rtp: add rtphdrextrfc6464

Header Extension for Client-to-Mixer Audio Level Indication as
defined in RFC 6464.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/630>
This commit is contained in:
Guillaume Desmottes 2021-01-21 18:04:58 +01:00
parent 4b6c3c9a1b
commit 7b7e49de31
7 changed files with 681 additions and 1 deletions

View file

@ -17013,6 +17013,36 @@
},
"rank": "none"
},
"rtphdrextrfc6464": {
"RTP-Header-Extension-URI": "urn:ietf:params:rtp-hdrext:ssrc-audio-level",
"author": "Guillaume Desmottes <guillaume.desmottes@collabora.com>",
"description": "Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension",
"hierarchy": [
"GstRTPHeaderExtensionRfc6464",
"GstRTPHeaderExtension",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Network/Extension/RTPHeader",
"long-name": "Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension",
"properties": {
"vad": {
"blurb": "If the vad extension attribute is enabled or not",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": false
}
},
"rank": "marginal"
},
"rtphdrexttwcc": {
"RTP-Header-Extension-URI": "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
"author": "Matthew Waters <matthew@centricular.com>",

View file

@ -0,0 +1,324 @@
/* GStreamer
* Copyright (C) <2018> Havard Graff <havard.graff@gmail.com>
* Copyright (C) <2020-2021> Guillaume Desmottes <guillaume.desmottes@collabora.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more
*/
/**
* SECTION:element-rtphdrextrfc6464
* @title: rtphdrextrfc6464
* @short_description: Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension
*
* Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension.
* The extension should be automatically created by payloader and depayloaders,
* if their `auto-header-extension` property is enabled, if the extension
* is part of the RTP caps.
*
* Since: 1.20
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtphdrext-rfc6464.h"
#include <gst/audio/audio.h>
#define RFC6464_HDR_EXT_URI GST_RTP_HDREXT_BASE"ssrc-audio-level"
GST_DEBUG_CATEGORY_STATIC (rtphdrrfc6464_twcc_debug);
#define GST_CAT_DEFAULT (rtphdrrfc6464_twcc_debug)
#define DEFAULT_VAD TRUE
enum
{
PROP_0,
PROP_VAD,
};
struct _GstRTPHeaderExtensionRfc6464
{
GstRTPHeaderExtension parent;
gboolean vad;
};
G_DEFINE_TYPE_WITH_CODE (GstRTPHeaderExtensionRfc6464,
gst_rtp_header_extension_rfc6464, GST_TYPE_RTP_HEADER_EXTENSION,
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "rtphdrextrfc6464", 0,
"RTP RFC 6464 Header Extensions");
)
static void
gst_rtp_header_extension_rfc6464_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstRTPHeaderExtensionRfc6464 *self =
GST_RTP_HEADER_EXTENSION_RFC6464 (object);
switch (prop_id) {
case PROP_VAD:
g_value_set_boolean (value, self->vad);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstRTPHeaderExtensionFlags
gst_rtp_header_extension_rfc6464_get_supported_flags (GstRTPHeaderExtension *
ext)
{
return GST_RTP_HEADER_EXTENSION_ONE_BYTE | GST_RTP_HEADER_EXTENSION_TWO_BYTE;
}
static gsize
gst_rtp_header_extension_rfc6464_get_max_size (GstRTPHeaderExtension * ext,
const GstBuffer * input_meta)
{
return 2;
}
static void
set_vad (GstRTPHeaderExtension * ext, gboolean vad)
{
GstRTPHeaderExtensionRfc6464 *self = GST_RTP_HEADER_EXTENSION_RFC6464 (ext);
if (self->vad == vad)
return;
GST_DEBUG_OBJECT (ext, "vad: %d", vad);
self->vad = vad;
g_object_notify (G_OBJECT (self), "vad");
}
static gboolean
gst_rtp_header_extension_rfc6464_set_attributes_from_caps (GstRTPHeaderExtension
* ext, const GstCaps * caps)
{
gchar *field_name = gst_rtp_header_extension_get_sdp_caps_field_name (ext);
GstStructure *s = gst_caps_get_structure (caps, 0);
const gchar *ext_uri;
const GValue *arr;
if (!field_name)
return FALSE;
if ((ext_uri = gst_structure_get_string (s, field_name))) {
if (g_strcmp0 (ext_uri, gst_rtp_header_extension_get_uri (ext)) != 0) {
/* incompatible extension uri for this instance */
goto error;
}
set_vad (ext, DEFAULT_VAD);
} else if ((arr = gst_structure_get_value (s, field_name))
&& GST_VALUE_HOLDS_ARRAY (arr)
&& gst_value_array_get_size (arr) == 3) {
const GValue *val;
const gchar *vad_attr;
val = gst_value_array_get_value (arr, 1);
if (!G_VALUE_HOLDS_STRING (val))
goto error;
if (g_strcmp0 (g_value_get_string (val),
gst_rtp_header_extension_get_uri (ext)) != 0)
goto error;
val = gst_value_array_get_value (arr, 2);
if (!G_VALUE_HOLDS_STRING (val))
goto error;
vad_attr = g_value_get_string (val);
if (g_str_equal (vad_attr, "vad=on"))
set_vad (ext, TRUE);
else if (g_str_equal (vad_attr, "vad=off"))
set_vad (ext, FALSE);
else {
GST_WARNING_OBJECT (ext, "Invalid attribute: %s", vad_attr);
goto error;
}
} else {
/* unknown caps format */
goto error;
}
g_free (field_name);
return TRUE;
error:
g_free (field_name);
return FALSE;
}
static gboolean
gst_rtp_header_extension_rfc6464_set_caps_from_attributes (GstRTPHeaderExtension
* ext, GstCaps * caps)
{
GstRTPHeaderExtensionRfc6464 *self = GST_RTP_HEADER_EXTENSION_RFC6464 (ext);
gchar *field_name = gst_rtp_header_extension_get_sdp_caps_field_name (ext);
GstStructure *s = gst_caps_get_structure (caps, 0);
GValue arr = G_VALUE_INIT;
GValue val = G_VALUE_INIT;
if (!field_name)
return FALSE;
g_value_init (&arr, GST_TYPE_ARRAY);
g_value_init (&val, G_TYPE_STRING);
/* direction */
g_value_set_string (&val, "");
gst_value_array_append_value (&arr, &val);
/* uri */
g_value_set_string (&val, gst_rtp_header_extension_get_uri (ext));
gst_value_array_append_value (&arr, &val);
/* attributes */
if (self->vad)
g_value_set_string (&val, "vad=on");
else
g_value_set_string (&val, "vad=off");
gst_value_array_append_value (&arr, &val);
gst_structure_set_value (s, field_name, &arr);
GST_DEBUG_OBJECT (self, "%" GST_PTR_FORMAT, caps);
g_value_unset (&val);
g_value_unset (&arr);
g_free (field_name);
return TRUE;
}
static gsize
gst_rtp_header_extension_rfc6464_write (GstRTPHeaderExtension * ext,
const GstBuffer * input_meta, GstRTPHeaderExtensionFlags write_flags,
GstBuffer * output, guint8 * data, gsize size)
{
GstAudioLevelMeta *meta;
g_return_val_if_fail (size >=
gst_rtp_header_extension_rfc6464_get_max_size (ext, NULL), -1);
g_return_val_if_fail (write_flags &
gst_rtp_header_extension_rfc6464_get_supported_flags (ext), -1);
meta = gst_buffer_get_audio_level_meta ((GstBuffer *) input_meta);
if (!meta) {
GST_LOG_OBJECT (ext, "no meta");
return 0;
}
if (meta->level > 127) {
GST_WARNING_OBJECT (ext, "level from meta is higher than 127: %d",
meta->level);
return -1;
}
GST_LOG_OBJECT (ext, "writing ext (level: %d voice: %d)", meta->level,
meta->voice_activity);
if (write_flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE) {
*data = (meta->level & 0x7F) | (meta->voice_activity << 7);
return 1;
} else {
guint16 payload;
payload = ((meta->level & 0x7F) | (meta->voice_activity << 7)) << 8;
GST_WRITE_UINT16_LE (data, payload);
return 2;
}
}
static gboolean
gst_rtp_header_extension_rfc6464_read (GstRTPHeaderExtension * ext,
GstRTPHeaderExtensionFlags read_flags, const guint8 * data, gsize size,
GstBuffer * buffer)
{
guint8 val;
guint8 level;
gboolean voice_activity;
g_return_val_if_fail (read_flags &
gst_rtp_header_extension_rfc6464_get_supported_flags (ext), -1);
if (read_flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE) {
val = data[0];
} else {
val = *((guint16 *) data) >> 8;
}
level = val & 0x7F;
voice_activity = (val & 0x80) >> 7;
GST_LOG_OBJECT (ext, "reading ext (level: %d voice: %d)", level,
voice_activity);
gst_buffer_add_audio_level_meta (buffer, level, voice_activity);
return TRUE;
}
static void
gst_rtp_header_extension_rfc6464_class_init (GstRTPHeaderExtensionRfc6464Class *
klass)
{
GstRTPHeaderExtensionClass *rtp_hdr_class;
GstElementClass *gstelement_class;
GObjectClass *gobject_class;
rtp_hdr_class = GST_RTP_HEADER_EXTENSION_CLASS (klass);
gobject_class = (GObjectClass *) klass;
gstelement_class = GST_ELEMENT_CLASS (klass);
gobject_class->get_property = gst_rtp_header_extension_rfc6464_get_property;
/**
* rtphdrextrfc6464:vad:
*
* If the vad extension attribute is enabled or not, default to %FALSE.
*
* Since: 1.20
*/
g_object_class_install_property (gobject_class, PROP_VAD,
g_param_spec_boolean ("vad", "vad",
"If the vad extension attribute is enabled or not",
DEFAULT_VAD, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
rtp_hdr_class->get_supported_flags =
gst_rtp_header_extension_rfc6464_get_supported_flags;
rtp_hdr_class->get_max_size = gst_rtp_header_extension_rfc6464_get_max_size;
rtp_hdr_class->set_attributes_from_caps =
gst_rtp_header_extension_rfc6464_set_attributes_from_caps;
rtp_hdr_class->set_caps_from_attributes =
gst_rtp_header_extension_rfc6464_set_caps_from_attributes;
rtp_hdr_class->write = gst_rtp_header_extension_rfc6464_write;
rtp_hdr_class->read = gst_rtp_header_extension_rfc6464_read;
gst_element_class_set_static_metadata (gstelement_class,
"Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension",
GST_RTP_HDREXT_ELEMENT_CLASS,
"Client-to-Mixer Audio Level Indication (RFC6464) RTP Header Extension",
"Guillaume Desmottes <guillaume.desmottes@collabora.com>");
gst_rtp_header_extension_class_set_uri (rtp_hdr_class, RFC6464_HDR_EXT_URI);
}
static void
gst_rtp_header_extension_rfc6464_init (GstRTPHeaderExtensionRfc6464 * self)
{
GST_DEBUG_OBJECT (self, "creating element");
self->vad = DEFAULT_VAD;
}

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@ -0,0 +1,30 @@
/* GStreamer
* Copyright (C) <2018> Havard Graff <havard.graff@gmail.com>
* Copyright (C) <2020-2021> Guillaume Desmottes <guillaume.desmottes@collabora.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more
*/
#ifndef __GST_RTPHDREXT_RFC6464_H__
#define __GST_RTPHDREXT_RFC6464_H__
#include <gst/gst.h>
#include <gst/rtp/gstrtphdrext.h>
G_BEGIN_DECLS
#define GST_TYPE_RTP_HEADER_EXTENSION_RFC6464 (gst_rtp_header_extension_rfc6464_get_type())
G_DECLARE_FINAL_TYPE (GstRTPHeaderExtensionRfc6464, gst_rtp_header_extension_rfc6464, GST, RTP_HEADER_EXTENSION_RFC6464, GstRTPHeaderExtension)
G_END_DECLS
#endif /* __GST_RTPHDREXT_RFC6464_H__ */

View file

@ -35,6 +35,7 @@
#include "gstrtpst2022-1-fecdec.h"
#include "gstrtpst2022-1-fecenc.h"
#include "gstrtphdrext-twcc.h"
#include "gstrtphdrext-rfc6464.h"
static gboolean
plugin_init (GstPlugin * plugin)
@ -89,6 +90,10 @@ plugin_init (GstPlugin * plugin)
GST_TYPE_RTP_HEADER_EXTENSION_TWCC))
return FALSE;
if (!gst_element_register (plugin, "rtphdrextrfc6464", GST_RANK_MARGINAL,
GST_TYPE_RTP_HEADER_EXTENSION_RFC6464))
return FALSE;
return TRUE;
}

View file

@ -4,6 +4,7 @@ rtpmanager_sources = [
'gstrtpdtmfmux.c',
'gstrtpjitterbuffer.c',
'gstrtphdrext-twcc.c',
'gstrtphdrext-rfc6464.c',
'gstrtpmux.c',
'gstrtpptdemux.c',
'gstrtprtxqueue.c',
@ -26,7 +27,7 @@ gstrtpmanager = library('gstrtpmanager',
rtpmanager_sources,
c_args : gst_plugins_good_args,
include_directories : [configinc, libsinc],
dependencies : [gstbase_dep, gstnet_dep, gstrtp_dep, gio_dep],
dependencies : [gstbase_dep, gstnet_dep, gstrtp_dep, gstaudio_dep, gio_dep],
install : true,
install_dir : plugins_install_dir,
)

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@ -0,0 +1,289 @@
/* GStreamer
*
* unit test for RTP RFC 6464 Header Extensions
*
* Copyright (C) <2020-2021> Guillaume Desmottes <guillaume.desmottes@collabora.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include <gst/check/gstcheck.h>
#include <gst/rtp/rtp.h>
#include <gst/sdp/gstsdpmessage.h>
#include <gst/audio/audio.h>
#define URN "urn:ietf:params:rtp-hdrext:ssrc-audio-level"
#define SDP "v=0\r\n" \
"o=- 123456 2 IN IP4 127.0.0.1 \r\n" \
"s=-\r\n" \
"t=0 0\r\n" \
"a=maxptime:60\r\n" \
"a=sendrecv\r\n" \
"m=audio 55815 RTP/SAVPF 100\r\n" \
"c=IN IP4 1.1.1.1\r\n" \
"a=rtpmap:100 opus/48000/2\r\n"
#define SDP_NO_VAD SDP \
"a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\n"
#define SDP_VAD_ON SDP \
"a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=on\r\n"
#define SDP_VAD_OFF SDP \
"a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=off\r\n"
#define SDP_VAD_WRONG SDP \
"a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level vad=badger\r\n"
static GstCaps *
create_caps (const gchar * sdp)
{
GstSDPMessage *message;
glong length = -1;
const GstSDPMedia *media;
GstCaps *caps;
gst_sdp_message_new (&message);
gst_sdp_message_parse_buffer ((guint8 *) sdp, length, message);
media = gst_sdp_message_get_media (message, 0);
fail_unless (media != NULL);
caps = gst_sdp_media_get_caps_from_media (media, 100);
gst_sdp_media_attributes_to_caps (media, caps);
gst_sdp_message_free (message);
return caps;
}
static void
check_caps (GstRTPHeaderExtension * ext, gboolean vad)
{
GstCaps *caps;
GstStructure *s;
const GValue *arr, *val;
caps = gst_caps_new_empty_simple ("application/x-rtp");
fail_unless (gst_rtp_header_extension_set_caps_from_attributes (ext, caps));
s = gst_caps_get_structure (caps, 0);
arr = gst_structure_get_value (s, "extmap-1");
fail_unless (arr != NULL);
fail_unless (GST_VALUE_HOLDS_ARRAY (arr));
fail_unless (gst_value_array_get_size (arr) == 3);
val = gst_value_array_get_value (arr, 0);
fail_unless_equals_string (g_value_get_string (val), "");
val = gst_value_array_get_value (arr, 1);
fail_unless_equals_string (g_value_get_string (val), URN);
val = gst_value_array_get_value (arr, 2);
if (vad) {
fail_unless_equals_string (g_value_get_string (val), "vad=on");
} else {
fail_unless_equals_string (g_value_get_string (val), "vad=off");
}
gst_caps_unref (caps);
}
GST_START_TEST (rtprfc6464_sdp)
{
GstRTPHeaderExtension *ext;
GstCaps *caps;
gboolean vad = FALSE;
ext = gst_rtp_header_extension_create_from_uri (URN);
fail_unless (ext != NULL);
gst_rtp_header_extension_set_id (ext, 1);
/* vad default to on */
caps = create_caps (SDP_NO_VAD);
fail_unless (gst_rtp_header_extension_set_attributes_from_caps (ext, caps));
gst_caps_unref (caps);
g_object_get (ext, "vad", &vad, NULL);
fail_unless (vad);
check_caps (ext, TRUE);
/* vad is disabled */
caps = create_caps (SDP_VAD_OFF);
fail_unless (gst_rtp_header_extension_set_attributes_from_caps (ext, caps));
gst_caps_unref (caps);
g_object_get (ext, "vad", &vad, NULL);
fail_if (vad);
/* vad is enabled */
caps = create_caps (SDP_VAD_ON);
fail_unless (gst_rtp_header_extension_set_attributes_from_caps (ext, caps));
gst_caps_unref (caps);
g_object_get (ext, "vad", &vad, NULL);
fail_unless (vad);
/* invalid vad */
caps = create_caps (SDP_VAD_WRONG);
fail_if (gst_rtp_header_extension_set_attributes_from_caps (ext, caps));
gst_caps_unref (caps);
gst_object_unref (ext);
}
GST_END_TEST;
GST_START_TEST (rtprfc6464_one_byte)
{
GstRTPHeaderExtension *ext;
GstRTPHeaderExtensionFlags flags;
GstBuffer *buffer;
guint8 *data;
gsize size, written;
GstAudioLevelMeta *meta;
guint8 level = 12;
gboolean voice = TRUE;
ext = gst_rtp_header_extension_create_from_uri (URN);
fail_unless (ext != NULL);
gst_rtp_header_extension_set_id (ext, 1);
flags = gst_rtp_header_extension_get_supported_flags (ext);
fail_unless (flags & GST_RTP_HEADER_EXTENSION_ONE_BYTE);
buffer = gst_buffer_new ();
meta = gst_buffer_add_audio_level_meta (buffer, level, voice);
size = gst_rtp_header_extension_get_max_size (ext, buffer);
fail_unless (size > 0);
data = g_malloc0 (size);
fail_unless (data != NULL);
/* Write extension */
written =
gst_rtp_header_extension_write (ext, buffer,
GST_RTP_HEADER_EXTENSION_ONE_BYTE, buffer, data, size);
fail_unless (written == 1);
/* Read it back */
fail_unless (gst_buffer_remove_meta (buffer, (GstMeta *) meta));
fail_unless (gst_rtp_header_extension_read (ext,
GST_RTP_HEADER_EXTENSION_ONE_BYTE, data, size, buffer));
meta = gst_buffer_get_audio_level_meta (buffer);
fail_unless (meta != NULL);
fail_unless_equals_int (meta->level, level);
fail_unless (meta->voice_activity == voice);
g_free (data);
gst_buffer_unref (buffer);
gst_object_unref (ext);
}
GST_END_TEST;
GST_START_TEST (rtprfc6464_two_bytes)
{
GstRTPHeaderExtension *ext;
GstRTPHeaderExtensionFlags flags;
GstBuffer *buffer;
guint8 *data;
gsize size, written;
GstAudioLevelMeta *meta;
guint8 level = 12;
gboolean voice = TRUE;
ext = gst_rtp_header_extension_create_from_uri (URN);
fail_unless (ext != NULL);
gst_rtp_header_extension_set_id (ext, 1);
flags = gst_rtp_header_extension_get_supported_flags (ext);
fail_unless (flags & GST_RTP_HEADER_EXTENSION_TWO_BYTE);
buffer = gst_buffer_new ();
meta = gst_buffer_add_audio_level_meta (buffer, level, voice);
size = gst_rtp_header_extension_get_max_size (ext, buffer);
fail_unless (size > 0);
data = g_malloc0 (size);
fail_unless (data != NULL);
/* Write extension */
written =
gst_rtp_header_extension_write (ext, buffer,
GST_RTP_HEADER_EXTENSION_TWO_BYTE, buffer, data, size);
fail_unless (written == 2);
/* Read it back */
fail_unless (gst_buffer_remove_meta (buffer, (GstMeta *) meta));
fail_unless (gst_rtp_header_extension_read (ext,
GST_RTP_HEADER_EXTENSION_TWO_BYTE, data, size, buffer));
meta = gst_buffer_get_audio_level_meta (buffer);
fail_unless (meta != NULL);
fail_unless_equals_int (meta->level, level);
fail_unless (meta->voice_activity == voice);
g_free (data);
gst_buffer_unref (buffer);
gst_object_unref (ext);
}
GST_END_TEST;
GST_START_TEST (rtprfc6464_no_meta)
{
GstRTPHeaderExtension *ext;
GstBuffer *buffer;
guint8 *data;
gsize size, written;
ext = gst_rtp_header_extension_create_from_uri (URN);
fail_unless (ext != NULL);
gst_rtp_header_extension_set_id (ext, 1);
buffer = gst_buffer_new ();
size = gst_rtp_header_extension_get_max_size (ext, buffer);
fail_unless (size > 0);
data = g_malloc0 (size);
fail_unless (data != NULL);
written =
gst_rtp_header_extension_write (ext, buffer,
GST_RTP_HEADER_EXTENSION_ONE_BYTE, buffer, data, size);
fail_unless (written == 0);
written =
gst_rtp_header_extension_write (ext, buffer,
GST_RTP_HEADER_EXTENSION_TWO_BYTE, buffer, data, size);
fail_unless (written == 0);
g_free (data);
gst_buffer_unref (buffer);
gst_object_unref (ext);
}
GST_END_TEST;
static Suite *
rtprfc6464_suite (void)
{
Suite *s = suite_create ("rtprfc6464");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, rtprfc6464_sdp);
tcase_add_test (tc_chain, rtprfc6464_one_byte);
tcase_add_test (tc_chain, rtprfc6464_two_bytes);
tcase_add_test (tc_chain, rtprfc6464_no_meta);
return s;
}
GST_CHECK_MAIN (rtprfc6464)

View file

@ -72,6 +72,7 @@ good_tests = [
[ 'elements/rtpbin_buffer_list' ],
[ 'elements/rtpcollision' ],
[ 'elements/rtpfunnel' ],
[ 'elements/rtphdrextrfc6464', false, [gstsdp_dep, gstaudio_dep] ],
[ 'elements/rtpjitterbuffer' ],
[ 'elements/rtpjpeg' ],