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webrtc lib: Make the datachannel struct private
This will prevent any unsafe access. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2241>
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dd2da6f2b4
commit
a931e31141
4 changed files with 66 additions and 63 deletions
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@ -26,6 +26,8 @@
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#include <gst/webrtc/datachannel.h>
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#include "sctptransport.h"
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#include "gst/webrtc/webrtc-priv.h"
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G_BEGIN_DECLS
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GType webrtc_data_channel_get_type(void);
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@ -33,6 +33,7 @@
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#endif
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#include "datachannel.h"
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#include "webrtc-priv.h"
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#define GST_CAT_DEFAULT gst_webrtc_data_channel_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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@ -36,69 +36,6 @@ GType gst_webrtc_data_channel_get_type(void);
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#define GST_IS_WEBRTC_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DATA_CHANNEL))
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#define GST_WEBRTC_DATA_CHANNEL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannelClass))
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#define GST_WEBRTC_DATA_CHANNEL_LOCK(channel) g_mutex_lock(&((GstWebRTCDataChannel *)(channel))->lock)
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#define GST_WEBRTC_DATA_CHANNEL_UNLOCK(channel) g_mutex_unlock(&((GstWebRTCDataChannel *)(channel))->lock)
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/**
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* GstWebRTCDataChannel:
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*
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* Since: 1.18
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*/
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struct _GstWebRTCDataChannel
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{
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GObject parent;
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GMutex lock;
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gchar *label;
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gboolean ordered;
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guint max_packet_lifetime;
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guint max_retransmits;
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gchar *protocol;
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gboolean negotiated;
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gint id;
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GstWebRTCPriorityType priority;
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GstWebRTCDataChannelState ready_state;
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guint64 buffered_amount;
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guint64 buffered_amount_low_threshold;
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gpointer _padding[GST_PADDING];
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};
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/**
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* GstWebRTCDataChannelClass:
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*
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* Since: 1.18
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*/
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struct _GstWebRTCDataChannelClass
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{
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GObjectClass parent_class;
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void (*send_data) (GstWebRTCDataChannel * channel, GBytes *data);
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void (*send_string) (GstWebRTCDataChannel * channel, const gchar *str);
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void (*close) (GstWebRTCDataChannel * channel);
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gpointer _padding[GST_PADDING];
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};
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_open (GstWebRTCDataChannel * channel);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_close (GstWebRTCDataChannel * channel);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_error (GstWebRTCDataChannel * channel, GError * error);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_message_data (GstWebRTCDataChannel * channel, GBytes * data);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_message_string (GstWebRTCDataChannel * channel, const gchar * str);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_send_data (GstWebRTCDataChannel * channel, GBytes * data);
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@ -231,6 +231,69 @@ GST_WEBRTC_API
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void gst_webrtc_dtls_transport_set_transport (GstWebRTCDTLSTransport * transport,
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GstWebRTCICETransport * ice);
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#define GST_WEBRTC_DATA_CHANNEL_LOCK(channel) g_mutex_lock(&((GstWebRTCDataChannel *)(channel))->lock)
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#define GST_WEBRTC_DATA_CHANNEL_UNLOCK(channel) g_mutex_unlock(&((GstWebRTCDataChannel *)(channel))->lock)
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/**
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* GstWebRTCDataChannel:
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*
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* Since: 1.18
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*/
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struct _GstWebRTCDataChannel
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{
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GObject parent;
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GMutex lock;
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gchar *label;
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gboolean ordered;
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guint max_packet_lifetime;
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guint max_retransmits;
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gchar *protocol;
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gboolean negotiated;
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gint id;
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GstWebRTCPriorityType priority;
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GstWebRTCDataChannelState ready_state;
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guint64 buffered_amount;
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guint64 buffered_amount_low_threshold;
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gpointer _padding[GST_PADDING];
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};
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/**
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* GstWebRTCDataChannelClass:
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*
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* Since: 1.18
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*/
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struct _GstWebRTCDataChannelClass
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{
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GObjectClass parent_class;
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void (*send_data) (GstWebRTCDataChannel * channel, GBytes *data);
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void (*send_string) (GstWebRTCDataChannel * channel, const gchar *str);
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void (*close) (GstWebRTCDataChannel * channel);
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gpointer _padding[GST_PADDING];
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};
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_open (GstWebRTCDataChannel * channel);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_close (GstWebRTCDataChannel * channel);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_error (GstWebRTCDataChannel * channel, GError * error);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_message_data (GstWebRTCDataChannel * channel, GBytes * data);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_message_string (GstWebRTCDataChannel * channel, const gchar * str);
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GST_WEBRTC_API
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void gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel);
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G_END_DECLS
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