mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-22 08:17:01 +00:00
7f84d638b6
Original commit message from CVS: Various changes and cleanups.
440 lines
13 KiB
C
440 lines
13 KiB
C
/* Gnome-Streamer
|
|
* 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.
|
|
*/
|
|
|
|
#include <string.h>
|
|
#include <sys/soundcard.h>
|
|
|
|
#include <vorbisdec.h>
|
|
|
|
|
|
extern GstPadTemplate *dec_src_template, *dec_sink_template;
|
|
|
|
/* elementfactory information */
|
|
GstElementDetails vorbisdec_details =
|
|
{
|
|
"Ogg Vorbis decoder",
|
|
"Filter/Audio/Decoder",
|
|
"Decodes OGG Vorbis audio",
|
|
VERSION,
|
|
"Monty <monty@xiph.org>, "
|
|
"Wim Taymans <wim.taymans@chello.be>",
|
|
"(C) 2000",
|
|
};
|
|
|
|
/* VorbisDec signals and args */
|
|
enum
|
|
{
|
|
/* FILL ME */
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
enum
|
|
{
|
|
ARG_0,
|
|
};
|
|
|
|
static void gst_vorbisdec_class_init (VorbisDecClass *klass);
|
|
static void gst_vorbisdec_init (VorbisDec *vorbisdec);
|
|
|
|
static void gst_vorbisdec_loop (GstElement *element);
|
|
|
|
static GstElementClass *parent_class = NULL;
|
|
/*static guint gst_vorbisdec_signals[LAST_SIGNAL] = { 0 }; */
|
|
|
|
GType
|
|
vorbisdec_get_type (void)
|
|
{
|
|
static GType vorbisdec_type = 0;
|
|
|
|
if (!vorbisdec_type) {
|
|
static const GTypeInfo vorbisdec_info = {
|
|
sizeof (VorbisDecClass),
|
|
NULL,
|
|
NULL,
|
|
(GClassInitFunc) gst_vorbisdec_class_init,
|
|
NULL,
|
|
NULL,
|
|
sizeof (VorbisDec),
|
|
0,
|
|
(GInstanceInitFunc) gst_vorbisdec_init,
|
|
};
|
|
|
|
vorbisdec_type = g_type_register_static (GST_TYPE_ELEMENT, "VorbisDec", &vorbisdec_info, 0);
|
|
}
|
|
return vorbisdec_type;
|
|
}
|
|
|
|
static void
|
|
gst_vorbisdec_class_init (VorbisDecClass * klass)
|
|
{
|
|
GstElementClass *gstelement_class;
|
|
|
|
gstelement_class = (GstElementClass *) klass;
|
|
|
|
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
|
|
}
|
|
|
|
static void
|
|
gst_vorbisdec_init (VorbisDec * vorbisdec)
|
|
{
|
|
vorbisdec->sinkpad = gst_pad_new_from_template (dec_sink_template, "sink");
|
|
gst_element_add_pad (GST_ELEMENT (vorbisdec), vorbisdec->sinkpad);
|
|
|
|
gst_element_set_loop_function (GST_ELEMENT (vorbisdec), gst_vorbisdec_loop);
|
|
vorbisdec->srcpad = gst_pad_new_from_template (dec_src_template, "src");
|
|
gst_element_add_pad (GST_ELEMENT (vorbisdec), vorbisdec->srcpad);
|
|
|
|
ogg_sync_init (&vorbisdec->oy); /* Now we can read pages */
|
|
vorbisdec->convsize = 4096;
|
|
}
|
|
|
|
static GstBuffer *
|
|
gst_vorbisdec_pull (VorbisDec * vorbisdec, ogg_sync_state * oy)
|
|
{
|
|
GstBuffer *buf;
|
|
|
|
do {
|
|
GST_DEBUG (0, "vorbisdec: pull \n");
|
|
|
|
buf = gst_pad_pull (vorbisdec->sinkpad);
|
|
|
|
if (GST_IS_EVENT (buf)) {
|
|
switch (GST_EVENT_TYPE (buf)) {
|
|
case GST_EVENT_FLUSH:
|
|
ogg_sync_reset (oy);
|
|
case GST_EVENT_EOS:
|
|
default:
|
|
gst_pad_event_default (vorbisdec->sinkpad, GST_EVENT (buf));
|
|
break;
|
|
}
|
|
buf = NULL;
|
|
}
|
|
} while (buf == NULL);
|
|
|
|
GST_DEBUG (0, "vorbisdec: pull done\n");
|
|
|
|
return buf;
|
|
}
|
|
|
|
static void
|
|
gst_vorbisdec_loop (GstElement * element)
|
|
{
|
|
VorbisDec *vorbisdec;
|
|
GstBuffer *buf;
|
|
GstBuffer *outbuf;
|
|
|
|
ogg_sync_state oy; /* sync and verify incoming physical bitstream */
|
|
ogg_stream_state os; /* take physical pages, weld into a logical
|
|
stream of packets */
|
|
ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
|
|
ogg_packet op; /* one raw packet of data for decode */
|
|
|
|
vorbis_info vi; /* struct that stores all the static vorbis bitstream
|
|
settings */
|
|
vorbis_comment vc; /* struct that stores all the bitstream user comments */
|
|
vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
|
|
vorbis_block vb; /* local working space for packet->PCM decode */
|
|
|
|
char *buffer;
|
|
int bytes;
|
|
|
|
g_return_if_fail (element != NULL);
|
|
g_return_if_fail (GST_IS_VORBISDEC (element));
|
|
|
|
vorbisdec = GST_VORBISDEC (element);
|
|
|
|
/********** Decode setup ************/
|
|
ogg_sync_init (&oy); /* Now we can read pages */
|
|
|
|
while (1) { /* we repeat if the bitstream is chained */
|
|
int eos = 0;
|
|
int i;
|
|
|
|
/* grab some data at the head of the stream. We want the first page
|
|
(which is guaranteed to be small and only contain the Vorbis
|
|
stream initial header) We need the first page to get the stream
|
|
serialno. */
|
|
|
|
gst_element_yield (GST_ELEMENT (vorbisdec));
|
|
|
|
/* submit a 4k block to libvorbis' Ogg layer */
|
|
buf = gst_vorbisdec_pull (vorbisdec, &oy);
|
|
|
|
bytes = GST_BUFFER_SIZE (buf);
|
|
buffer = ogg_sync_buffer (&oy, bytes);
|
|
memcpy (buffer, GST_BUFFER_DATA (buf), bytes);
|
|
gst_buffer_unref (buf);
|
|
|
|
ogg_sync_wrote (&oy, bytes);
|
|
|
|
/* Get the first page. */
|
|
if (ogg_sync_pageout (&oy, &og) != 1) {
|
|
/* error case. Must not be Vorbis data */
|
|
gst_element_error (element, "input does not appear to be an Ogg bitstream.");
|
|
break;
|
|
}
|
|
|
|
/* Get the serial number and set up the rest of decode. */
|
|
/* serialno first; use it to set up a logical stream */
|
|
ogg_stream_init (&os, ogg_page_serialno (&og));
|
|
|
|
/* extract the initial header from the first page and verify that the
|
|
Ogg bitstream is in fact Vorbis data */
|
|
|
|
/* I handle the initial header first instead of just having the code
|
|
read all three Vorbis headers at once because reading the initial
|
|
header is an easy way to identify a Vorbis bitstream and it's
|
|
useful to see that functionality seperated out. */
|
|
|
|
vorbis_info_init (&vi);
|
|
vorbis_comment_init (&vc);
|
|
if (ogg_stream_pagein (&os, &og) < 0) {
|
|
/* error; stream version mismatch perhaps */
|
|
g_warning ("Error reading first page of Ogg bitstream data.\n");
|
|
return;
|
|
}
|
|
|
|
if (ogg_stream_packetout (&os, &op) != 1) {
|
|
/* no page? must not be vorbis */
|
|
g_warning ("Error reading initial header packet.\n");
|
|
return;
|
|
}
|
|
|
|
if (vorbis_synthesis_headerin (&vi, &vc, &op) < 0) {
|
|
/* error case; not a vorbis header */
|
|
g_warning ("This Ogg bitstream does not contain Vorbis audio data.\n");
|
|
return;
|
|
}
|
|
|
|
/* At this point, we're sure we're Vorbis. We've set up the logical
|
|
(Ogg) bitstream decoder. Get the comment and codebook headers and
|
|
set up the Vorbis decoder */
|
|
|
|
/* The next two packets in order are the comment and codebook headers.
|
|
They're likely large and may span multiple pages. Thus we reead
|
|
and submit data until we get our two pacakets, watching that no
|
|
pages are missing. If a page is missing, error out; losing a
|
|
header page is the only place where missing data is fatal. */
|
|
|
|
i = 0;
|
|
while (i < 2) {
|
|
while (i < 2) {
|
|
int result = ogg_sync_pageout (&oy, &og);
|
|
|
|
if (result == 0)
|
|
break; /* Need more data */
|
|
/* Don't complain about missing or corrupt data yet. We'll
|
|
catch it at the packet output phase */
|
|
if (result == 1) {
|
|
ogg_stream_pagein (&os, &og); /* we can ignore any errors here
|
|
as they'll also become apparent
|
|
at packetout */
|
|
while (i < 2) {
|
|
result = ogg_stream_packetout (&os, &op);
|
|
if (result == 0)
|
|
break;
|
|
if (result == -1) {
|
|
/* Uh oh; data at some point was corrupted or missing!
|
|
We can't tolerate that in a header. Die. */
|
|
g_warning ("Corrupt secondary header. expect trouble\n");
|
|
}
|
|
vorbis_synthesis_headerin (&vi, &vc, &op);
|
|
i++;
|
|
}
|
|
}
|
|
}
|
|
gst_element_yield (GST_ELEMENT (vorbisdec));
|
|
|
|
buf = gst_vorbisdec_pull (vorbisdec, &oy);
|
|
bytes = GST_BUFFER_SIZE (buf);
|
|
buffer = ogg_sync_buffer (&oy, bytes);
|
|
memcpy (buffer, GST_BUFFER_DATA (buf), bytes);
|
|
gst_buffer_unref (buf);
|
|
|
|
if (bytes == 0 && i < 2) {
|
|
g_warning ("End of file before finding all Vorbis headers! expect trouble..\n");
|
|
}
|
|
ogg_sync_wrote (&oy, bytes);
|
|
}
|
|
|
|
/* Throw the comments plus a few lines about the bitstream we're
|
|
decoding */
|
|
{
|
|
char **ptr = vc.user_comments;
|
|
|
|
while (*ptr) {
|
|
gst_element_send_event (GST_ELEMENT (vorbisdec),
|
|
gst_event_new_info ("comment", GST_PROPS_STRING (*ptr), NULL));
|
|
++ptr;
|
|
}
|
|
gst_element_send_event (GST_ELEMENT (vorbisdec),
|
|
gst_event_new_info ("vendor", GST_PROPS_STRING (vc.vendor), NULL));
|
|
|
|
gst_element_send_event (GST_ELEMENT (vorbisdec),
|
|
gst_event_new_info ("version", GST_PROPS_INT (vi.version), NULL));
|
|
gst_element_send_event (GST_ELEMENT (vorbisdec),
|
|
gst_event_new_info ("channels", GST_PROPS_INT (vi.channels), NULL));
|
|
gst_element_send_event (GST_ELEMENT (vorbisdec),
|
|
gst_event_new_info ("rate", GST_PROPS_INT (vi.rate), NULL));
|
|
gst_element_send_event (GST_ELEMENT (vorbisdec),
|
|
gst_event_new_info ("bitrate_upper", GST_PROPS_INT (vi.bitrate_upper), NULL));
|
|
gst_element_send_event (GST_ELEMENT (vorbisdec),
|
|
gst_event_new_info ("bitrate_nominal", GST_PROPS_INT (vi.bitrate_nominal), NULL));
|
|
gst_element_send_event (GST_ELEMENT (vorbisdec),
|
|
gst_event_new_info ("bitrate_lower", GST_PROPS_INT (vi.bitrate_lower), NULL));
|
|
gst_element_send_event (GST_ELEMENT (vorbisdec),
|
|
gst_event_new_info ("bitrate_window", GST_PROPS_INT (vi.bitrate_window), NULL));
|
|
}
|
|
|
|
gst_pad_set_caps (vorbisdec->srcpad,
|
|
gst_caps_new ("vorbisdec_src",
|
|
"audio/raw",
|
|
gst_props_new ("format", GST_PROPS_STRING ("int"),
|
|
"law", GST_PROPS_INT (0),
|
|
"endianness", GST_PROPS_INT (G_BYTE_ORDER),
|
|
"signed", GST_PROPS_BOOLEAN (TRUE),
|
|
"width", GST_PROPS_INT (16),
|
|
"depth", GST_PROPS_INT (16),
|
|
"rate", GST_PROPS_INT (vi.rate),
|
|
"channels", GST_PROPS_INT (vi.channels),
|
|
NULL)));
|
|
|
|
vorbisdec->convsize = 4096 / vi.channels;
|
|
|
|
/* OK, got and parsed all three headers. Initialize the Vorbis
|
|
packet->PCM decoder. */
|
|
vorbis_synthesis_init (&vd, &vi); /* central decode state */
|
|
vorbis_block_init (&vd, &vb); /* local state for most of the decode
|
|
so multiple block decodes can
|
|
proceed in parallel. We could init
|
|
multiple vorbis_block structures
|
|
for vd here */
|
|
|
|
/* The rest is just a straight decode loop until end of stream */
|
|
while (!eos) {
|
|
while (!eos) {
|
|
int result = ogg_sync_pageout (&oy, &og);
|
|
|
|
if (result == 0)
|
|
break; /* need more data */
|
|
if (result == -1) { /* missing or corrupt data at this page position */
|
|
}
|
|
else {
|
|
ogg_stream_pagein (&os, &og); /* can safely ignore errors at
|
|
this point */
|
|
while (1) {
|
|
result = ogg_stream_packetout (&os, &op);
|
|
|
|
if (result == 0)
|
|
break; /* need more data */
|
|
if (result == -1) { /* missing or corrupt data at this page position */
|
|
/* no reason to complain; already complained above */
|
|
}
|
|
else {
|
|
/* we have a packet. Decode it */
|
|
float **pcm;
|
|
int samples;
|
|
|
|
if (vorbis_synthesis (&vb, &op) == 0) /* test for success! */
|
|
vorbis_synthesis_blockin (&vd, &vb);
|
|
|
|
/*
|
|
**pcm is a multichannel double vector. In stereo, for
|
|
example, pcm[0] is left, and pcm[1] is right. samples is
|
|
the size of each channel. Convert the float values
|
|
(-1.<=range<=1.) to whatever PCM format and write it out */
|
|
|
|
while ((samples = vorbis_synthesis_pcmout (&vd, &pcm)) > 0) {
|
|
int j;
|
|
int clipflag = 0;
|
|
int bout = (samples < vorbisdec->convsize ? samples : vorbisdec->convsize);
|
|
|
|
outbuf = gst_buffer_new ();
|
|
GST_BUFFER_DATA (outbuf) = g_malloc (2 * vi.channels * bout);
|
|
GST_BUFFER_SIZE (outbuf) = 2 * vi.channels * bout;
|
|
|
|
/* convert doubles to 16 bit signed ints (host order) and
|
|
interleave */
|
|
for (i = 0; i < vi.channels; i++) {
|
|
int16_t *ptr = ((int16_t *) GST_BUFFER_DATA (outbuf)) + i;
|
|
float *mono = pcm[i];
|
|
|
|
for (j = 0; j < bout; j++) {
|
|
int val = mono[j] * 32767.;
|
|
|
|
/* might as well guard against clipping */
|
|
if (val > 32767) {
|
|
val = 32767;
|
|
clipflag = 1;
|
|
}
|
|
if (val < -32768) {
|
|
val = -32768;
|
|
clipflag = 1;
|
|
}
|
|
*ptr = val;
|
|
ptr += vi.channels;
|
|
}
|
|
}
|
|
|
|
GST_DEBUG (0, "vorbisdec: push\n");
|
|
gst_pad_push (vorbisdec->srcpad, outbuf);
|
|
GST_DEBUG (0, "vorbisdec: push done\n");
|
|
|
|
vorbis_synthesis_read (&vd, bout); /* tell libvorbis how
|
|
many samples we
|
|
actually consumed */
|
|
}
|
|
}
|
|
}
|
|
if (ogg_page_eos (&og))
|
|
eos = 1;
|
|
}
|
|
}
|
|
if (!eos) {
|
|
gst_element_yield (GST_ELEMENT (vorbisdec));
|
|
|
|
buf = gst_vorbisdec_pull (vorbisdec, &oy);
|
|
bytes = GST_BUFFER_SIZE (buf);
|
|
buffer = ogg_sync_buffer (&oy, bytes);
|
|
memcpy (buffer, GST_BUFFER_DATA (buf), bytes);
|
|
gst_buffer_unref (buf);
|
|
|
|
ogg_sync_wrote (&oy, bytes);
|
|
if (bytes == 0) {
|
|
eos = 1;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* clean up this logical bitstream; before exit we see if we're
|
|
followed by another [chained] */
|
|
|
|
ogg_stream_clear (&os);
|
|
|
|
/* ogg_page and ogg_packet structs always point to storage in
|
|
libvorbis. They're never freed or manipulated directly */
|
|
|
|
vorbis_block_clear (&vb);
|
|
vorbis_dsp_clear (&vd);
|
|
vorbis_info_clear (&vi); /* must be called last */
|
|
}
|
|
|
|
/* OK, clean up the framer */
|
|
ogg_sync_clear (&oy);
|
|
}
|