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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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rtsp: cleanup headers
Add padding, fix indentation, remove deprecated stuff
This commit is contained in:
parent
107d5a3d05
commit
bdf3845498
7 changed files with 49 additions and 159 deletions
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@ -3569,109 +3569,3 @@ gst_rtsp_watch_send_message (GstRTSPWatch * watch, GstRTSPMessage * message,
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return gst_rtsp_watch_write_data (watch,
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(guint8 *) g_string_free (str, FALSE), size, id);
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}
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/**
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* gst_rtsp_watch_queue_data:
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* @watch: a #GstRTSPWatch
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* @data: the data to queue
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* @size: the size of @data
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*
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* Queue @data for transmission in @watch. It will be transmitted when the
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* connection of the @watch becomes writable.
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*
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* This function will take ownership of @data and g_free() it after use.
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*
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* The return value of this function will be used as the id argument in the
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* message_sent callback.
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*
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* Deprecated: Use gst_rtsp_watch_write_data()
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*
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* Returns: an id.
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*
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* Since: 0.10.24
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*/
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#ifndef GST_REMOVE_DEPRECATED
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#ifdef GST_DISABLE_DEPRECATED
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guint
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gst_rtsp_watch_queue_data (GstRTSPWatch * watch, const guint8 * data,
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guint size);
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#endif
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guint
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gst_rtsp_watch_queue_data (GstRTSPWatch * watch, const guint8 * data,
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guint size)
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{
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GstRTSPRec *rec;
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GMainContext *context = NULL;
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g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
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g_return_val_if_fail (data != NULL, GST_RTSP_EINVAL);
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g_return_val_if_fail (size != 0, GST_RTSP_EINVAL);
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g_mutex_lock (watch->mutex);
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/* make a record with the data and id */
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rec = g_slice_new (GstRTSPRec);
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rec->data = (guint8 *) data;
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rec->size = size;
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do {
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/* make sure rec->id is never 0 */
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rec->id = ++watch->id;
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} while (G_UNLIKELY (rec->id == 0));
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/* add the record to a queue. FIXME we would like to have an upper limit here */
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g_queue_push_head (watch->messages, rec);
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/* make sure the main context will now also check for writability on the
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* socket */
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if (watch->writefd.events != WRITE_COND) {
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watch->writefd.events = WRITE_COND;
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context = ((GSource *) watch)->context;
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}
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g_mutex_unlock (watch->mutex);
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if (context)
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g_main_context_wakeup (context);
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return rec->id;
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}
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#endif /* GST_REMOVE_DEPRECATED */
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/**
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* gst_rtsp_watch_queue_message:
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* @watch: a #GstRTSPWatch
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* @message: a #GstRTSPMessage
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*
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* Queue a @message for transmission in @watch. The contents of this
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* message will be serialized and transmitted when the connection of the
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* @watch becomes writable.
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*
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* The return value of this function will be used as the id argument in the
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* message_sent callback.
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*
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* Deprecated: Use gst_rtsp_watch_send_message()
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*
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* Returns: an id.
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*
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* Since: 0.10.23
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*/
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#ifndef GST_REMOVE_DEPRECATED
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#ifdef GST_DISABLE_DEPRECATED
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guint
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gst_rtsp_watch_queue_message (GstRTSPWatch * watch, GstRTSPMessage * message);
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#endif
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guint
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gst_rtsp_watch_queue_message (GstRTSPWatch * watch, GstRTSPMessage * message)
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{
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GString *str;
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guint size;
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g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL);
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g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
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/* make a record with the message as a string and id */
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str = message_to_string (watch->conn, message);
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size = str->len;
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return gst_rtsp_watch_queue_data (watch,
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(guint8 *) g_string_free (str, FALSE), size);
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}
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#endif /* GST_REMOVE_DEPRECATED */
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@ -60,75 +60,75 @@ G_BEGIN_DECLS
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typedef struct _GstRTSPConnection GstRTSPConnection;
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/* opening/closing a connection */
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GstRTSPResult gst_rtsp_connection_create (const GstRTSPUrl *url, GstRTSPConnection **conn);
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GstRTSPResult gst_rtsp_connection_create (const GstRTSPUrl *url, GstRTSPConnection **conn);
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GstRTSPResult gst_rtsp_connection_create_from_fd (gint fd,
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const gchar * ip,
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guint16 port,
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const gchar * initial_buffer,
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GstRTSPConnection ** conn);
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GstRTSPResult gst_rtsp_connection_accept (gint sock, GstRTSPConnection **conn);
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GstRTSPResult gst_rtsp_connection_connect (GstRTSPConnection *conn, GTimeVal *timeout);
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GstRTSPResult gst_rtsp_connection_close (GstRTSPConnection *conn);
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GstRTSPResult gst_rtsp_connection_free (GstRTSPConnection *conn);
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GstRTSPResult gst_rtsp_connection_accept (gint sock, GstRTSPConnection **conn);
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GstRTSPResult gst_rtsp_connection_connect (GstRTSPConnection *conn, GTimeVal *timeout);
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GstRTSPResult gst_rtsp_connection_close (GstRTSPConnection *conn);
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GstRTSPResult gst_rtsp_connection_free (GstRTSPConnection *conn);
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/* sending/receiving raw bytes */
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GstRTSPResult gst_rtsp_connection_read (GstRTSPConnection * conn, guint8 * data,
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guint size, GTimeVal * timeout);
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GstRTSPResult gst_rtsp_connection_write (GstRTSPConnection * conn, const guint8 * data,
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guint size, GTimeVal * timeout);
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GstRTSPResult gst_rtsp_connection_read (GstRTSPConnection * conn, guint8 * data,
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guint size, GTimeVal * timeout);
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GstRTSPResult gst_rtsp_connection_write (GstRTSPConnection * conn, const guint8 * data,
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guint size, GTimeVal * timeout);
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/* sending/receiving messages */
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GstRTSPResult gst_rtsp_connection_send (GstRTSPConnection *conn, GstRTSPMessage *message,
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GTimeVal *timeout);
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GstRTSPResult gst_rtsp_connection_receive (GstRTSPConnection *conn, GstRTSPMessage *message,
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GTimeVal *timeout);
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GstRTSPResult gst_rtsp_connection_send (GstRTSPConnection *conn, GstRTSPMessage *message,
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GTimeVal *timeout);
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GstRTSPResult gst_rtsp_connection_receive (GstRTSPConnection *conn, GstRTSPMessage *message,
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GTimeVal *timeout);
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/* status management */
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GstRTSPResult gst_rtsp_connection_poll (GstRTSPConnection *conn, GstRTSPEvent events,
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GstRTSPEvent *revents, GTimeVal *timeout);
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GstRTSPResult gst_rtsp_connection_poll (GstRTSPConnection *conn, GstRTSPEvent events,
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GstRTSPEvent *revents, GTimeVal *timeout);
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/* reset the timeout */
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GstRTSPResult gst_rtsp_connection_next_timeout (GstRTSPConnection *conn, GTimeVal *timeout);
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GstRTSPResult gst_rtsp_connection_reset_timeout (GstRTSPConnection *conn);
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GstRTSPResult gst_rtsp_connection_next_timeout (GstRTSPConnection *conn, GTimeVal *timeout);
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GstRTSPResult gst_rtsp_connection_reset_timeout (GstRTSPConnection *conn);
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/* flushing state */
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GstRTSPResult gst_rtsp_connection_flush (GstRTSPConnection *conn, gboolean flush);
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GstRTSPResult gst_rtsp_connection_flush (GstRTSPConnection *conn, gboolean flush);
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/* HTTP proxy support */
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GstRTSPResult gst_rtsp_connection_set_proxy (GstRTSPConnection *conn,
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const gchar *host, guint port);
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GstRTSPResult gst_rtsp_connection_set_proxy (GstRTSPConnection *conn,
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const gchar *host, guint port);
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/* configure authentication data */
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GstRTSPResult gst_rtsp_connection_set_auth (GstRTSPConnection *conn, GstRTSPAuthMethod method,
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const gchar *user, const gchar *pass);
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GstRTSPResult gst_rtsp_connection_set_auth (GstRTSPConnection *conn, GstRTSPAuthMethod method,
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const gchar *user, const gchar *pass);
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void gst_rtsp_connection_set_auth_param (GstRTSPConnection *conn,
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const gchar * param,
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const gchar *value);
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const gchar * param,
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const gchar *value);
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void gst_rtsp_connection_clear_auth_params (GstRTSPConnection *conn);
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/* configure DSCP */
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GstRTSPResult gst_rtsp_connection_set_qos_dscp (GstRTSPConnection *conn,
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guint qos_dscp);
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GstRTSPResult gst_rtsp_connection_set_qos_dscp (GstRTSPConnection *conn,
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guint qos_dscp);
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/* accessors */
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GstRTSPUrl * gst_rtsp_connection_get_url (const GstRTSPConnection *conn);
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const gchar * gst_rtsp_connection_get_ip (const GstRTSPConnection *conn);
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void gst_rtsp_connection_set_ip (GstRTSPConnection *conn, const gchar *ip);
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GstRTSPUrl * gst_rtsp_connection_get_url (const GstRTSPConnection *conn);
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const gchar * gst_rtsp_connection_get_ip (const GstRTSPConnection *conn);
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void gst_rtsp_connection_set_ip (GstRTSPConnection *conn, const gchar *ip);
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gint gst_rtsp_connection_get_readfd (const GstRTSPConnection *conn);
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gint gst_rtsp_connection_get_writefd (const GstRTSPConnection *conn);
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gint gst_rtsp_connection_get_readfd (const GstRTSPConnection *conn);
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gint gst_rtsp_connection_get_writefd (const GstRTSPConnection *conn);
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void gst_rtsp_connection_set_http_mode (GstRTSPConnection *conn,
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gboolean enable);
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void gst_rtsp_connection_set_http_mode (GstRTSPConnection *conn,
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gboolean enable);
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/* tunneling */
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void gst_rtsp_connection_set_tunneled (GstRTSPConnection *conn, gboolean tunneled);
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gboolean gst_rtsp_connection_is_tunneled (const GstRTSPConnection *conn);
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void gst_rtsp_connection_set_tunneled (GstRTSPConnection *conn, gboolean tunneled);
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gboolean gst_rtsp_connection_is_tunneled (const GstRTSPConnection *conn);
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const gchar * gst_rtsp_connection_get_tunnelid (const GstRTSPConnection *conn);
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GstRTSPResult gst_rtsp_connection_do_tunnel (GstRTSPConnection *conn, GstRTSPConnection *conn2);
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const gchar * gst_rtsp_connection_get_tunnelid (const GstRTSPConnection *conn);
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GstRTSPResult gst_rtsp_connection_do_tunnel (GstRTSPConnection *conn, GstRTSPConnection *conn2);
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/* async IO */
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@ -176,13 +176,13 @@ typedef struct {
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GstRTSPResult (*tunnel_lost) (GstRTSPWatch *watch, gpointer user_data);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING - 2];
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gpointer _gst_reserved[GST_PADDING];
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} GstRTSPWatchFuncs;
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GstRTSPWatch * gst_rtsp_watch_new (GstRTSPConnection *conn,
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GstRTSPWatchFuncs *funcs,
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gpointer user_data,
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GDestroyNotify notify);
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gpointer user_data,
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GDestroyNotify notify);
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void gst_rtsp_watch_reset (GstRTSPWatch *watch);
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void gst_rtsp_watch_unref (GstRTSPWatch *watch);
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@ -196,14 +196,6 @@ GstRTSPResult gst_rtsp_watch_send_message (GstRTSPWatch *watch,
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GstRTSPMessage *message,
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guint *id);
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#ifndef GST_DISABLE_DEPRECATED
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guint gst_rtsp_watch_queue_data (GstRTSPWatch * watch,
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const guint8 * data,
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guint size);
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guint gst_rtsp_watch_queue_message (GstRTSPWatch *watch,
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GstRTSPMessage *message);
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#endif
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G_END_DECLS
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#endif /* __GST_RTSP_CONNECTION_H__ */
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@ -68,14 +68,14 @@ struct _GstRTSPExtensionInterface {
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GstRTSPResult (*receive_request) (GstRTSPExtension *ext, GstRTSPMessage *req);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING - 1];
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gpointer _gst_reserved[GST_PADDING];
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};
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GType gst_rtsp_extension_get_type (void);
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/* invoke vfunction on interface */
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gboolean gst_rtsp_extension_detect_server (GstRTSPExtension *ext, GstRTSPMessage *resp);
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GstRTSPResult gst_rtsp_extension_before_send (GstRTSPExtension *ext, GstRTSPMessage *req);
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GstRTSPResult gst_rtsp_extension_after_send (GstRTSPExtension *ext, GstRTSPMessage *req,
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GstRTSPMessage *resp);
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@ -104,6 +104,8 @@ struct _GstRTSPMessage
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guint8 *body;
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guint body_size;
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gpointer _gst_reserved[GST_PADDING];
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};
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/* memory management */
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@ -54,7 +54,7 @@ G_BEGIN_DECLS
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* @GST_RTSP_RANGE_SMPTE: SMPTE timecode
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* @GST_RTSP_RANGE_SMPTE_30_DROP: 29.97 frames per second
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* @GST_RTSP_RANGE_SMPTE_25: 25 frames per second
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* @GST_RTSP_RANGE_NPT: Normal play time
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* @GST_RTSP_RANGE_NPT: Normal play time
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* @GST_RTSP_RANGE_CLOCK: Absolute time expressed as ISO 8601 timestamps
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*
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* Different possible time range units.
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@ -88,7 +88,7 @@ typedef enum {
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/**
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* GstRTSPTime:
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* @type: the time of the time
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* @seconds: seconds when @type is GST_RTSP_TIME_SECONDS
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* @seconds: seconds when @type is GST_RTSP_TIME_SECONDS
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*
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* A time indication.
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*/
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@ -43,6 +43,7 @@
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#ifndef __GST_RTSP_TRANSPORT_H__
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#define __GST_RTSP_TRANSPORT_H__
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#include <gst/gstconfig.h>
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#include <gst/rtsp/gstrtspdefs.h>
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G_BEGIN_DECLS
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/* RTP specific */
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guint ssrc;
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gpointer _gst_reserved[GST_PADDING];
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};
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GstRTSPResult gst_rtsp_transport_new (GstRTSPTransport **transport);
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@ -84,7 +84,7 @@ struct _GstRTSPUrl {
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guint16 port;
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gchar *abspath;
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gchar *query;
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};
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};
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GType gst_rtsp_url_get_type (void);
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