mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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5455dbbf3d
Original commit message from CVS: Bring the plugins in sync with the new core capsnego system. Added some features, enhancements...
357 lines
11 KiB
C
357 lines
11 KiB
C
/* Gnome-Streamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <stdlib.h>
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#include <string.h>
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#include <vorbis/vorbisenc.h>
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#include "vorbisenc.h"
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extern GstPadTemplate *enc_src_template, *enc_sink_template;
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/* elementfactory information */
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GstElementDetails vorbisenc_details = {
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"Ogg Vorbis encoder",
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"Filter/Audio/Encoder",
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"Encodes audio in OGG Vorbis format",
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VERSION,
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"Monty <monty@xiph.org>, "
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"Wim Taymans <wim.taymans@chello.be>",
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"(C) 2000",
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};
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/* VorbisEnc signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0,
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ARG_BITRATE,
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};
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static void gst_vorbisenc_class_init (VorbisEncClass * klass);
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static void gst_vorbisenc_init (VorbisEnc * vorbisenc);
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static void gst_vorbisenc_chain (GstPad * pad, GstBuffer * buf);
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static void gst_vorbisenc_setup (VorbisEnc * vorbisenc);
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static void gst_vorbisenc_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec);
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static void gst_vorbisenc_set_property (GObject * object, guint prop_id, const GValue * value,
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GParamSpec * pspec);
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static GstElementClass *parent_class = NULL;
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/*static guint gst_vorbisenc_signals[LAST_SIGNAL] = { 0 }; */
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GType
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vorbisenc_get_type (void)
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{
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static GType vorbisenc_type = 0;
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if (!vorbisenc_type) {
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static const GTypeInfo vorbisenc_info = {
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sizeof (VorbisEncClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_vorbisenc_class_init,
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NULL,
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NULL,
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sizeof (VorbisEnc),
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0,
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(GInstanceInitFunc) gst_vorbisenc_init,
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};
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vorbisenc_type = g_type_register_static (GST_TYPE_ELEMENT, "VorbisEnc", &vorbisenc_info, 0);
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}
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return vorbisenc_type;
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}
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static void
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gst_vorbisenc_class_init (VorbisEncClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BITRATE,
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g_param_spec_int ("bitrate", "bitrate", "bitrate",
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G_MININT, G_MAXINT, 0, G_PARAM_READWRITE));
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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gobject_class->set_property = gst_vorbisenc_set_property;
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gobject_class->get_property = gst_vorbisenc_get_property;
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}
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static GstPadConnectReturn
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gst_vorbisenc_sinkconnect (GstPad * pad, GstCaps * caps)
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{
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VorbisEnc *vorbisenc;
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vorbisenc = GST_VORBISENC (gst_pad_get_parent (pad));
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if (!GST_CAPS_IS_FIXED (caps))
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return GST_PAD_CONNECT_DELAYED;
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vorbisenc->channels = gst_caps_get_int (caps, "channels");
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vorbisenc->frequency = gst_caps_get_int (caps, "rate");
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gst_vorbisenc_setup (vorbisenc);
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if (vorbisenc->setup)
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return GST_PAD_CONNECT_OK;
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return GST_PAD_CONNECT_REFUSED;
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}
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static void
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gst_vorbisenc_init (VorbisEnc * vorbisenc)
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{
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vorbisenc->sinkpad = gst_pad_new_from_template (enc_sink_template, "sink");
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gst_element_add_pad (GST_ELEMENT (vorbisenc), vorbisenc->sinkpad);
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gst_pad_set_chain_function (vorbisenc->sinkpad, gst_vorbisenc_chain);
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gst_pad_set_connect_function (vorbisenc->sinkpad, gst_vorbisenc_sinkconnect);
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vorbisenc->srcpad = gst_pad_new_from_template (enc_src_template, "src");
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gst_element_add_pad (GST_ELEMENT (vorbisenc), vorbisenc->srcpad);
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vorbisenc->channels = 2;
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vorbisenc->frequency = 44100;
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vorbisenc->bitrate = 128000;
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vorbisenc->setup = FALSE;
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/* we're chained and we can deal with events */
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GST_FLAG_SET (vorbisenc, GST_ELEMENT_EVENT_AWARE);
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}
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static void
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gst_vorbisenc_setup (VorbisEnc * vorbisenc)
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{
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static const gchar *comment = "Track encoded with GStreamer";
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/********** Encode setup ************/
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/* choose an encoding mode */
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/* (mode 0: 44kHz stereo uncoupled, roughly 128kbps VBR) */
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vorbis_info_init (&vorbisenc->vi);
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vorbis_encode_init (&vorbisenc->vi, vorbisenc->channels, vorbisenc->frequency,
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-1, vorbisenc->bitrate, -1);
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gst_element_send_event (GST_ELEMENT (vorbisenc),
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gst_event_new_info ("channels", GST_PROPS_INT (vorbisenc->channels), NULL));
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gst_element_send_event (GST_ELEMENT (vorbisenc),
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gst_event_new_info ("rate", GST_PROPS_INT (vorbisenc->frequency), NULL));
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gst_element_send_event (GST_ELEMENT (vorbisenc),
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gst_event_new_info ("bitrate_nominal", GST_PROPS_INT (vorbisenc->bitrate), NULL));
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/* add a comment */
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vorbis_comment_init (&vorbisenc->vc);
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vorbis_comment_add (&vorbisenc->vc, (gchar *)comment);
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gst_element_send_event (GST_ELEMENT (vorbisenc),
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gst_event_new_info ("comment", GST_PROPS_STRING (comment), NULL));
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/* set up the analysis state and auxiliary encoding storage */
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vorbis_analysis_init (&vorbisenc->vd, &vorbisenc->vi);
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vorbis_block_init (&vorbisenc->vd, &vorbisenc->vb);
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/* set up our packet->stream encoder */
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/* pick a random serial number; that way we can more likely build
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chained streams just by concatenation */
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srand (time (NULL));
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ogg_stream_init (&vorbisenc->os, rand ());
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/* Vorbis streams begin with three headers; the initial header (with
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most of the codec setup parameters) which is mandated by the Ogg
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bitstream spec. The second header holds any comment fields. The
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third header holds the bitstream codebook. We merely need to
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make the headers, then pass them to libvorbis one at a time;
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libvorbis handles the additional Ogg bitstream constraints */
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{
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ogg_packet header;
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ogg_packet header_comm;
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ogg_packet header_code;
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vorbis_analysis_headerout (&vorbisenc->vd, &vorbisenc->vc, &header, &header_comm, &header_code);
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ogg_stream_packetin (&vorbisenc->os, &header); /* automatically placed in its own
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page */
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ogg_stream_packetin (&vorbisenc->os, &header_comm);
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ogg_stream_packetin (&vorbisenc->os, &header_code);
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/* no need to write out here. We'll get to that in the main loop */
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}
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vorbisenc->setup = TRUE;
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}
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static void
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gst_vorbisenc_chain (GstPad * pad, GstBuffer * buf)
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{
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VorbisEnc *vorbisenc;
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g_return_if_fail (pad != NULL);
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g_return_if_fail (GST_IS_PAD (pad));
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g_return_if_fail (buf != NULL);
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vorbisenc = GST_VORBISENC (gst_pad_get_parent (pad));
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if (!vorbisenc->setup) {
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gst_element_error (GST_ELEMENT (vorbisenc), "encoder not initialized (input is not audio?)");
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if (GST_IS_BUFFER (buf))
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gst_buffer_unref (buf);
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else
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gst_pad_event_default (pad, GST_EVENT (buf));
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return;
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}
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if (GST_IS_EVENT (buf)) {
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switch (GST_EVENT_TYPE (buf)) {
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case GST_EVENT_EOS:
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/* end of file. this can be done implicitly in the mainline,
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but it's easier to see here in non-clever fashion.
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Tell the library we're at end of stream so that it can handle
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the last frame and mark end of stream in the output properly */
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vorbis_analysis_wrote (&vorbisenc->vd, 0);
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default:
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gst_pad_event_default (pad, GST_EVENT (buf));
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break;
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}
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}
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else {
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gint16 *data;
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gulong size;
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gulong i, j;
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float **buffer;
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/* data to encode */
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data = (gint16 *) GST_BUFFER_DATA (buf);
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size = GST_BUFFER_SIZE (buf) / 2;
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/* expose the buffer to submit data */
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buffer = vorbis_analysis_buffer (&vorbisenc->vd, size / vorbisenc->channels);
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/* uninterleave samples */
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for (i = 0; i < size / vorbisenc->channels; i++) {
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for (j = 0; j < vorbisenc->channels; j++)
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buffer[j][i] = data[i * vorbisenc->channels + j] / 32768.f;
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}
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/* tell the library how much we actually submitted */
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vorbis_analysis_wrote (&vorbisenc->vd, size / vorbisenc->channels);
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gst_buffer_unref (buf);
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}
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/* vorbis does some data preanalysis, then divvies up blocks for
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more involved (potentially parallel) processing. Get a single
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block for encoding now */
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while (vorbis_analysis_blockout (&vorbisenc->vd, &vorbisenc->vb) == 1) {
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/* analysis */
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vorbis_analysis (&vorbisenc->vb, &vorbisenc->op);
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/* weld the packet into the bitstream */
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ogg_stream_packetin (&vorbisenc->os, &vorbisenc->op);
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/* write out pages (if any) */
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while (!vorbisenc->eos) {
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int result = ogg_stream_pageout (&vorbisenc->os, &vorbisenc->og);
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GstBuffer *outbuf;
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if (result == 0)
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break;
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outbuf = gst_buffer_new ();
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GST_BUFFER_DATA (outbuf) = g_malloc (vorbisenc->og.header_len + vorbisenc->og.body_len);
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GST_BUFFER_SIZE (outbuf) = vorbisenc->og.header_len + vorbisenc->og.body_len;
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memcpy (GST_BUFFER_DATA (outbuf), vorbisenc->og.header, vorbisenc->og.header_len);
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memcpy (GST_BUFFER_DATA (outbuf) + vorbisenc->og.header_len, vorbisenc->og.body,
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vorbisenc->og.body_len);
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GST_DEBUG (0, "vorbisenc: encoded buffer of %d bytes\n", GST_BUFFER_SIZE (outbuf));
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gst_pad_push (vorbisenc->srcpad, outbuf);
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/* this could be set above, but for illustrative purposes, I do
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it here (to show that vorbis does know where the stream ends) */
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if (ogg_page_eos (&vorbisenc->og)) {
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vorbisenc->eos = 1;
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}
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}
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}
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if (vorbisenc->eos) {
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/* clean up and exit. vorbis_info_clear() must be called last */
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ogg_stream_clear (&vorbisenc->os);
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vorbis_block_clear (&vorbisenc->vb);
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vorbis_dsp_clear (&vorbisenc->vd);
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vorbis_info_clear (&vorbisenc->vi);
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}
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}
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static void
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gst_vorbisenc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec)
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{
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VorbisEnc *vorbisenc;
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/* it's not null if we got it, but it might not be ours */
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g_return_if_fail (GST_IS_VORBISENC (object));
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vorbisenc = GST_VORBISENC (object);
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switch (prop_id) {
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case ARG_BITRATE:
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g_value_set_int (value, vorbisenc->bitrate);
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break;
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default:
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break;
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}
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}
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static void
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gst_vorbisenc_set_property (GObject * object, guint prop_id, const GValue * value,
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GParamSpec * pspec)
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{
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VorbisEnc *vorbisenc;
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/* it's not null if we got it, but it might not be ours */
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g_return_if_fail (GST_IS_VORBISENC (object));
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vorbisenc = GST_VORBISENC (object);
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switch (prop_id) {
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case ARG_BITRATE:
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vorbisenc->bitrate = g_value_get_int (value);
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break;
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default:
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break;
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}
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}
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