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webrtc lib: Make the transceiver struct private

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This will prevent any unsafe access.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2241>
This commit is contained in:
Olivier Crête 2021-04-21 16:00:34 -04:00 committed by GStreamer Marge Bot
parent 08323f382c
commit b5f2de3124
4 changed files with 94 additions and 61 deletions
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1
ext/webrtc/webrtctransceiver.h
View file

@ -22,6 +22,7 @@
#include "fwd.h"
#include <gst/webrtc/rtptransceiver.h>
#include "gst/webrtc/webrtc-priv.h"
#include "transportstream.h"
G_BEGIN_DECLS

2
gst-libs/gst/webrtc/rtptransceiver.c
View file

@ -32,6 +32,8 @@
#include "rtptransceiver.h"
#include "webrtc-priv.h"
#define GST_CAT_DEFAULT gst_webrtc_rtp_transceiver_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);

61
gst-libs/gst/webrtc/rtptransceiver.h
View file

@ -22,8 +22,6 @@
#include <gst/gst.h>
#include <gst/webrtc/webrtc_fwd.h>
#include <gst/webrtc/rtpsender.h>
#include <gst/webrtc/rtpreceiver.h>
G_BEGIN_DECLS
@ -36,65 +34,6 @@ GType gst_webrtc_rtp_transceiver_get_type(void);
#define GST_IS_WEBRTC_RTP_TRANSCEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_TRANSCEIVER))
#define GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_TRANSCEIVER,GstWebRTCRTPTransceiverClass))
/**
* GstWebRTCRTPTransceiver:
* @mline: the mline number this transceiver corresponds to
* @mid: The media ID of the m-line associated with this
* transceiver. This association is established, when possible,
* whenever either a local or remote description is applied. This
* field is NULL if neither a local or remote description has been
* applied, or if its associated m-line is rejected by either a remote
* offer or any answer.
* @stopped: Indicates whether or not sending and receiving using the paired
* #GstWebRTCRTPSender and #GstWebRTCRTPReceiver has been permanently disabled,
* either due to SDP offer/answer
* @sender: The #GstWebRTCRTPSender object responsible sending data to the
* remote peer
* @receiver: The #GstWebRTCRTPReceiver object responsible for receiver data from
* the remote peer.
* @direction: The transceiver's desired direction.
* @current_direction: The transceiver's current direction (read-only)
* @codec_preferences: A caps representing the codec preferences (read-only)
* @kind: Type of media (Since: 1.20)
*
* Mostly matches the WebRTC RTCRtpTransceiver interface.
*
* Since: 1.16
*/
/**
* GstWebRTCRTPTransceiver.kind:
*
* Type of media
*
* Since: 1.20
*/
struct _GstWebRTCRTPTransceiver
{
GstObject parent;
guint mline;
gchar *mid;
gboolean stopped;
GstWebRTCRTPSender *sender;
GstWebRTCRTPReceiver *receiver;
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiverDirection current_direction;
GstCaps *codec_preferences;
GstWebRTCKind kind;
gpointer _padding[GST_PADDING];
};
struct _GstWebRTCRTPTransceiverClass
{
GstObjectClass parent_class;
/* FIXME; reset */
gpointer _padding[GST_PADDING];
};
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPTransceiver, gst_object_unref)
G_END_DECLS

91
gst-libs/gst/webrtc/webrtc-priv.h Normal file
View file

@ -0,0 +1,91 @@
/* GStreamer
* Copyright (C) 2017 Matthew Waters <matthew@centricular.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.
*/
#ifndef __GST_WEBRTC_PRIV_H__
#define __GST_WEBRTC_PRIV_H__
#include <gst/gst.h>
#include <gst/webrtc/webrtc_fwd.h>
#include <gst/webrtc/rtpsender.h>
#include <gst/webrtc/rtpreceiver.h>
G_BEGIN_DECLS
/**
* GstWebRTCRTPTransceiver:
* @mline: the mline number this transceiver corresponds to
* @mid: The media ID of the m-line associated with this
* transceiver. This association is established, when possible,
* whenever either a local or remote description is applied. This
* field is NULL if neither a local or remote description has been
* applied, or if its associated m-line is rejected by either a remote
* offer or any answer.
* @stopped: Indicates whether or not sending and receiving using the paired
* #GstWebRTCRTPSender and #GstWebRTCRTPReceiver has been permanently disabled,
* either due to SDP offer/answer
* @sender: The #GstWebRTCRTPSender object responsible sending data to the
* remote peer
* @receiver: The #GstWebRTCRTPReceiver object responsible for receiver data from
* the remote peer.
* @direction: The transceiver's desired direction.
* @current_direction: The transceiver's current direction (read-only)
* @codec_preferences: A caps representing the codec preferences (read-only)
* @kind: Type of media (Since: 1.20)
*
* Mostly matches the WebRTC RTCRtpTransceiver interface.
*
* Since: 1.16
*/
/**
* GstWebRTCRTPTransceiver.kind:
*
* Type of media
*
* Since: 1.20
*/
struct _GstWebRTCRTPTransceiver
{
GstObject parent;
guint mline;
gchar *mid;
gboolean stopped;
GstWebRTCRTPSender *sender;
GstWebRTCRTPReceiver *receiver;
GstWebRTCRTPTransceiverDirection direction;
GstWebRTCRTPTransceiverDirection current_direction;
GstCaps *codec_preferences;
GstWebRTCKind kind;
gpointer _padding[GST_PADDING];
};
struct _GstWebRTCRTPTransceiverClass
{
GstObjectClass parent_class;
/* FIXME; reset */
gpointer _padding[GST_PADDING];
};
G_END_DECLS
#endif /* __GST_WEBRTC_PRIV_H__ */