = rate;
mp3parse->bit_rate = bitrate;
}
outbuf = gst_adapter_take_buffer (mp3parse->adapter, bpf);
if (!mp3parse->skip) {
gint spf; /* samples per frame */
mp3parse->resyncing = FALSE;
GST_DEBUG_OBJECT (mp3parse, "pushing buffer of %d bytes",
GST_BUFFER_SIZE (outbuf));
/* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */
if (mp3parse->layer == 1)
spf = 384;
else if (mp3parse->layer == 2)
spf = 1152;
else {
/* Any sample_rate < 32000 indicates MPEG-2 or MPEG-2.5 */
if (mp3parse->rate < 32000)
spf = 576;
else
spf = 1152;
}
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (GST_SECOND, spf, mp3parse->rate);
if (GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts)) {
GST_BUFFER_TIMESTAMP (outbuf) = mp3parse->next_ts;
mp3parse->next_ts = GST_CLOCK_TIME_NONE;
} else if (GST_CLOCK_TIME_IS_VALID (mp3parse->last_ts)) {
GST_BUFFER_TIMESTAMP (outbuf) = mp3parse->last_ts +
GST_BUFFER_DURATION (outbuf);
} else {
GST_BUFFER_TIMESTAMP (outbuf) = 0;
}
mp3parse->last_ts = GST_BUFFER_TIMESTAMP (outbuf);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (mp3parse->srcpad));
flow = gst_pad_push (mp3parse->srcpad, outbuf);
} else {
GST_DEBUG_OBJECT (mp3parse, "skipping buffer of %d bytes",
GST_BUFFER_SIZE (outbuf));
gst_buffer_unref (outbuf);
mp3parse->skip--;
}
}
} else {
mp3parse->resyncing = TRUE;
gst_adapter_flush (mp3parse->adapter, 1);
GST_DEBUG_OBJECT (mp3parse, "wrong header, skipping byte");
}
if (GST_FLOW_IS_FATAL (flow))
break;
}
return flow;
}
static gboolean
head_check (GstMPEGAudioParse * mp3parse, unsigned long head)
{
GST_DEBUG_OBJECT (mp3parse, "checking mp3 header 0x%08lx", head);
/* if it's not a valid sync */
if ((head & 0xffe00000) != 0xffe00000) {
GST_DEBUG_OBJECT (mp3parse, "invalid sync");
return FALSE;
}
/* if it's an invalid MPEG version */
if (((head >> 19) & 3) == 0x1) {
GST_DEBUG_OBJECT (mp3parse, "invalid MPEG version");
return FALSE;
}
/* if it's an invalid layer */
if (!((head >> 17) & 3)) {
GST_DEBUG_OBJECT (mp3parse, "invalid layer");
return FALSE;
}
/* if it's an invalid bitrate */
if (((head >> 12) & 0xf) == 0x0) {
GST_DEBUG_OBJECT (mp3parse, "invalid bitrate");
return FALSE;
}
if (((head >> 12) & 0xf) == 0xf) {
GST_DEBUG_OBJECT (mp3parse, "invalid bitrate");
return FALSE;
}
/* if it's an invalid samplerate */
if (((head >> 10) & 0x3) == 0x3) {
GST_DEBUG_OBJECT (mp3parse, "invalid samplerate");
return FALSE;
}
if ((head & 0xffff0000) == 0xfffe0000) {
GST_DEBUG_OBJECT (mp3parse, "invalid sync");
return FALSE;
}
if (head & 0x00000002) {
GST_DEBUG_OBJECT (mp3parse, "invalid emphasis");
return FALSE;
}
return TRUE;
}
static void
gst_mp3parse_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstMPEGAudioParse *src;
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;
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 GstStateChangeReturn
gst_mp3parse_change_state (GstElement * element, GstStateChange transition)
{
GstMPEGAudioParse *mp3parse;
GstStateChangeReturn result;
mp3parse = GST_MP3PARSE (element);
result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_mp3parse_reset (mp3parse);
break;
default:
break;
}
return result;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (mp3parse_debug, "mp3parse", 0, "MP3 Parser");
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_NAME, GST_PACKAGE_ORIGIN);