mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-28 04:31:06 +00:00
d11dbb0338
Original commit message from CVS: * a hack to work around intltool's brokenness * a current check for mpeg2dec * details->klass reorganizations * an element browser that uses details->klass * separated cdxa parse out from the avi directory
356 lines
10 KiB
C
356 lines
10 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.
|
|
*/
|
|
|
|
|
|
#include <stdlib.h>
|
|
#include <string.h>
|
|
|
|
#include <vorbis/vorbisenc.h>
|
|
|
|
#include "vorbisenc.h"
|
|
|
|
|
|
|
|
extern GstPadTemplate *enc_src_template, *enc_sink_template;
|
|
|
|
/* elementfactory information */
|
|
GstElementDetails vorbisenc_details = {
|
|
"Ogg Vorbis encoder",
|
|
"Codec/Audio/Encoder",
|
|
"Encodes audio in OGG Vorbis format",
|
|
VERSION,
|
|
"Monty <monty@xiph.org>, "
|
|
"Wim Taymans <wim.taymans@chello.be>",
|
|
"(C) 2000",
|
|
};
|
|
|
|
/* VorbisEnc signals and args */
|
|
enum
|
|
{
|
|
/* FILL ME */
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
enum
|
|
{
|
|
ARG_0,
|
|
ARG_BITRATE,
|
|
};
|
|
|
|
static void gst_vorbisenc_class_init (VorbisEncClass * klass);
|
|
static void gst_vorbisenc_init (VorbisEnc * vorbisenc);
|
|
|
|
static void gst_vorbisenc_chain (GstPad * pad, GstBuffer * buf);
|
|
static void gst_vorbisenc_setup (VorbisEnc * vorbisenc);
|
|
|
|
static void gst_vorbisenc_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec);
|
|
static void gst_vorbisenc_set_property (GObject * object, guint prop_id, const GValue * value,
|
|
GParamSpec * pspec);
|
|
|
|
static GstElementClass *parent_class = NULL;
|
|
/*static guint gst_vorbisenc_signals[LAST_SIGNAL] = { 0 }; */
|
|
|
|
GType
|
|
vorbisenc_get_type (void)
|
|
{
|
|
static GType vorbisenc_type = 0;
|
|
|
|
if (!vorbisenc_type) {
|
|
static const GTypeInfo vorbisenc_info = {
|
|
sizeof (VorbisEncClass),
|
|
NULL,
|
|
NULL,
|
|
(GClassInitFunc) gst_vorbisenc_class_init,
|
|
NULL,
|
|
NULL,
|
|
sizeof (VorbisEnc),
|
|
0,
|
|
(GInstanceInitFunc) gst_vorbisenc_init,
|
|
};
|
|
|
|
vorbisenc_type = g_type_register_static (GST_TYPE_ELEMENT, "VorbisEnc", &vorbisenc_info, 0);
|
|
}
|
|
return vorbisenc_type;
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_class_init (VorbisEncClass * 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_BITRATE,
|
|
g_param_spec_int ("bitrate", "bitrate", "bitrate",
|
|
G_MININT, G_MAXINT, 0, G_PARAM_READWRITE));
|
|
|
|
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
|
|
|
|
gobject_class->set_property = gst_vorbisenc_set_property;
|
|
gobject_class->get_property = gst_vorbisenc_get_property;
|
|
}
|
|
|
|
static GstPadConnectReturn
|
|
gst_vorbisenc_sinkconnect (GstPad * pad, GstCaps * caps)
|
|
{
|
|
VorbisEnc *vorbisenc;
|
|
|
|
vorbisenc = GST_VORBISENC (gst_pad_get_parent (pad));
|
|
|
|
if (!GST_CAPS_IS_FIXED (caps))
|
|
return GST_PAD_CONNECT_DELAYED;
|
|
|
|
gst_caps_get_int (caps, "channels", &vorbisenc->channels);
|
|
gst_caps_get_int (caps, "rate", &vorbisenc->frequency);
|
|
|
|
gst_vorbisenc_setup (vorbisenc);
|
|
|
|
if (vorbisenc->setup)
|
|
return GST_PAD_CONNECT_OK;
|
|
|
|
return GST_PAD_CONNECT_REFUSED;
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_init (VorbisEnc * vorbisenc)
|
|
{
|
|
vorbisenc->sinkpad = gst_pad_new_from_template (enc_sink_template, "sink");
|
|
gst_element_add_pad (GST_ELEMENT (vorbisenc), vorbisenc->sinkpad);
|
|
gst_pad_set_chain_function (vorbisenc->sinkpad, gst_vorbisenc_chain);
|
|
gst_pad_set_connect_function (vorbisenc->sinkpad, gst_vorbisenc_sinkconnect);
|
|
|
|
vorbisenc->srcpad = gst_pad_new_from_template (enc_src_template, "src");
|
|
gst_element_add_pad (GST_ELEMENT (vorbisenc), vorbisenc->srcpad);
|
|
|
|
vorbisenc->channels = 2;
|
|
vorbisenc->frequency = 44100;
|
|
vorbisenc->bitrate = 128000;
|
|
vorbisenc->setup = FALSE;
|
|
|
|
/* we're chained and we can deal with events */
|
|
GST_FLAG_SET (vorbisenc, GST_ELEMENT_EVENT_AWARE);
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_setup (VorbisEnc * vorbisenc)
|
|
{
|
|
static const gchar *comment = "Track encoded with GStreamer";
|
|
/********** Encode setup ************/
|
|
|
|
/* choose an encoding mode */
|
|
/* (mode 0: 44kHz stereo uncoupled, roughly 128kbps VBR) */
|
|
vorbis_info_init (&vorbisenc->vi);
|
|
vorbis_encode_init (&vorbisenc->vi, vorbisenc->channels, vorbisenc->frequency,
|
|
-1, vorbisenc->bitrate, -1);
|
|
|
|
/* add a comment */
|
|
vorbis_comment_init (&vorbisenc->vc);
|
|
vorbis_comment_add (&vorbisenc->vc, (gchar *)comment);
|
|
/*
|
|
gst_element_send_event (GST_ELEMENT (vorbisenc),
|
|
gst_event_new_info ("comment", GST_PROPS_STRING (comment), NULL));
|
|
*/
|
|
|
|
/* set up the analysis state and auxiliary encoding storage */
|
|
vorbis_analysis_init (&vorbisenc->vd, &vorbisenc->vi);
|
|
vorbis_block_init (&vorbisenc->vd, &vorbisenc->vb);
|
|
|
|
/* set up our packet->stream encoder */
|
|
/* pick a random serial number; that way we can more likely build
|
|
chained streams just by concatenation */
|
|
srand (time (NULL));
|
|
ogg_stream_init (&vorbisenc->os, rand ());
|
|
|
|
/* Vorbis streams begin with three headers; the initial header (with
|
|
most of the codec setup parameters) which is mandated by the Ogg
|
|
bitstream spec. The second header holds any comment fields. The
|
|
third header holds the bitstream codebook. We merely need to
|
|
make the headers, then pass them to libvorbis one at a time;
|
|
libvorbis handles the additional Ogg bitstream constraints */
|
|
|
|
{
|
|
ogg_packet header;
|
|
ogg_packet header_comm;
|
|
ogg_packet header_code;
|
|
|
|
vorbis_analysis_headerout (&vorbisenc->vd, &vorbisenc->vc, &header, &header_comm, &header_code);
|
|
ogg_stream_packetin (&vorbisenc->os, &header); /* automatically placed in its own
|
|
page */
|
|
ogg_stream_packetin (&vorbisenc->os, &header_comm);
|
|
ogg_stream_packetin (&vorbisenc->os, &header_code);
|
|
|
|
/* no need to write out here. We'll get to that in the main loop */
|
|
}
|
|
|
|
vorbisenc->setup = TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_chain (GstPad * pad, GstBuffer * buf)
|
|
{
|
|
VorbisEnc *vorbisenc;
|
|
|
|
g_return_if_fail (pad != NULL);
|
|
g_return_if_fail (GST_IS_PAD (pad));
|
|
g_return_if_fail (buf != NULL);
|
|
|
|
vorbisenc = GST_VORBISENC (gst_pad_get_parent (pad));
|
|
|
|
if (!vorbisenc->setup) {
|
|
gst_element_error (GST_ELEMENT (vorbisenc), "encoder not initialized (input is not audio?)");
|
|
if (GST_IS_BUFFER (buf))
|
|
gst_buffer_unref (buf);
|
|
else
|
|
gst_pad_event_default (pad, GST_EVENT (buf));
|
|
return;
|
|
}
|
|
|
|
if (GST_IS_EVENT (buf)) {
|
|
switch (GST_EVENT_TYPE (buf)) {
|
|
case GST_EVENT_EOS:
|
|
/* end of file. this can be done implicitly in the mainline,
|
|
but it's easier to see here in non-clever fashion.
|
|
Tell the library we're at end of stream so that it can handle
|
|
the last frame and mark end of stream in the output properly */
|
|
vorbis_analysis_wrote (&vorbisenc->vd, 0);
|
|
default:
|
|
gst_pad_event_default (pad, GST_EVENT (buf));
|
|
break;
|
|
}
|
|
}
|
|
else {
|
|
gint16 *data;
|
|
gulong size;
|
|
gulong i, j;
|
|
float **buffer;
|
|
|
|
/* data to encode */
|
|
data = (gint16 *) GST_BUFFER_DATA (buf);
|
|
size = GST_BUFFER_SIZE (buf) / 2;
|
|
|
|
/* expose the buffer to submit data */
|
|
buffer = vorbis_analysis_buffer (&vorbisenc->vd, size / vorbisenc->channels);
|
|
|
|
/* uninterleave samples */
|
|
for (i = 0; i < size / vorbisenc->channels; i++) {
|
|
for (j = 0; j < vorbisenc->channels; j++)
|
|
buffer[j][i] = data[i * vorbisenc->channels + j] / 32768.f;
|
|
}
|
|
|
|
/* tell the library how much we actually submitted */
|
|
vorbis_analysis_wrote (&vorbisenc->vd, size / vorbisenc->channels);
|
|
|
|
gst_buffer_unref (buf);
|
|
}
|
|
|
|
/* vorbis does some data preanalysis, then divvies up blocks for
|
|
more involved (potentially parallel) processing. Get a single
|
|
block for encoding now */
|
|
while (vorbis_analysis_blockout (&vorbisenc->vd, &vorbisenc->vb) == 1) {
|
|
|
|
/* analysis */
|
|
vorbis_analysis (&vorbisenc->vb, NULL);
|
|
vorbis_bitrate_addblock(&vorbisenc->vb);
|
|
|
|
while(vorbis_bitrate_flushpacket(&vorbisenc->vd, &vorbisenc->op)) {
|
|
|
|
/* weld the packet into the bitstream */
|
|
ogg_stream_packetin (&vorbisenc->os, &vorbisenc->op);
|
|
|
|
/* write out pages (if any) */
|
|
while (!vorbisenc->eos) {
|
|
int result = ogg_stream_pageout (&vorbisenc->os, &vorbisenc->og);
|
|
GstBuffer *outbuf;
|
|
|
|
if (result == 0)
|
|
break;
|
|
|
|
outbuf = gst_buffer_new ();
|
|
GST_BUFFER_DATA (outbuf) = g_malloc (vorbisenc->og.header_len + vorbisenc->og.body_len);
|
|
GST_BUFFER_SIZE (outbuf) = vorbisenc->og.header_len + vorbisenc->og.body_len;
|
|
|
|
memcpy (GST_BUFFER_DATA (outbuf), vorbisenc->og.header, vorbisenc->og.header_len);
|
|
memcpy (GST_BUFFER_DATA (outbuf) + vorbisenc->og.header_len, vorbisenc->og.body,
|
|
vorbisenc->og.body_len);
|
|
|
|
GST_DEBUG (0, "vorbisenc: encoded buffer of %d bytes", GST_BUFFER_SIZE (outbuf));
|
|
|
|
gst_pad_push (vorbisenc->srcpad, outbuf);
|
|
|
|
/* this could be set above, but for illustrative purposes, I do
|
|
it here (to show that vorbis does know where the stream ends) */
|
|
if (ogg_page_eos (&vorbisenc->og)) {
|
|
vorbisenc->eos = 1;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (vorbisenc->eos) {
|
|
/* clean up and exit. vorbis_info_clear() must be called last */
|
|
|
|
ogg_stream_clear (&vorbisenc->os);
|
|
vorbis_block_clear (&vorbisenc->vb);
|
|
vorbis_dsp_clear (&vorbisenc->vd);
|
|
vorbis_info_clear (&vorbisenc->vi);
|
|
}
|
|
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec)
|
|
{
|
|
VorbisEnc *vorbisenc;
|
|
|
|
/* it's not null if we got it, but it might not be ours */
|
|
g_return_if_fail (GST_IS_VORBISENC (object));
|
|
|
|
vorbisenc = GST_VORBISENC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_BITRATE:
|
|
g_value_set_int (value, vorbisenc->bitrate);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_set_property (GObject * object, guint prop_id, const GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
VorbisEnc *vorbisenc;
|
|
|
|
/* it's not null if we got it, but it might not be ours */
|
|
g_return_if_fail (GST_IS_VORBISENC (object));
|
|
|
|
vorbisenc = GST_VORBISENC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_BITRATE:
|
|
vorbisenc->bitrate = g_value_get_int (value);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|