mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-27 00:38:21 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtsp: cleanup headers

Browse source 
Add padding, fix indentation, remove deprecated stuff
This commit is contained in:
Wim Taymans 2011-11-11 19:35:33 +01:00
parent 107d5a3d05
commit bdf3845498
7 changed files with 49 additions and 159 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

106
gst-libs/gst/rtsp/gstrtspconnection.c
View file

@ -3569,109 +3569,3 @@ gst_rtsp_watch_send_message (GstRTSPWatch * watch, GstRTSPMessage * message,
return gst_rtsp_watch_write_data (watch,
(guint8 *) g_string_free (str, FALSE), size, id);
}
/**
* gst_rtsp_watch_queue_data:
* @watch: a #GstRTSPWatch
* @data: the data to queue
* @size: the size of @data
*
* Queue @data for transmission in @watch. It will be transmitted when the
* connection of the @watch becomes writable.
*
* This function will take ownership of @data and g_free() it after use.
*
* The return value of this function will be used as the id argument in the
* message_sent callback.
*
* Deprecated: Use gst_rtsp_watch_write_data()
*
* Returns: an id.
*
* Since: 0.10.24
*/
#ifndef GST_REMOVE_DEPRECATED
#ifdef GST_DISABLE_DEPRECATED
guint
gst_rtsp_watch_queue_data (GstRTSPWatch * watch, const guint8 * data,
guint size);
#endif
guint
gst_rtsp_watch_queue_data (GstRTSPWatch * watch, const guint8 * data,
guint size)
{
GstRTSPRec *rec;
GMainContext *context = NULL;
g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (data != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (size != 0, GST_RTSP_EINVAL);
g_mutex_lock (watch->mutex);
/* make a record with the data and id */
rec = g_slice_new (GstRTSPRec);
rec->data = (guint8 *) data;
rec->size = size;
do {
/* make sure rec->id is never 0 */
rec->id = ++watch->id;
} while (G_UNLIKELY (rec->id == 0));
/* add the record to a queue. FIXME we would like to have an upper limit here */
g_queue_push_head (watch->messages, rec);
/* make sure the main context will now also check for writability on the
* socket */
if (watch->writefd.events != WRITE_COND) {
watch->writefd.events = WRITE_COND;
context = ((GSource *) watch)->context;
}
g_mutex_unlock (watch->mutex);
if (context)
g_main_context_wakeup (context);
return rec->id;
}
#endif /* GST_REMOVE_DEPRECATED */
/**
* gst_rtsp_watch_queue_message:
* @watch: a #GstRTSPWatch
* @message: a #GstRTSPMessage
*
* Queue a @message for transmission in @watch. The contents of this
* message will be serialized and transmitted when the connection of the
* @watch becomes writable.
*
* The return value of this function will be used as the id argument in the
* message_sent callback.
*
* Deprecated: Use gst_rtsp_watch_send_message()
*
* Returns: an id.
*
* Since: 0.10.23
*/
#ifndef GST_REMOVE_DEPRECATED
#ifdef GST_DISABLE_DEPRECATED
guint
gst_rtsp_watch_queue_message (GstRTSPWatch * watch, GstRTSPMessage * message);
#endif
guint
gst_rtsp_watch_queue_message (GstRTSPWatch * watch, GstRTSPMessage * message)
{
GString *str;
guint size;
g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
/* make a record with the message as a string and id */
str = message_to_string (watch->conn, message);
size = str->len;
return gst_rtsp_watch_queue_data (watch,
(guint8 *) g_string_free (str, FALSE), size);
}
#endif /* GST_REMOVE_DEPRECATED */

88
gst-libs/gst/rtsp/gstrtspconnection.h
View file

@ -60,75 +60,75 @@ G_BEGIN_DECLS
typedef struct _GstRTSPConnection GstRTSPConnection;
/* opening/closing a connection */
GstRTSPResult gst_rtsp_connection_create (const GstRTSPUrl *url, GstRTSPConnection **conn);
GstRTSPResult gst_rtsp_connection_create (const GstRTSPUrl *url, GstRTSPConnection **conn);
GstRTSPResult gst_rtsp_connection_create_from_fd (gint fd,
const gchar * ip,
guint16 port,
const gchar * initial_buffer,
GstRTSPConnection ** conn);
GstRTSPResult gst_rtsp_connection_accept (gint sock, GstRTSPConnection **conn);
GstRTSPResult gst_rtsp_connection_connect (GstRTSPConnection *conn, GTimeVal *timeout);
GstRTSPResult gst_rtsp_connection_close (GstRTSPConnection *conn);
GstRTSPResult gst_rtsp_connection_free (GstRTSPConnection *conn);
GstRTSPResult gst_rtsp_connection_accept (gint sock, GstRTSPConnection **conn);
GstRTSPResult gst_rtsp_connection_connect (GstRTSPConnection *conn, GTimeVal *timeout);
GstRTSPResult gst_rtsp_connection_close (GstRTSPConnection *conn);
GstRTSPResult gst_rtsp_connection_free (GstRTSPConnection *conn);
/* sending/receiving raw bytes */
GstRTSPResult gst_rtsp_connection_read (GstRTSPConnection * conn, guint8 * data,
guint size, GTimeVal * timeout);
GstRTSPResult gst_rtsp_connection_write (GstRTSPConnection * conn, const guint8 * data,
guint size, GTimeVal * timeout);
GstRTSPResult gst_rtsp_connection_read (GstRTSPConnection * conn, guint8 * data,
guint size, GTimeVal * timeout);
GstRTSPResult gst_rtsp_connection_write (GstRTSPConnection * conn, const guint8 * data,
guint size, GTimeVal * timeout);
/* sending/receiving messages */
GstRTSPResult gst_rtsp_connection_send (GstRTSPConnection *conn, GstRTSPMessage *message,
GTimeVal *timeout);
GstRTSPResult gst_rtsp_connection_receive (GstRTSPConnection *conn, GstRTSPMessage *message,
GTimeVal *timeout);
GstRTSPResult gst_rtsp_connection_send (GstRTSPConnection *conn, GstRTSPMessage *message,
GTimeVal *timeout);
GstRTSPResult gst_rtsp_connection_receive (GstRTSPConnection *conn, GstRTSPMessage *message,
GTimeVal *timeout);
/* status management */
GstRTSPResult gst_rtsp_connection_poll (GstRTSPConnection *conn, GstRTSPEvent events,
GstRTSPEvent *revents, GTimeVal *timeout);
GstRTSPResult gst_rtsp_connection_poll (GstRTSPConnection *conn, GstRTSPEvent events,
GstRTSPEvent *revents, GTimeVal *timeout);
/* reset the timeout */
GstRTSPResult gst_rtsp_connection_next_timeout (GstRTSPConnection *conn, GTimeVal *timeout);
GstRTSPResult gst_rtsp_connection_reset_timeout (GstRTSPConnection *conn);
GstRTSPResult gst_rtsp_connection_next_timeout (GstRTSPConnection *conn, GTimeVal *timeout);
GstRTSPResult gst_rtsp_connection_reset_timeout (GstRTSPConnection *conn);
/* flushing state */
GstRTSPResult gst_rtsp_connection_flush (GstRTSPConnection *conn, gboolean flush);
GstRTSPResult gst_rtsp_connection_flush (GstRTSPConnection *conn, gboolean flush);
/* HTTP proxy support */
GstRTSPResult gst_rtsp_connection_set_proxy (GstRTSPConnection *conn,
const gchar *host, guint port);
GstRTSPResult gst_rtsp_connection_set_proxy (GstRTSPConnection *conn,
const gchar *host, guint port);
/* configure authentication data */
GstRTSPResult gst_rtsp_connection_set_auth (GstRTSPConnection *conn, GstRTSPAuthMethod method,
const gchar *user, const gchar *pass);
GstRTSPResult gst_rtsp_connection_set_auth (GstRTSPConnection *conn, GstRTSPAuthMethod method,
const gchar *user, const gchar *pass);
void gst_rtsp_connection_set_auth_param (GstRTSPConnection *conn,
const gchar * param,
const gchar *value);
const gchar * param,
const gchar *value);
void gst_rtsp_connection_clear_auth_params (GstRTSPConnection *conn);
/* configure DSCP */
GstRTSPResult gst_rtsp_connection_set_qos_dscp (GstRTSPConnection *conn,
guint qos_dscp);
GstRTSPResult gst_rtsp_connection_set_qos_dscp (GstRTSPConnection *conn,
guint qos_dscp);
/* accessors */
GstRTSPUrl * gst_rtsp_connection_get_url (const GstRTSPConnection *conn);
const gchar * gst_rtsp_connection_get_ip (const GstRTSPConnection *conn);
void gst_rtsp_connection_set_ip (GstRTSPConnection *conn, const gchar *ip);
GstRTSPUrl * gst_rtsp_connection_get_url (const GstRTSPConnection *conn);
const gchar * gst_rtsp_connection_get_ip (const GstRTSPConnection *conn);
void gst_rtsp_connection_set_ip (GstRTSPConnection *conn, const gchar *ip);
gint gst_rtsp_connection_get_readfd (const GstRTSPConnection *conn);
gint gst_rtsp_connection_get_writefd (const GstRTSPConnection *conn);
gint gst_rtsp_connection_get_readfd (const GstRTSPConnection *conn);
gint gst_rtsp_connection_get_writefd (const GstRTSPConnection *conn);
void gst_rtsp_connection_set_http_mode (GstRTSPConnection *conn,
gboolean enable);
void gst_rtsp_connection_set_http_mode (GstRTSPConnection *conn,
gboolean enable);
/* tunneling */
void gst_rtsp_connection_set_tunneled (GstRTSPConnection *conn, gboolean tunneled);
gboolean gst_rtsp_connection_is_tunneled (const GstRTSPConnection *conn);
void gst_rtsp_connection_set_tunneled (GstRTSPConnection *conn, gboolean tunneled);
gboolean gst_rtsp_connection_is_tunneled (const GstRTSPConnection *conn);
const gchar * gst_rtsp_connection_get_tunnelid (const GstRTSPConnection *conn);
GstRTSPResult gst_rtsp_connection_do_tunnel (GstRTSPConnection *conn, GstRTSPConnection *conn2);
const gchar * gst_rtsp_connection_get_tunnelid (const GstRTSPConnection *conn);
GstRTSPResult gst_rtsp_connection_do_tunnel (GstRTSPConnection *conn, GstRTSPConnection *conn2);
/* async IO */
@ -176,13 +176,13 @@ typedef struct {
GstRTSPResult (*tunnel_lost) (GstRTSPWatch *watch, gpointer user_data);
/*< private >*/
gpointer _gst_reserved[GST_PADDING - 2];
gpointer _gst_reserved[GST_PADDING];
} GstRTSPWatchFuncs;
GstRTSPWatch * gst_rtsp_watch_new (GstRTSPConnection *conn,
GstRTSPWatchFuncs *funcs,
gpointer user_data,
GDestroyNotify notify);
gpointer user_data,
GDestroyNotify notify);
void gst_rtsp_watch_reset (GstRTSPWatch *watch);
void gst_rtsp_watch_unref (GstRTSPWatch *watch);
@ -196,14 +196,6 @@ GstRTSPResult gst_rtsp_watch_send_message (GstRTSPWatch *watch,
GstRTSPMessage *message,
guint *id);
#ifndef GST_DISABLE_DEPRECATED
guint gst_rtsp_watch_queue_data (GstRTSPWatch * watch,
const guint8 * data,
guint size);
guint gst_rtsp_watch_queue_message (GstRTSPWatch *watch,
GstRTSPMessage *message);
#endif
G_END_DECLS
#endif /* __GST_RTSP_CONNECTION_H__ */

4
gst-libs/gst/rtsp/gstrtspextension.h
View file

@ -68,14 +68,14 @@ struct _GstRTSPExtensionInterface {
GstRTSPResult (*receive_request) (GstRTSPExtension *ext, GstRTSPMessage *req);
/*< private >*/
gpointer _gst_reserved[GST_PADDING - 1];
gpointer _gst_reserved[GST_PADDING];
};
GType gst_rtsp_extension_get_type (void);
/* invoke vfunction on interface */
gboolean gst_rtsp_extension_detect_server (GstRTSPExtension *ext, GstRTSPMessage *resp);
GstRTSPResult gst_rtsp_extension_before_send (GstRTSPExtension *ext, GstRTSPMessage *req);
GstRTSPResult gst_rtsp_extension_after_send (GstRTSPExtension *ext, GstRTSPMessage *req,
GstRTSPMessage *resp);

2
gst-libs/gst/rtsp/gstrtspmessage.h
View file

@ -104,6 +104,8 @@ struct _GstRTSPMessage
guint8 *body;
guint body_size;
gpointer _gst_reserved[GST_PADDING];
};
/* memory management */

4
gst-libs/gst/rtsp/gstrtsprange.h
View file

@ -54,7 +54,7 @@ G_BEGIN_DECLS
* @GST_RTSP_RANGE_SMPTE: SMPTE timecode
* @GST_RTSP_RANGE_SMPTE_30_DROP: 29.97 frames per second
* @GST_RTSP_RANGE_SMPTE_25: 25 frames per second
* @GST_RTSP_RANGE_NPT: Normal play time
* @GST_RTSP_RANGE_NPT: Normal play time
* @GST_RTSP_RANGE_CLOCK: Absolute time expressed as ISO 8601 timestamps
*
* Different possible time range units.
@ -88,7 +88,7 @@ typedef enum {
/**
* GstRTSPTime:
* @type: the time of the time
* @seconds: seconds when @type is GST_RTSP_TIME_SECONDS
* @seconds: seconds when @type is GST_RTSP_TIME_SECONDS
*
* A time indication.
*/

2
gst-libs/gst/rtsp/gstrtsptransport.h
View file

@ -43,6 +43,7 @@
#ifndef __GST_RTSP_TRANSPORT_H__
#define __GST_RTSP_TRANSPORT_H__
#include <gst/gstconfig.h>
#include <gst/rtsp/gstrtspdefs.h>
G_BEGIN_DECLS
@ -156,6 +157,7 @@ struct _GstRTSPTransport {
/* RTP specific */
guint ssrc;
gpointer _gst_reserved[GST_PADDING];
};
GstRTSPResult gst_rtsp_transport_new (GstRTSPTransport **transport);

2
gst-libs/gst/rtsp/gstrtspurl.h
View file

@ -84,7 +84,7 @@ struct _GstRTSPUrl {
guint16 port;
gchar *abspath;
gchar *query;
};
};
GType gst_rtsp_url_get_type (void);