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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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5ec82c1c4e
Since the base class now does the parsing, there is no need to reproduce that code in all the subclasses, just pass the attributes which are the only relevant bit anyway. Also, only store the direction if the subclass accepted the caps Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/906>
258 lines
7.6 KiB
C
258 lines
7.6 KiB
C
/* GStreamer
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* Copyright (C) <2018> Havard Graff <havard.graff@gmail.com>
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* Copyright (C) <2020-2021> Guillaume Desmottes <guillaume.desmottes@collabora.com>
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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
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*/
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/**
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* SECTION:element-rtphdrextrfc6464
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* @title: rtphdrextrfc6464
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* @short_description: Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension
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*
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* Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension.
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* The extension should be automatically created by payloader and depayloaders,
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* if their `auto-header-extension` property is enabled, if the extension
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* is part of the RTP caps.
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*
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* ## Example pipeline
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* |[
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* gst-launch-1.0 pulsesrc ! level audio-level-meta=true ! audiconvert !
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* rtpL16pay ! application/x-rtp,
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* extmap-1=(string)\< \"\", urn:ietf:params:rtp-hdrext:ssrc-audio-level,
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* \"vad=on\" \> ! udpsink
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* ]|
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*
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* Since: 1.20
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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 "gstrtphdrext-rfc6464.h"
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#include <gst/audio/audio.h>
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#define RFC6464_HDR_EXT_URI GST_RTP_HDREXT_BASE"ssrc-audio-level"
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GST_DEBUG_CATEGORY_STATIC (rtphdrrfc6464_twcc_debug);
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#define GST_CAT_DEFAULT (rtphdrrfc6464_twcc_debug)
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#define DEFAULT_VAD TRUE
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enum
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{
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PROP_0,
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PROP_VAD,
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};
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struct _GstRTPHeaderExtensionRfc6464
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{
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GstRTPHeaderExtension parent;
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gboolean vad;
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};
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G_DEFINE_TYPE_WITH_CODE (GstRTPHeaderExtensionRfc6464,
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gst_rtp_header_extension_rfc6464, GST_TYPE_RTP_HEADER_EXTENSION,
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GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "rtphdrextrfc6464", 0,
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"RTP RFC 6464 Header Extensions"););
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GST_ELEMENT_REGISTER_DEFINE (rtphdrextrfc6464, "rtphdrextrfc6464",
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GST_RANK_MARGINAL, GST_TYPE_RTP_HEADER_EXTENSION_RFC6464);
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static void
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gst_rtp_header_extension_rfc6464_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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GstRTPHeaderExtensionRfc6464 *self =
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GST_RTP_HEADER_EXTENSION_RFC6464 (object);
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switch (prop_id) {
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case PROP_VAD:
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g_value_set_boolean (value, self->vad);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstRTPHeaderExtensionFlags
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gst_rtp_header_extension_rfc6464_get_supported_flags (GstRTPHeaderExtension *
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ext)
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{
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return GST_RTP_HEADER_EXTENSION_ONE_BYTE | GST_RTP_HEADER_EXTENSION_TWO_BYTE;
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}
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static gsize
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gst_rtp_header_extension_rfc6464_get_max_size (GstRTPHeaderExtension * ext,
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const GstBuffer * input_meta)
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{
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return 2;
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}
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static void
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set_vad (GstRTPHeaderExtension * ext, gboolean vad)
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{
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GstRTPHeaderExtensionRfc6464 *self = GST_RTP_HEADER_EXTENSION_RFC6464 (ext);
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if (self->vad == vad)
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return;
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GST_DEBUG_OBJECT (ext, "vad: %d", vad);
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self->vad = vad;
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g_object_notify (G_OBJECT (self), "vad");
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}
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static gboolean
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gst_rtp_header_extension_rfc6464_set_attributes (GstRTPHeaderExtension * ext,
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GstRTPHeaderExtensionDirection direction, const gchar * attributes)
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{
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if (g_str_equal (attributes, "vad=on") || g_str_equal (attributes, "")) {
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set_vad (ext, TRUE);
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} else if (g_str_equal (attributes, "vad=off")) {
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set_vad (ext, FALSE);
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} else {
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GST_WARNING_OBJECT (ext, "Invalid attribute: %s", attributes);
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return FALSE;
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}
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return TRUE;
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}
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static gboolean
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gst_rtp_header_extension_rfc6464_set_caps_from_attributes (GstRTPHeaderExtension
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* ext, GstCaps * caps)
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{
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GstRTPHeaderExtensionRfc6464 *self = GST_RTP_HEADER_EXTENSION_RFC6464 (ext);
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const gchar *vad;
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if (self->vad)
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vad = "vad=on";
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else
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vad = "vad=off";
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return gst_rtp_header_extension_set_caps_from_attributes_helper (ext, caps,
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vad);
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}
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static gssize
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gst_rtp_header_extension_rfc6464_write (GstRTPHeaderExtension * ext,
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const GstBuffer * input_meta, GstRTPHeaderExtensionFlags write_flags,
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GstBuffer * output, guint8 * data, gsize size)
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{
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GstAudioLevelMeta *meta;
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guint level;
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g_return_val_if_fail (size >=
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gst_rtp_header_extension_rfc6464_get_max_size (ext, NULL), -1);
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g_return_val_if_fail (write_flags &
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gst_rtp_header_extension_rfc6464_get_supported_flags (ext), -1);
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meta = gst_buffer_get_audio_level_meta ((GstBuffer *) input_meta);
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if (!meta) {
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GST_LOG_OBJECT (ext, "no meta");
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return 0;
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}
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level = meta->level;
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if (level > 127) {
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GST_LOG_OBJECT (ext, "level from meta is higher than 127: %d, cropping",
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meta->level);
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level = 127;
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}
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GST_LOG_OBJECT (ext, "writing ext (level: %d voice: %d)", meta->level,
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meta->voice_activity);
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/* Both one & two byte use the same format, the second byte being padding */
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data[0] = (meta->level & 0x7F) | (meta->voice_activity << 7);
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if (write_flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE) {
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return 1;
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}
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data[1] = 0;
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return 2;
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}
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static gboolean
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gst_rtp_header_extension_rfc6464_read (GstRTPHeaderExtension * ext,
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GstRTPHeaderExtensionFlags read_flags, const guint8 * data, gsize size,
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GstBuffer * buffer)
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{
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guint8 level;
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gboolean voice_activity;
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g_return_val_if_fail (read_flags &
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gst_rtp_header_extension_rfc6464_get_supported_flags (ext), -1);
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/* Both one & two byte use the same format, the second byte being padding */
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level = data[0] & 0x7F;
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voice_activity = (data[0] & 0x80) >> 7;
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GST_LOG_OBJECT (ext, "reading ext (level: %d voice: %d)", level,
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voice_activity);
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gst_buffer_add_audio_level_meta (buffer, level, voice_activity);
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return TRUE;
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}
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static void
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gst_rtp_header_extension_rfc6464_class_init (GstRTPHeaderExtensionRfc6464Class *
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klass)
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{
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GstRTPHeaderExtensionClass *rtp_hdr_class;
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GstElementClass *gstelement_class;
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GObjectClass *gobject_class;
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rtp_hdr_class = GST_RTP_HEADER_EXTENSION_CLASS (klass);
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gobject_class = (GObjectClass *) klass;
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gstelement_class = GST_ELEMENT_CLASS (klass);
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gobject_class->get_property = gst_rtp_header_extension_rfc6464_get_property;
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/**
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* rtphdrextrfc6464:vad:
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*
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* If the vad extension attribute is enabled or not, default to %FALSE.
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*
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* Since: 1.20
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*/
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g_object_class_install_property (gobject_class, PROP_VAD,
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g_param_spec_boolean ("vad", "vad",
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"If the vad extension attribute is enabled or not",
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DEFAULT_VAD, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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rtp_hdr_class->get_supported_flags =
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gst_rtp_header_extension_rfc6464_get_supported_flags;
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rtp_hdr_class->get_max_size = gst_rtp_header_extension_rfc6464_get_max_size;
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rtp_hdr_class->set_attributes =
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gst_rtp_header_extension_rfc6464_set_attributes;
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rtp_hdr_class->set_caps_from_attributes =
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gst_rtp_header_extension_rfc6464_set_caps_from_attributes;
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rtp_hdr_class->write = gst_rtp_header_extension_rfc6464_write;
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rtp_hdr_class->read = gst_rtp_header_extension_rfc6464_read;
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gst_element_class_set_static_metadata (gstelement_class,
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"Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension",
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GST_RTP_HDREXT_ELEMENT_CLASS,
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"Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension",
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"Guillaume Desmottes <guillaume.desmottes@collabora.com>");
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gst_rtp_header_extension_class_set_uri (rtp_hdr_class, RFC6464_HDR_EXT_URI);
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}
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static void
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gst_rtp_header_extension_rfc6464_init (GstRTPHeaderExtensionRfc6464 * self)
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{
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GST_DEBUG_OBJECT (self, "creating element");
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self->vad = DEFAULT_VAD;
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}
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