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3b2a0751f9
Original commit message from CVS: * ext/vorbis/vorbisenc.c: * gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_new_from_id3v1): * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add), (gst_tag_to_vorbis_comments): Fix handling of GST_TAG_DATE, which is now of GST_TYPE_DATE.
1164 lines
34 KiB
C
1164 lines
34 KiB
C
/* GStreamer
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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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/**
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* SECTION:element-vorbisenc
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* @see_also: vorbisdec, oggmux
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*
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* <refsect2>
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* <para>
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* This element encodes raw float audio into a Vorbis stream.
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* <ulink url="http://www.vorbis.com/">Vorbis</ulink> is a royalty-free
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* audio codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org
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* Foundation</ulink>.
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* </para>
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* <title>Example pipelines</title>
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* <para>
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* Encode a test sine signal to Ogg/Vorbis. Note that the resulting file
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* will be really small because a sine signal compresses very well.
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* </para>
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* <programlisting>
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* gst-launch -v sinesrc num-buffers=100 ! audioconvert ! vorbisenc ! oggmux ! filesink location=sinesrc.ogg
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* </programlisting>
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* <para>
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* Record from a sound card using ALSA and encode to Ogg/Vorbis.
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* </para>
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* <programlisting>
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* gst-launch -v alsasrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
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* </programlisting>
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* </refsect2>
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*/
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/**
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* SECTION:vorbisenc
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* @short_description: an encoder that encodes to Ogg/Vorbis
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* @see_also: oggdemux
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <stdlib.h>
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#include <string.h>
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#include <time.h>
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#include <vorbis/vorbisenc.h>
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#include <gst/gsttagsetter.h>
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#include <gst/tag/tag.h>
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#include "vorbisenc.h"
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GST_DEBUG_CATEGORY_EXTERN (vorbisenc_debug);
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#define GST_CAT_DEFAULT vorbisenc_debug
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static GstPadTemplate *gst_vorbisenc_src_template, *gst_vorbisenc_sink_template;
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/* elementfactory information */
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GstElementDetails vorbisenc_details = {
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"Vorbis encoder",
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"Codec/Encoder/Audio",
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"Encodes audio in Vorbis format",
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"Monty <monty@xiph.org>, " "Wim Taymans <wim@fluendo.com>",
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};
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/* GstVorbisEnc 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_MAX_BITRATE,
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ARG_BITRATE,
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ARG_MIN_BITRATE,
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ARG_QUALITY,
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ARG_MANAGED,
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ARG_LAST_MESSAGE
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};
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static void gst_vorbisenc_output_buffers (GstVorbisEnc * vorbisenc);
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/* FIXME:
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* vorbis_granule_time was added between 1.0 and 1.0.1; it's too silly
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* to require a new version for such a simple function, but once we move
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* beyond 1.0 for other reasons we can remove this copy */
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static double
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vorbis_granule_time_copy (vorbis_dsp_state * v, ogg_int64_t granulepos)
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{
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if (granulepos >= 0)
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return ((double) granulepos / v->vi->rate);
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return (-1);
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}
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#if 0
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static const GstFormat *
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gst_vorbisenc_get_formats (GstPad * pad)
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{
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static const GstFormat src_formats[] = {
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GST_FORMAT_BYTES,
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GST_FORMAT_TIME,
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0
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};
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static const GstFormat sink_formats[] = {
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GST_FORMAT_BYTES,
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GST_FORMAT_DEFAULT,
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GST_FORMAT_TIME,
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0
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};
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return (GST_PAD_IS_SRC (pad) ? src_formats : sink_formats);
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}
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#endif
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#define MAX_BITRATE_DEFAULT -1
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#define BITRATE_DEFAULT -1
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#define MIN_BITRATE_DEFAULT -1
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#define QUALITY_DEFAULT 0.3
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#define LOWEST_BITRATE 6000 /* lowest allowed for a 8 kHz stream */
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#define HIGHEST_BITRATE 250001 /* highest allowed for a 44 kHz stream */
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static void gst_vorbisenc_base_init (gpointer g_class);
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static void gst_vorbisenc_class_init (GstVorbisEncClass * klass);
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static void gst_vorbisenc_init (GstVorbisEnc * vorbisenc);
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static gboolean gst_vorbisenc_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_vorbisenc_chain (GstPad * pad, GstBuffer * buffer);
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static gboolean gst_vorbisenc_setup (GstVorbisEnc * vorbisenc);
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static void gst_vorbisenc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_vorbisenc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn gst_vorbisenc_change_state (GstElement * element,
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GstStateChange transition);
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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 (GstVorbisEncClass),
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gst_vorbisenc_base_init,
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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 (GstVorbisEnc),
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0,
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(GInstanceInitFunc) gst_vorbisenc_init,
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};
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static const GInterfaceInfo tag_setter_info = {
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NULL,
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NULL,
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NULL
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};
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vorbisenc_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstVorbisEnc",
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&vorbisenc_info, 0);
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g_type_add_interface_static (vorbisenc_type, GST_TYPE_TAG_SETTER,
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&tag_setter_info);
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}
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return vorbisenc_type;
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}
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static GstCaps *
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vorbis_caps_factory (void)
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{
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return gst_caps_new_simple ("audio/x-vorbis", NULL);
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}
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static GstCaps *
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raw_caps_factory (void)
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{
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/* lowest sample rate is in vorbis/lib/modes/setup_8.h, 8000 Hz
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* highest sample rate is in vorbis/lib/modes/setup_44.h, 50000 Hz */
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return
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gst_caps_new_simple ("audio/x-raw-float",
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"rate", GST_TYPE_INT_RANGE, 8000, 50000,
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"channels", GST_TYPE_INT_RANGE, 1, 2,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"width", G_TYPE_INT, 32, "buffer-frames", G_TYPE_INT, 0, NULL);
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}
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static void
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gst_vorbisenc_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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GstCaps *raw_caps, *vorbis_caps;
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raw_caps = raw_caps_factory ();
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vorbis_caps = vorbis_caps_factory ();
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gst_vorbisenc_sink_template = gst_pad_template_new ("sink", GST_PAD_SINK,
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GST_PAD_ALWAYS, raw_caps);
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gst_vorbisenc_src_template = gst_pad_template_new ("src", GST_PAD_SRC,
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GST_PAD_ALWAYS, vorbis_caps);
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gst_element_class_add_pad_template (element_class,
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gst_vorbisenc_sink_template);
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gst_element_class_add_pad_template (element_class,
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gst_vorbisenc_src_template);
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gst_element_class_set_details (element_class, &vorbisenc_details);
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}
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static void
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gst_vorbisenc_class_init (GstVorbisEncClass * 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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gobject_class->set_property = gst_vorbisenc_set_property;
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gobject_class->get_property = gst_vorbisenc_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MAX_BITRATE,
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g_param_spec_int ("max-bitrate", "Maximum Bitrate",
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"Specify a maximum bitrate (in bps). Useful for streaming "
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"applications. (-1 == disabled)",
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-1, HIGHEST_BITRATE, MAX_BITRATE_DEFAULT, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BITRATE,
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g_param_spec_int ("bitrate", "Target Bitrate",
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"Attempt to encode at a bitrate averaging this (in bps). "
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"This uses the bitrate management engine, and is not recommended for most users. "
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"Quality is a better alternative. (-1 == disabled)",
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-1, HIGHEST_BITRATE, BITRATE_DEFAULT, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MIN_BITRATE,
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g_param_spec_int ("min_bitrate", "Minimum Bitrate",
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"Specify a minimum bitrate (in bps). Useful for encoding for a "
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"fixed-size channel. (-1 == disabled)",
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-1, HIGHEST_BITRATE, MIN_BITRATE_DEFAULT, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_QUALITY,
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g_param_spec_float ("quality", "Quality",
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"Specify quality instead of specifying a particular bitrate.",
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-0.1, 1.0, QUALITY_DEFAULT, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MANAGED,
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g_param_spec_boolean ("managed", "Managed",
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"Enable bitrate management engine", FALSE, G_PARAM_READWRITE));
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_LAST_MESSAGE,
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g_param_spec_string ("last-message", "last-message",
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"The last status message", NULL, G_PARAM_READABLE));
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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gstelement_class->change_state = gst_vorbisenc_change_state;
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}
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static gboolean
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gst_vorbisenc_sink_setcaps (GstPad * pad, GstCaps * caps)
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{
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GstVorbisEnc *vorbisenc;
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GstStructure *structure;
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vorbisenc = GST_VORBISENC (GST_PAD_PARENT (pad));
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vorbisenc->setup = FALSE;
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (structure, "channels", &vorbisenc->channels);
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gst_structure_get_int (structure, "rate", &vorbisenc->frequency);
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gst_vorbisenc_setup (vorbisenc);
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if (vorbisenc->setup)
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return TRUE;
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return FALSE;
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}
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static gboolean
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gst_vorbisenc_convert_src (GstPad * pad, GstFormat src_format, gint64 src_value,
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GstFormat * dest_format, gint64 * dest_value)
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{
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gboolean res = TRUE;
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GstVorbisEnc *vorbisenc;
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gint64 avg;
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vorbisenc = GST_VORBISENC (gst_pad_get_parent (pad));
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if (vorbisenc->samples_in == 0 ||
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vorbisenc->bytes_out == 0 || vorbisenc->frequency == 0)
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return FALSE;
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avg = (vorbisenc->bytes_out * vorbisenc->frequency) / (vorbisenc->samples_in);
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switch (src_format) {
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case GST_FORMAT_BYTES:
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switch (*dest_format) {
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case GST_FORMAT_TIME:
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*dest_value = src_value * GST_SECOND / avg;
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break;
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default:
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res = FALSE;
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}
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break;
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case GST_FORMAT_TIME:
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switch (*dest_format) {
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case GST_FORMAT_BYTES:
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*dest_value = src_value * avg / GST_SECOND;
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break;
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default:
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res = FALSE;
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}
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break;
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default:
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res = FALSE;
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}
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return res;
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}
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static gboolean
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gst_vorbisenc_convert_sink (GstPad * pad, GstFormat src_format,
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gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
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{
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gboolean res = TRUE;
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guint scale = 1;
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gint bytes_per_sample;
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GstVorbisEnc *vorbisenc;
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vorbisenc = GST_VORBISENC (gst_pad_get_parent (pad));
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bytes_per_sample = vorbisenc->channels * 2;
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switch (src_format) {
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case GST_FORMAT_BYTES:
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switch (*dest_format) {
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case GST_FORMAT_DEFAULT:
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if (bytes_per_sample == 0)
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return FALSE;
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*dest_value = src_value / bytes_per_sample;
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break;
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case GST_FORMAT_TIME:
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{
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gint byterate = bytes_per_sample * vorbisenc->frequency;
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if (byterate == 0)
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return FALSE;
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*dest_value = src_value * GST_SECOND / byterate;
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break;
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}
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default:
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res = FALSE;
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}
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break;
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case GST_FORMAT_DEFAULT:
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switch (*dest_format) {
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case GST_FORMAT_BYTES:
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*dest_value = src_value * bytes_per_sample;
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break;
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case GST_FORMAT_TIME:
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if (vorbisenc->frequency == 0)
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return FALSE;
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*dest_value = src_value * GST_SECOND / vorbisenc->frequency;
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break;
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default:
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res = FALSE;
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}
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break;
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case GST_FORMAT_TIME:
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switch (*dest_format) {
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case GST_FORMAT_BYTES:
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scale = bytes_per_sample;
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/* fallthrough */
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case GST_FORMAT_DEFAULT:
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*dest_value = src_value * scale * vorbisenc->frequency / GST_SECOND;
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break;
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default:
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res = FALSE;
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}
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break;
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default:
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res = FALSE;
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}
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return res;
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}
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static const GstQueryType *
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gst_vorbisenc_get_query_types (GstPad * pad)
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{
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static const GstQueryType gst_vorbisenc_src_query_types[] = {
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GST_QUERY_POSITION,
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0
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};
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return gst_vorbisenc_src_query_types;
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}
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static gboolean
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gst_vorbisenc_src_query (GstPad * pad, GstQuery * query)
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{
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gboolean res = TRUE;
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GstVorbisEnc *vorbisenc;
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vorbisenc = GST_VORBISENC (GST_PAD_PARENT (pad));
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_POSITION:
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{
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#if 0
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switch (*format) {
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case GST_FORMAT_BYTES:
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case GST_FORMAT_TIME:
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{
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gint64 peer_value;
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const GstFormat *peer_formats;
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res = FALSE;
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peer_formats =
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gst_pad_get_formats (GST_PAD_PEER (vorbisenc->sinkpad));
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while (peer_formats && *peer_formats && !res) {
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GstFormat peer_format = *peer_formats;
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/* do the probe */
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if (gst_pad_query (GST_PAD_PEER (vorbisenc->sinkpad),
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GST_QUERY_TOTAL, &peer_format, &peer_value)) {
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GstFormat conv_format;
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/* convert to TIME */
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conv_format = GST_FORMAT_TIME;
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res = gst_vorbisenc_convert_sink (vorbisenc->sinkpad,
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peer_format, peer_value, &conv_format, value);
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/* and to final format */
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res &= gst_vorbisenc_convert_src (pad,
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GST_FORMAT_TIME, *value, format, value);
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}
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peer_formats++;
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}
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break;
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}
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default:
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res = FALSE;
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break;
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}
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#endif
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res = FALSE;
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break;
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}
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case GST_QUERY_CONVERT:
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{
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GstFormat src_fmt, dest_fmt;
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gint64 src_val, dest_val;
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gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
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if (!(res =
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gst_vorbisenc_convert_src (pad, src_fmt, src_val, &dest_fmt,
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&dest_val)))
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goto error;
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gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
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break;
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}
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default:
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res = FALSE;
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break;
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}
|
|
|
|
error:
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_vorbisenc_sink_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
gboolean res = TRUE;
|
|
GstVorbisEnc *vorbisenc;
|
|
|
|
vorbisenc = GST_VORBISENC (GST_PAD_PARENT (pad));
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_CONVERT:
|
|
{
|
|
GstFormat src_fmt, dest_fmt;
|
|
gint64 src_val, dest_val;
|
|
|
|
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
|
|
if (!(res =
|
|
gst_vorbisenc_convert_sink (pad, src_fmt, src_val, &dest_fmt,
|
|
&dest_val)))
|
|
goto error;
|
|
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
|
|
break;
|
|
}
|
|
default:
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
|
|
error:
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_init (GstVorbisEnc * vorbisenc)
|
|
{
|
|
vorbisenc->sinkpad =
|
|
gst_pad_new_from_template (gst_vorbisenc_sink_template, "sink");
|
|
gst_element_add_pad (GST_ELEMENT (vorbisenc), vorbisenc->sinkpad);
|
|
gst_pad_set_event_function (vorbisenc->sinkpad, gst_vorbisenc_sink_event);
|
|
gst_pad_set_chain_function (vorbisenc->sinkpad, gst_vorbisenc_chain);
|
|
gst_pad_set_setcaps_function (vorbisenc->sinkpad, gst_vorbisenc_sink_setcaps);
|
|
gst_pad_set_query_function (vorbisenc->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_vorbisenc_sink_query));
|
|
|
|
vorbisenc->srcpad =
|
|
gst_pad_new_from_template (gst_vorbisenc_src_template, "src");
|
|
gst_pad_set_query_function (vorbisenc->srcpad,
|
|
GST_DEBUG_FUNCPTR (gst_vorbisenc_src_query));
|
|
gst_pad_set_query_type_function (vorbisenc->srcpad,
|
|
GST_DEBUG_FUNCPTR (gst_vorbisenc_get_query_types));
|
|
gst_element_add_pad (GST_ELEMENT (vorbisenc), vorbisenc->srcpad);
|
|
|
|
vorbisenc->channels = -1;
|
|
vorbisenc->frequency = -1;
|
|
|
|
vorbisenc->managed = FALSE;
|
|
vorbisenc->max_bitrate = MAX_BITRATE_DEFAULT;
|
|
vorbisenc->bitrate = BITRATE_DEFAULT;
|
|
vorbisenc->min_bitrate = MIN_BITRATE_DEFAULT;
|
|
vorbisenc->quality = QUALITY_DEFAULT;
|
|
vorbisenc->quality_set = FALSE;
|
|
vorbisenc->last_message = NULL;
|
|
|
|
vorbisenc->setup = FALSE;
|
|
vorbisenc->header_sent = FALSE;
|
|
}
|
|
|
|
|
|
static gchar *
|
|
gst_vorbisenc_get_tag_value (const GstTagList * list, const gchar * tag,
|
|
int index)
|
|
{
|
|
GType tag_type;
|
|
gchar *vorbisvalue = NULL;
|
|
|
|
if (tag == NULL)
|
|
return NULL;
|
|
|
|
tag_type = gst_tag_get_type (tag);
|
|
|
|
/* get tag name right */
|
|
if ((strcmp (tag, GST_TAG_TRACK_NUMBER) == 0)
|
|
|| (strcmp (tag, GST_TAG_ALBUM_VOLUME_NUMBER) == 0)
|
|
|| (strcmp (tag, GST_TAG_TRACK_COUNT) == 0)
|
|
|| (strcmp (tag, GST_TAG_ALBUM_VOLUME_COUNT) == 0)) {
|
|
guint track_no;
|
|
|
|
if (!gst_tag_list_get_uint_index (list, tag, index, &track_no))
|
|
g_return_val_if_reached (NULL);
|
|
|
|
vorbisvalue = g_strdup_printf ("%u", track_no);
|
|
} else if (tag_type == GST_TYPE_DATE) {
|
|
GDate *date;
|
|
|
|
if (!gst_tag_list_get_date_index (list, tag, index, &date))
|
|
g_return_val_if_reached (NULL);
|
|
|
|
vorbisvalue =
|
|
g_strdup_printf ("%04d-%02d-%02d", (gint) g_date_get_year (date),
|
|
(gint) g_date_get_month (date), (gint) g_date_get_day (date));
|
|
g_date_free (date);
|
|
} else if (tag_type == G_TYPE_STRING) {
|
|
if (!gst_tag_list_get_string_index (list, tag, index, &vorbisvalue))
|
|
g_return_val_if_reached (NULL);
|
|
}
|
|
|
|
return vorbisvalue;
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_metadata_set1 (const GstTagList * list, const gchar * tag,
|
|
gpointer vorbisenc)
|
|
{
|
|
const gchar *vorbistag = NULL;
|
|
gchar *vorbisvalue = NULL;
|
|
guint i, count;
|
|
GstVorbisEnc *enc = GST_VORBISENC (vorbisenc);
|
|
|
|
vorbistag = gst_tag_to_vorbis_tag (tag);
|
|
if (vorbistag == NULL) {
|
|
return;
|
|
}
|
|
|
|
count = gst_tag_list_get_tag_size (list, tag);
|
|
for (i = 0; i < count; i++) {
|
|
vorbisvalue = gst_vorbisenc_get_tag_value (list, tag, i);
|
|
|
|
if (vorbisvalue != NULL) {
|
|
vorbis_comment_add_tag (&enc->vc, g_strdup (vorbistag), vorbisvalue);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_set_metadata (GstVorbisEnc * vorbisenc)
|
|
{
|
|
GstTagList *copy;
|
|
const GstTagList *user_tags;
|
|
|
|
user_tags = gst_tag_setter_get_list (GST_TAG_SETTER (vorbisenc));
|
|
if (!(vorbisenc->tags || user_tags))
|
|
return;
|
|
|
|
copy =
|
|
gst_tag_list_merge (user_tags, vorbisenc->tags,
|
|
gst_tag_setter_get_merge_mode (GST_TAG_SETTER (vorbisenc)));
|
|
vorbis_comment_init (&vorbisenc->vc);
|
|
gst_tag_list_foreach (copy, gst_vorbisenc_metadata_set1, vorbisenc);
|
|
gst_tag_list_free (copy);
|
|
}
|
|
|
|
static gchar *
|
|
get_constraints_string (GstVorbisEnc * vorbisenc)
|
|
{
|
|
gint min = vorbisenc->min_bitrate;
|
|
gint max = vorbisenc->max_bitrate;
|
|
gchar *result;
|
|
|
|
if (min > 0 && max > 0)
|
|
result = g_strdup_printf ("(min %d bps, max %d bps)", min, max);
|
|
else if (min > 0)
|
|
result = g_strdup_printf ("(min %d bps, no max)", min);
|
|
else if (max > 0)
|
|
result = g_strdup_printf ("(no min, max %d bps)", max);
|
|
else
|
|
result = g_strdup_printf ("(no min or max)");
|
|
|
|
return result;
|
|
}
|
|
|
|
static void
|
|
update_start_message (GstVorbisEnc * vorbisenc)
|
|
{
|
|
gchar *constraints;
|
|
|
|
g_free (vorbisenc->last_message);
|
|
|
|
if (vorbisenc->bitrate > 0) {
|
|
if (vorbisenc->managed) {
|
|
constraints = get_constraints_string (vorbisenc);
|
|
vorbisenc->last_message =
|
|
g_strdup_printf ("encoding at average bitrate %d bps %s",
|
|
vorbisenc->bitrate, constraints);
|
|
g_free (constraints);
|
|
} else {
|
|
vorbisenc->last_message =
|
|
g_strdup_printf
|
|
("encoding at approximate bitrate %d bps (VBR encoding enabled)",
|
|
vorbisenc->bitrate);
|
|
}
|
|
} else {
|
|
if (vorbisenc->quality_set) {
|
|
if (vorbisenc->managed) {
|
|
constraints = get_constraints_string (vorbisenc);
|
|
vorbisenc->last_message =
|
|
g_strdup_printf
|
|
("encoding at quality level %2.2f using constrained VBR %s",
|
|
vorbisenc->quality, constraints);
|
|
g_free (constraints);
|
|
} else {
|
|
vorbisenc->last_message =
|
|
g_strdup_printf ("encoding at quality level %2.2f",
|
|
vorbisenc->quality);
|
|
}
|
|
} else {
|
|
constraints = get_constraints_string (vorbisenc);
|
|
vorbisenc->last_message =
|
|
g_strdup_printf ("encoding using bitrate management %s", constraints);
|
|
g_free (constraints);
|
|
}
|
|
}
|
|
|
|
g_object_notify (G_OBJECT (vorbisenc), "last_message");
|
|
}
|
|
|
|
static gboolean
|
|
gst_vorbisenc_setup (GstVorbisEnc * vorbisenc)
|
|
{
|
|
vorbisenc->setup = FALSE;
|
|
|
|
if (vorbisenc->bitrate < 0 && vorbisenc->min_bitrate < 0
|
|
&& vorbisenc->max_bitrate < 0) {
|
|
vorbisenc->quality_set = TRUE;
|
|
}
|
|
|
|
update_start_message (vorbisenc);
|
|
|
|
/* choose an encoding mode */
|
|
/* (mode 0: 44kHz stereo uncoupled, roughly 128kbps VBR) */
|
|
vorbis_info_init (&vorbisenc->vi);
|
|
|
|
if (vorbisenc->quality_set) {
|
|
if (vorbis_encode_setup_vbr (&vorbisenc->vi,
|
|
vorbisenc->channels, vorbisenc->frequency,
|
|
vorbisenc->quality) != 0) {
|
|
GST_ERROR_OBJECT (vorbisenc,
|
|
"vorbisenc: initialisation failed: invalid parameters for quality");
|
|
vorbis_info_clear (&vorbisenc->vi);
|
|
return FALSE;
|
|
}
|
|
|
|
/* do we have optional hard quality restrictions? */
|
|
if (vorbisenc->max_bitrate > 0 || vorbisenc->min_bitrate > 0) {
|
|
struct ovectl_ratemanage_arg ai;
|
|
|
|
vorbis_encode_ctl (&vorbisenc->vi, OV_ECTL_RATEMANAGE_GET, &ai);
|
|
|
|
/* the bitrates used by libvorbisenc are in kbit/sec, ours in bit/sec
|
|
* also remember that in telecom kbit/sec is 1000 bit/sec */
|
|
ai.bitrate_hard_min = vorbisenc->min_bitrate / 1000;
|
|
ai.bitrate_hard_max = vorbisenc->max_bitrate / 1000;
|
|
ai.management_active = 1;
|
|
|
|
vorbis_encode_ctl (&vorbisenc->vi, OV_ECTL_RATEMANAGE_SET, &ai);
|
|
}
|
|
} else {
|
|
long min_bitrate, max_bitrate;
|
|
|
|
min_bitrate = vorbisenc->min_bitrate > 0 ? vorbisenc->min_bitrate : -1;
|
|
max_bitrate = vorbisenc->max_bitrate > 0 ? vorbisenc->max_bitrate : -1;
|
|
|
|
if (vorbis_encode_setup_managed (&vorbisenc->vi,
|
|
vorbisenc->channels,
|
|
vorbisenc->frequency,
|
|
max_bitrate, vorbisenc->bitrate, min_bitrate) != 0) {
|
|
GST_ERROR_OBJECT (vorbisenc,
|
|
"vorbis_encode_setup_managed "
|
|
"(c %d, rate %d, max br %ld, br %ld, min br %ld) failed",
|
|
vorbisenc->channels, vorbisenc->frequency, max_bitrate,
|
|
vorbisenc->bitrate, min_bitrate);
|
|
vorbis_info_clear (&vorbisenc->vi);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
if (vorbisenc->managed && vorbisenc->bitrate < 0) {
|
|
vorbis_encode_ctl (&vorbisenc->vi, OV_ECTL_RATEMANAGE_AVG, NULL);
|
|
} else if (!vorbisenc->managed) {
|
|
/* Turn off management entirely (if it was turned on). */
|
|
vorbis_encode_ctl (&vorbisenc->vi, OV_ECTL_RATEMANAGE_SET, NULL);
|
|
}
|
|
vorbis_encode_setup_init (&vorbisenc->vi);
|
|
|
|
/* set up the analysis state and auxiliary encoding storage */
|
|
vorbis_analysis_init (&vorbisenc->vd, &vorbisenc->vi);
|
|
vorbis_block_init (&vorbisenc->vd, &vorbisenc->vb);
|
|
|
|
vorbisenc->setup = TRUE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_clear (GstVorbisEnc * vorbisenc)
|
|
{
|
|
if (vorbisenc->setup) {
|
|
vorbis_analysis_wrote (&vorbisenc->vd, 0);
|
|
gst_vorbisenc_output_buffers (vorbisenc);
|
|
|
|
vorbisenc->setup = FALSE;
|
|
}
|
|
|
|
/* clean up and exit. vorbis_info_clear() must be called last */
|
|
vorbis_block_clear (&vorbisenc->vb);
|
|
vorbis_dsp_clear (&vorbisenc->vd);
|
|
vorbis_info_clear (&vorbisenc->vi);
|
|
|
|
vorbisenc->header_sent = FALSE;
|
|
}
|
|
|
|
/* prepare a buffer for transmission by passing data through libvorbis */
|
|
static GstBuffer *
|
|
gst_vorbisenc_buffer_from_packet (GstVorbisEnc * vorbisenc, ogg_packet * packet)
|
|
{
|
|
GstBuffer *outbuf;
|
|
|
|
outbuf = gst_buffer_new_and_alloc (packet->bytes);
|
|
memcpy (GST_BUFFER_DATA (outbuf), packet->packet, packet->bytes);
|
|
GST_BUFFER_OFFSET (outbuf) = vorbisenc->bytes_out;
|
|
GST_BUFFER_OFFSET_END (outbuf) = packet->granulepos;
|
|
GST_BUFFER_TIMESTAMP (outbuf) =
|
|
vorbis_granule_time_copy (&vorbisenc->vd,
|
|
packet->granulepos) * GST_SECOND;
|
|
|
|
GST_DEBUG ("encoded buffer of %d bytes", GST_BUFFER_SIZE (outbuf));
|
|
return outbuf;
|
|
}
|
|
|
|
/* push out the buffer and do internal bookkeeping */
|
|
static void
|
|
gst_vorbisenc_push_buffer (GstVorbisEnc * vorbisenc, GstBuffer * buffer)
|
|
{
|
|
vorbisenc->bytes_out += GST_BUFFER_SIZE (buffer);
|
|
|
|
if (GST_PAD_IS_USABLE (vorbisenc->srcpad)) {
|
|
gst_pad_push (vorbisenc->srcpad, buffer);
|
|
} else {
|
|
gst_buffer_unref (buffer);
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_push_packet (GstVorbisEnc * vorbisenc, ogg_packet * packet)
|
|
{
|
|
GstBuffer *outbuf;
|
|
|
|
outbuf = gst_vorbisenc_buffer_from_packet (vorbisenc, packet);
|
|
gst_vorbisenc_push_buffer (vorbisenc, outbuf);
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_vorbisenc_set_header_on_caps (GstCaps * caps, GstBuffer * buf1,
|
|
GstBuffer * buf2, GstBuffer * buf3)
|
|
{
|
|
GstStructure *structure;
|
|
GValue list = { 0 };
|
|
GValue value = { 0 };
|
|
|
|
caps = gst_caps_make_writable (caps);
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
|
|
/* mark buffers */
|
|
GST_BUFFER_FLAG_SET (buf1, GST_BUFFER_FLAG_IN_CAPS);
|
|
GST_BUFFER_FLAG_SET (buf2, GST_BUFFER_FLAG_IN_CAPS);
|
|
GST_BUFFER_FLAG_SET (buf3, GST_BUFFER_FLAG_IN_CAPS);
|
|
|
|
/* put buffers in a fixed list */
|
|
g_value_init (&list, GST_TYPE_ARRAY);
|
|
g_value_init (&value, GST_TYPE_BUFFER);
|
|
gst_value_set_buffer (&value, buf1);
|
|
gst_value_list_append_value (&list, &value);
|
|
g_value_unset (&value);
|
|
g_value_init (&value, GST_TYPE_BUFFER);
|
|
gst_value_set_buffer (&value, buf2);
|
|
gst_value_list_append_value (&list, &value);
|
|
g_value_unset (&value);
|
|
g_value_init (&value, GST_TYPE_BUFFER);
|
|
gst_value_set_buffer (&value, buf3);
|
|
gst_value_list_append_value (&list, &value);
|
|
gst_structure_set_value (structure, "streamheader", &list);
|
|
g_value_unset (&value);
|
|
g_value_unset (&list);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static gboolean
|
|
gst_vorbisenc_sink_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
gboolean res = TRUE;
|
|
GstVorbisEnc *vorbisenc;
|
|
|
|
vorbisenc = GST_VORBISENC (GST_PAD_PARENT (pad));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_EOS:
|
|
/* 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 */
|
|
GST_DEBUG_OBJECT (vorbisenc, "EOS, clearing state and sending event on");
|
|
gst_vorbisenc_clear (vorbisenc);
|
|
|
|
res = gst_pad_event_default (pad, event);
|
|
break;
|
|
case GST_EVENT_TAG:
|
|
if (vorbisenc->tags) {
|
|
GstTagList *list;
|
|
|
|
gst_event_parse_tag (event, &list);
|
|
gst_tag_list_insert (vorbisenc->tags, list,
|
|
gst_tag_setter_get_merge_mode (GST_TAG_SETTER (vorbisenc)));
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
res = gst_pad_event_default (pad, event);
|
|
break;
|
|
default:
|
|
res = gst_pad_event_default (pad, event);
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_vorbisenc_chain (GstPad * pad, GstBuffer * buffer)
|
|
{
|
|
GstBuffer *buf = GST_BUFFER (buffer);
|
|
GstVorbisEnc *vorbisenc;
|
|
|
|
vorbisenc = GST_VORBISENC (GST_PAD_PARENT (pad));
|
|
|
|
{
|
|
gfloat *data;
|
|
gulong size;
|
|
gulong i, j;
|
|
float **buffer;
|
|
|
|
if (!vorbisenc->setup) {
|
|
gst_buffer_unref (buf);
|
|
GST_ELEMENT_ERROR (vorbisenc, CORE, NEGOTIATION, (NULL),
|
|
("encoder not initialized (input is not audio?)"));
|
|
return GST_FLOW_UNEXPECTED;
|
|
}
|
|
|
|
if (!vorbisenc->header_sent) {
|
|
/* 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;
|
|
GstBuffer *buf1, *buf2, *buf3;
|
|
GstCaps *caps;
|
|
|
|
GST_DEBUG_OBJECT (vorbisenc, "creating and sending header packets");
|
|
gst_vorbisenc_set_metadata (vorbisenc);
|
|
vorbis_analysis_headerout (&vorbisenc->vd, &vorbisenc->vc, &header,
|
|
&header_comm, &header_code);
|
|
|
|
/* create header buffers */
|
|
buf1 = gst_vorbisenc_buffer_from_packet (vorbisenc, &header);
|
|
buf2 = gst_vorbisenc_buffer_from_packet (vorbisenc, &header_comm);
|
|
buf3 = gst_vorbisenc_buffer_from_packet (vorbisenc, &header_code);
|
|
|
|
/* mark and put on caps */
|
|
caps = gst_pad_get_caps (vorbisenc->srcpad);
|
|
caps = gst_vorbisenc_set_header_on_caps (caps, buf1, buf2, buf3);
|
|
|
|
/* negotiate with these caps */
|
|
GST_DEBUG ("here are the caps: %" GST_PTR_FORMAT, caps);
|
|
gst_pad_set_caps (vorbisenc->srcpad, caps);
|
|
|
|
gst_buffer_set_caps (buf1, caps);
|
|
gst_buffer_set_caps (buf2, caps);
|
|
gst_buffer_set_caps (buf3, caps);
|
|
|
|
/* push out buffers */
|
|
gst_vorbisenc_push_buffer (vorbisenc, buf1);
|
|
gst_vorbisenc_push_buffer (vorbisenc, buf2);
|
|
gst_vorbisenc_push_buffer (vorbisenc, buf3);
|
|
|
|
vorbisenc->header_sent = TRUE;
|
|
}
|
|
|
|
/* data to encode */
|
|
data = (gfloat *) GST_BUFFER_DATA (buf);
|
|
size = GST_BUFFER_SIZE (buf) / (vorbisenc->channels * sizeof (float));
|
|
|
|
/* expose the buffer to submit data */
|
|
buffer = vorbis_analysis_buffer (&vorbisenc->vd, size);
|
|
|
|
/* uninterleave samples */
|
|
for (i = 0; i < size; i++) {
|
|
for (j = 0; j < vorbisenc->channels; j++) {
|
|
buffer[j][i] = *data++;
|
|
}
|
|
}
|
|
|
|
/* tell the library how much we actually submitted */
|
|
vorbis_analysis_wrote (&vorbisenc->vd, size);
|
|
|
|
vorbisenc->samples_in += size;
|
|
|
|
gst_buffer_unref (buf);
|
|
}
|
|
|
|
gst_vorbisenc_output_buffers (vorbisenc);
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_output_buffers (GstVorbisEnc * vorbisenc)
|
|
{
|
|
/* vorbis does some data preanalysis, then divides up blocks for
|
|
more involved (potentially parallel) processing. Get a single
|
|
block for encoding now */
|
|
while (vorbis_analysis_blockout (&vorbisenc->vd, &vorbisenc->vb) == 1) {
|
|
ogg_packet op;
|
|
|
|
GST_LOG_OBJECT (vorbisenc, "analysed to a block");
|
|
|
|
/* analysis */
|
|
vorbis_analysis (&vorbisenc->vb, NULL);
|
|
vorbis_bitrate_addblock (&vorbisenc->vb);
|
|
|
|
while (vorbis_bitrate_flushpacket (&vorbisenc->vd, &op)) {
|
|
GST_LOG_OBJECT (vorbisenc, "pushing out a data packet");
|
|
gst_vorbisenc_push_packet (vorbisenc, &op);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstVorbisEnc *vorbisenc;
|
|
|
|
g_return_if_fail (GST_IS_VORBISENC (object));
|
|
|
|
vorbisenc = GST_VORBISENC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_MAX_BITRATE:
|
|
g_value_set_int (value, vorbisenc->max_bitrate);
|
|
break;
|
|
case ARG_BITRATE:
|
|
g_value_set_int (value, vorbisenc->bitrate);
|
|
break;
|
|
case ARG_MIN_BITRATE:
|
|
g_value_set_int (value, vorbisenc->min_bitrate);
|
|
break;
|
|
case ARG_QUALITY:
|
|
g_value_set_float (value, vorbisenc->quality);
|
|
break;
|
|
case ARG_MANAGED:
|
|
g_value_set_boolean (value, vorbisenc->managed);
|
|
break;
|
|
case ARG_LAST_MESSAGE:
|
|
g_value_set_string (value, vorbisenc->last_message);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_vorbisenc_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstVorbisEnc *vorbisenc;
|
|
|
|
g_return_if_fail (GST_IS_VORBISENC (object));
|
|
|
|
vorbisenc = GST_VORBISENC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_MAX_BITRATE:
|
|
{
|
|
gboolean old_value = vorbisenc->managed;
|
|
|
|
vorbisenc->max_bitrate = g_value_get_int (value);
|
|
if (vorbisenc->max_bitrate >= 0
|
|
&& vorbisenc->max_bitrate < LOWEST_BITRATE) {
|
|
g_warning ("Lowest allowed bitrate is %d", LOWEST_BITRATE);
|
|
vorbisenc->max_bitrate = LOWEST_BITRATE;
|
|
}
|
|
if (vorbisenc->min_bitrate > 0 && vorbisenc->max_bitrate > 0)
|
|
vorbisenc->managed = TRUE;
|
|
else
|
|
vorbisenc->managed = FALSE;
|
|
|
|
if (old_value != vorbisenc->managed)
|
|
g_object_notify (object, "managed");
|
|
break;
|
|
}
|
|
case ARG_BITRATE:
|
|
vorbisenc->bitrate = g_value_get_int (value);
|
|
if (vorbisenc->bitrate >= 0 && vorbisenc->bitrate < LOWEST_BITRATE) {
|
|
g_warning ("Lowest allowed bitrate is %d", LOWEST_BITRATE);
|
|
vorbisenc->bitrate = LOWEST_BITRATE;
|
|
}
|
|
break;
|
|
case ARG_MIN_BITRATE:
|
|
{
|
|
gboolean old_value = vorbisenc->managed;
|
|
|
|
vorbisenc->min_bitrate = g_value_get_int (value);
|
|
if (vorbisenc->min_bitrate >= 0
|
|
&& vorbisenc->min_bitrate < LOWEST_BITRATE) {
|
|
g_warning ("Lowest allowed bitrate is %d", LOWEST_BITRATE);
|
|
vorbisenc->min_bitrate = LOWEST_BITRATE;
|
|
}
|
|
if (vorbisenc->min_bitrate > 0 && vorbisenc->max_bitrate > 0)
|
|
vorbisenc->managed = TRUE;
|
|
else
|
|
vorbisenc->managed = FALSE;
|
|
|
|
if (old_value != vorbisenc->managed)
|
|
g_object_notify (object, "managed");
|
|
break;
|
|
}
|
|
case ARG_QUALITY:
|
|
vorbisenc->quality = g_value_get_float (value);
|
|
if (vorbisenc->quality >= 0.0)
|
|
vorbisenc->quality_set = TRUE;
|
|
else
|
|
vorbisenc->quality_set = FALSE;
|
|
break;
|
|
case ARG_MANAGED:
|
|
vorbisenc->managed = g_value_get_boolean (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_vorbisenc_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstVorbisEnc *vorbisenc = GST_VORBISENC (element);
|
|
GstStateChangeReturn res;
|
|
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
vorbisenc->tags = gst_tag_list_new ();
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
default:
|
|
break;
|
|
}
|
|
|
|
res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_vorbisenc_clear (vorbisenc);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
gst_tag_list_free (vorbisenc->tags);
|
|
vorbisenc->tags = NULL;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|