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webrtc lib: Make the datachannel 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 16:24:00 -04:00 committed by GStreamer Marge Bot
parent dd2da6f2b4
commit a931e31141
4 changed files with 66 additions and 63 deletions
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2
ext/webrtc/webrtcdatachannel.h
View file

@ -26,6 +26,8 @@
#include <gst/webrtc/datachannel.h>
#include "sctptransport.h"
#include "gst/webrtc/webrtc-priv.h"
G_BEGIN_DECLS
GType webrtc_data_channel_get_type(void);

1
gst-libs/gst/webrtc/datachannel.c
View file

@ -33,6 +33,7 @@
#endif
#include "datachannel.h"
#include "webrtc-priv.h"
#define GST_CAT_DEFAULT gst_webrtc_data_channel_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);

63
gst-libs/gst/webrtc/datachannel.h
View file

@ -36,69 +36,6 @@ GType gst_webrtc_data_channel_get_type(void);
#define GST_IS_WEBRTC_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DATA_CHANNEL))
#define GST_WEBRTC_DATA_CHANNEL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannelClass))
#define GST_WEBRTC_DATA_CHANNEL_LOCK(channel) g_mutex_lock(&((GstWebRTCDataChannel *)(channel))->lock)
#define GST_WEBRTC_DATA_CHANNEL_UNLOCK(channel) g_mutex_unlock(&((GstWebRTCDataChannel *)(channel))->lock)
/**
* GstWebRTCDataChannel:
*
* Since: 1.18
*/
struct _GstWebRTCDataChannel
{
GObject parent;
GMutex lock;
gchar *label;
gboolean ordered;
guint max_packet_lifetime;
guint max_retransmits;
gchar *protocol;
gboolean negotiated;
gint id;
GstWebRTCPriorityType priority;
GstWebRTCDataChannelState ready_state;
guint64 buffered_amount;
guint64 buffered_amount_low_threshold;
gpointer _padding[GST_PADDING];
};
/**
* GstWebRTCDataChannelClass:
*
* Since: 1.18
*/
struct _GstWebRTCDataChannelClass
{
GObjectClass parent_class;
void (*send_data) (GstWebRTCDataChannel * channel, GBytes *data);
void (*send_string) (GstWebRTCDataChannel * channel, const gchar *str);
void (*close) (GstWebRTCDataChannel * channel);
gpointer _padding[GST_PADDING];
};
GST_WEBRTC_API
void gst_webrtc_data_channel_on_open (GstWebRTCDataChannel * channel);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_close (GstWebRTCDataChannel * channel);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_error (GstWebRTCDataChannel * channel, GError * error);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_message_data (GstWebRTCDataChannel * channel, GBytes * data);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_message_string (GstWebRTCDataChannel * channel, const gchar * str);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel);
GST_WEBRTC_API
void gst_webrtc_data_channel_send_data (GstWebRTCDataChannel * channel, GBytes * data);

63
gst-libs/gst/webrtc/webrtc-priv.h
View file

@ -231,6 +231,69 @@ GST_WEBRTC_API
void gst_webrtc_dtls_transport_set_transport (GstWebRTCDTLSTransport * transport,
GstWebRTCICETransport * ice);
#define GST_WEBRTC_DATA_CHANNEL_LOCK(channel) g_mutex_lock(&((GstWebRTCDataChannel *)(channel))->lock)
#define GST_WEBRTC_DATA_CHANNEL_UNLOCK(channel) g_mutex_unlock(&((GstWebRTCDataChannel *)(channel))->lock)
/**
* GstWebRTCDataChannel:
*
* Since: 1.18
*/
struct _GstWebRTCDataChannel
{
GObject parent;
GMutex lock;
gchar *label;
gboolean ordered;
guint max_packet_lifetime;
guint max_retransmits;
gchar *protocol;
gboolean negotiated;
gint id;
GstWebRTCPriorityType priority;
GstWebRTCDataChannelState ready_state;
guint64 buffered_amount;
guint64 buffered_amount_low_threshold;
gpointer _padding[GST_PADDING];
};
/**
* GstWebRTCDataChannelClass:
*
* Since: 1.18
*/
struct _GstWebRTCDataChannelClass
{
GObjectClass parent_class;
void (*send_data) (GstWebRTCDataChannel * channel, GBytes *data);
void (*send_string) (GstWebRTCDataChannel * channel, const gchar *str);
void (*close) (GstWebRTCDataChannel * channel);
gpointer _padding[GST_PADDING];
};
GST_WEBRTC_API
void gst_webrtc_data_channel_on_open (GstWebRTCDataChannel * channel);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_close (GstWebRTCDataChannel * channel);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_error (GstWebRTCDataChannel * channel, GError * error);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_message_data (GstWebRTCDataChannel * channel, GBytes * data);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_message_string (GstWebRTCDataChannel * channel, const gchar * str);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel);
G_END_DECLS