gstreamer 1.1.0-x86-mingw32

data/vendor/local/include/gstreamer-0.10/gst/rtp/gstrtppayloads.h

@@ -0,0 +1,193 @@
+/* GStreamer
+ * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * gstrtppayloads.h: various helper functions to deal with RTP payload
+ *     types.
+ *
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_RTPPAYLOADS_H__
+#define __GST_RTPPAYLOADS_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+/** 
+ * GstRTPPayload:
+ * @GST_RTP_PAYLOAD_PCMU: ITU-T G.711. mu-law audio (RFC 3551)
+ * @GST_RTP_PAYLOAD_1016: RFC 3551 says reserved
+ * @GST_RTP_PAYLOAD_G721: RFC 3551 says reserved
+ * @GST_RTP_PAYLOAD_GSM: GSM audio
+ * @GST_RTP_PAYLOAD_G723: ITU G.723.1 audio
+ * @GST_RTP_PAYLOAD_DVI4_8000: IMA ADPCM wave type (RFC 3551)
+ * @GST_RTP_PAYLOAD_DVI4_16000: IMA ADPCM wave type (RFC 3551)
+ * @GST_RTP_PAYLOAD_LPC: experimental linear predictive encoding
+ * @GST_RTP_PAYLOAD_PCMA: ITU-T G.711 A-law audio (RFC 3551)
+ * @GST_RTP_PAYLOAD_G722: ITU-T G.722 (RFC 3551)
+ * @GST_RTP_PAYLOAD_L16_STEREO: stereo PCM
+ * @GST_RTP_PAYLOAD_L16_MONO: mono PCM
+ * @GST_RTP_PAYLOAD_QCELP: EIA & TIA standard IS-733
+ * @GST_RTP_PAYLOAD_CN: Comfort Noise (RFC 3389)
+ * @GST_RTP_PAYLOAD_MPA: Audio MPEG 1-3.
+ * @GST_RTP_PAYLOAD_G728: ITU-T G.728 Speech coder (RFC 3551)
+ * @GST_RTP_PAYLOAD_DVI4_11025: IMA ADPCM wave type (RFC 3551)
+ * @GST_RTP_PAYLOAD_DVI4_22050: IMA ADPCM wave type (RFC 3551)
+ * @GST_RTP_PAYLOAD_G729: ITU-T G.729 Speech coder (RFC 3551)
+ * @GST_RTP_PAYLOAD_CELLB: See RFC 2029
+ * @GST_RTP_PAYLOAD_JPEG: ISO Standards 10918-1 and 10918-2 (RFC 2435)
+ * @GST_RTP_PAYLOAD_NV: nv encoding by Ron Frederick
+ * @GST_RTP_PAYLOAD_H261: ITU-T Recommendation H.261 (RFC 2032)
+ * @GST_RTP_PAYLOAD_MPV: Video MPEG 1 & 2 (RFC 2250)
+ * @GST_RTP_PAYLOAD_MP2T: MPEG-2 transport stream (RFC 2250)
+ * @GST_RTP_PAYLOAD_H263: Video H263 (RFC 2190)
+ *
+ *
+ * Standard predefined fixed payload types.
+ *
+ * The official list is at:
+ * http://www.iana.org/assignments/rtp-parameters
+ *
+ * Audio:
+ * reserved: 19
+ * unassigned: 20-23, 
+ *
+ * Video:
+ * unassigned: 24, 27, 29, 30, 35-71, 77-95
+ * Reserved for RTCP conflict avoidance: 72-76
+ */
+typedef enum
+{
+  /* Audio: */
+  GST_RTP_PAYLOAD_PCMU = 0,
+  GST_RTP_PAYLOAD_1016 = 1, /* RFC 3551 says reserved */
+  GST_RTP_PAYLOAD_G721 = 2, /* RFC 3551 says reserved */
+  GST_RTP_PAYLOAD_GSM = 3,
+  GST_RTP_PAYLOAD_G723 = 4,
+  GST_RTP_PAYLOAD_DVI4_8000 = 5,
+  GST_RTP_PAYLOAD_DVI4_16000 = 6,
+  GST_RTP_PAYLOAD_LPC = 7,
+  GST_RTP_PAYLOAD_PCMA = 8,
+  GST_RTP_PAYLOAD_G722 = 9,
+  GST_RTP_PAYLOAD_L16_STEREO = 10,
+  GST_RTP_PAYLOAD_L16_MONO = 11,
+  GST_RTP_PAYLOAD_QCELP = 12,
+  GST_RTP_PAYLOAD_CN = 13,
+  GST_RTP_PAYLOAD_MPA = 14,
+  GST_RTP_PAYLOAD_G728 = 15,
+  GST_RTP_PAYLOAD_DVI4_11025 = 16,
+  GST_RTP_PAYLOAD_DVI4_22050 = 17,
+  GST_RTP_PAYLOAD_G729 = 18,   
+
+  /* Video: */
+
+  GST_RTP_PAYLOAD_CELLB = 25,
+  GST_RTP_PAYLOAD_JPEG = 26,
+  GST_RTP_PAYLOAD_NV = 28,
+  GST_RTP_PAYLOAD_H261 = 31,
+  GST_RTP_PAYLOAD_MPV = 32,
+  GST_RTP_PAYLOAD_MP2T = 33,
+  GST_RTP_PAYLOAD_H263 = 34,
+
+  /* BOTH */
+} GstRTPPayload;
+
+/* backward compatibility */
+#define GST_RTP_PAYLOAD_G723_63 16
+#define GST_RTP_PAYLOAD_G723_53 17
+#define GST_RTP_PAYLOAD_TS48 18
+#define GST_RTP_PAYLOAD_TS41 19
+
+#define GST_RTP_PAYLOAD_G723_63_STRING "16"
+#define GST_RTP_PAYLOAD_G723_53_STRING "17"
+#define GST_RTP_PAYLOAD_TS48_STRING "18"
+#define GST_RTP_PAYLOAD_TS41_STRING "19"
+
+/* Defining the above as strings, to make the declaration of pad_templates
+ * easier. So if please keep these synchronized with the above.
+ */
+#define GST_RTP_PAYLOAD_PCMU_STRING "0"
+#define GST_RTP_PAYLOAD_1016_STRING "1"
+#define GST_RTP_PAYLOAD_G721_STRING "2"
+#define GST_RTP_PAYLOAD_GSM_STRING "3"
+#define GST_RTP_PAYLOAD_G723_STRING "4"
+#define GST_RTP_PAYLOAD_DVI4_8000_STRING "5"
+#define GST_RTP_PAYLOAD_DVI4_16000_STRING "6"
+#define GST_RTP_PAYLOAD_LPC_STRING "7"
+#define GST_RTP_PAYLOAD_PCMA_STRING "8"
+#define GST_RTP_PAYLOAD_G722_STRING "9"
+#define GST_RTP_PAYLOAD_L16_STEREO_STRING "10"
+#define GST_RTP_PAYLOAD_L16_MONO_STRING "11"
+#define GST_RTP_PAYLOAD_QCELP_STRING "12"
+#define GST_RTP_PAYLOAD_CN_STRING "13"
+#define GST_RTP_PAYLOAD_MPA_STRING "14"
+#define GST_RTP_PAYLOAD_G728_STRING "15"
+#define GST_RTP_PAYLOAD_DVI4_11025_STRING "16"
+#define GST_RTP_PAYLOAD_DVI4_22050_STRING "17"
+#define GST_RTP_PAYLOAD_G729_STRING "18"
+
+#define GST_RTP_PAYLOAD_CELLB_STRING "25"
+#define GST_RTP_PAYLOAD_JPEG_STRING "26"
+#define GST_RTP_PAYLOAD_NV_STRING "28"
+
+#define GST_RTP_PAYLOAD_H261_STRING "31"
+#define GST_RTP_PAYLOAD_MPV_STRING "32"
+#define GST_RTP_PAYLOAD_MP2T_STRING "33"
+#define GST_RTP_PAYLOAD_H263_STRING "34"
+
+#define GST_RTP_PAYLOAD_DYNAMIC_STRING "[96, 127]"
+
+/**
+ * GST_RTP_PAYLOAD_IS_DYNAMIC:
+ * @pt: a payload type
+ *
+ * Check if @pt is a dynamic payload type.
+ */
+#define GST_RTP_PAYLOAD_IS_DYNAMIC(pt) ((pt) >= 96 && (pt) <= 127)
+
+typedef struct _GstRTPPayloadInfo GstRTPPayloadInfo;
+
+/**
+ * GstRTPPayloadInfo:
+ * @payload_type: payload type, -1 means dynamic
+ * @media: the media type(s), usually "audio", "video", "application", "text",
+ * "message".
+ * @encoding_name: the encoding name of @pt
+ * @clock_rate: default clock rate, 0 = unknown/variable
+ * @encoding_parameters: encoding parameters. For audio this is the number of
+ * channels. NULL = not applicable.
+ * @bitrate: the bitrate of the media. 0 = unknown/variable.
+ *
+ * Structure holding default payload type information.
+ */
+struct _GstRTPPayloadInfo
+{
+  guint8       payload_type;
+  const gchar *media;
+  const gchar *encoding_name;
+  guint        clock_rate;
+  const gchar *encoding_parameters;
+  guint        bitrate;
+};
+
+const GstRTPPayloadInfo *  gst_rtp_payload_info_for_pt     (guint8 payload_type);
+const GstRTPPayloadInfo *  gst_rtp_payload_info_for_name   (const gchar *media, const gchar *encoding_name);
+
+G_END_DECLS
+
+#endif /* __GST_RTPPAYLOADS_H__ */
+

data/vendor/local/include/gstreamer-0.10/gst/rtsp/gstrtsp-enumtypes.h

@@ -0,0 +1,35 @@
+
+
+
+#ifndef __gst_rtsp_ENUM_TYPES_H__
+#define __gst_rtsp_ENUM_TYPES_H__
+
+#include <glib-object.h>
+
+G_BEGIN_DECLS
+
+/* enumerations from "gstrtspdefs.h" */
+GType gst_rtsp_result_get_type (void);
+#define GST_TYPE_RTSP_RESULT (gst_rtsp_result_get_type())
+GType gst_rtsp_event_get_type (void);
+#define GST_TYPE_RTSP_EVENT (gst_rtsp_event_get_type())
+GType gst_rtsp_family_get_type (void);
+#define GST_TYPE_RTSP_FAMILY (gst_rtsp_family_get_type())
+GType gst_rtsp_state_get_type (void);
+#define GST_TYPE_RTSP_STATE (gst_rtsp_state_get_type())
+GType gst_rtsp_version_get_type (void);
+#define GST_TYPE_RTSP_VERSION (gst_rtsp_version_get_type())
+GType gst_rtsp_method_get_type (void);
+#define GST_TYPE_RTSP_METHOD (gst_rtsp_method_get_type())
+GType gst_rtsp_auth_method_get_type (void);
+#define GST_TYPE_RTSP_AUTH_METHOD (gst_rtsp_auth_method_get_type())
+GType gst_rtsp_header_field_get_type (void);
+#define GST_TYPE_RTSP_HEADER_FIELD (gst_rtsp_header_field_get_type())
+GType gst_rtsp_status_code_get_type (void);
+#define GST_TYPE_RTSP_STATUS_CODE (gst_rtsp_status_code_get_type())
+G_END_DECLS
+
+#endif /* __gst_rtsp_ENUM_TYPES_H__ */
+
+
+

data/vendor/local/include/gstreamer-0.10/gst/rtsp/gstrtspbase64.h

@@ -0,0 +1,37 @@
+/* GStreamer
+ * Copyright (C) <2007> Mike Smith <msmith@xiph.org>
+ *
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_RTSP_BASE64_H__
+#define __GST_RTSP_BASE64_H__
+
+#include <glib.h>
+
+G_BEGIN_DECLS
+
+#ifndef GST_DISABLE_DEPRECATED
+gchar *gst_rtsp_base64_encode    (const gchar *data, gsize len);
+#endif
+
+#ifndef GST_DISABLE_DEPRECATED
+void   gst_rtsp_base64_decode_ip (gchar *data, gsize *len);
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_BASE64_H__ */

data/vendor/local/include/gstreamer-0.10/gst/rtsp/gstrtspconnection.h

@@ -0,0 +1,209 @@
+/* GStreamer
+ * Copyright (C) <2005,2009> Wim Taymans <wim.taymans@gmail.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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_RTSP_CONNECTION_H__
+#define __GST_RTSP_CONNECTION_H__
+
+#include <glib.h>
+
+#include <gst/gstconfig.h>
+#include <gst/rtsp/gstrtspdefs.h>
+#include <gst/rtsp/gstrtspurl.h>
+#include <gst/rtsp/gstrtspmessage.h>
+
+G_BEGIN_DECLS
+
+/**
+ * GstRTSPConnection:
+ *
+ * Opaque RTSP connection object.
+ */
+typedef struct _GstRTSPConnection GstRTSPConnection;
+
+/* opening/closing a connection */
+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);
+
+
+/* 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);
+
+/* 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);
+
+/* status management */
+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);
+
+/* flushing state */
+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);
+
+/* configure authentication data */
+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);
+void               gst_rtsp_connection_clear_auth_params (GstRTSPConnection *conn);
+
+/* configure 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);
+
+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);
+
+/* tunneling */
+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);
+
+/* async IO */
+
+/**
+ * GstRTSPWatch:
+ *
+ * Opaque RTSP watch object that can be used for asynchronous RTSP
+ * operations.
+ */
+typedef struct _GstRTSPWatch GstRTSPWatch;
+
+/**
+ * GstRTSPWatchFuncs:
+ * @message_received: callback when a message was received
+ * @message_sent: callback when a message was sent
+ * @closed: callback when the connection is closed
+ * @error: callback when an error occured
+ * @tunnel_start: a client started a tunneled connection. The tunnelid of the
+ *   connection must be saved.
+ * @tunnel_complete: a client finished a tunneled connection. In this callback
+ *   you usually pair the tunnelid of this connection with the saved one using
+ *   gst_rtsp_connection_do_tunnel().
+ * @error_full: callback when an error occured with more information than
+ *   the @error callback. Since 0.10.25
+ * @tunnel_lost: callback when the post connection of a tunnel is closed.
+ *   Since 0.10.29
+ *
+ * Callback functions from a #GstRTSPWatch.
+ *
+ * Since: 0.10.23
+ */
+typedef struct {
+  GstRTSPResult     (*message_received) (GstRTSPWatch *watch, GstRTSPMessage *message,
+                                         gpointer user_data);
+  GstRTSPResult     (*message_sent)     (GstRTSPWatch *watch, guint id,
+                                         gpointer user_data);
+  GstRTSPResult     (*closed)           (GstRTSPWatch *watch, gpointer user_data);
+  GstRTSPResult     (*error)            (GstRTSPWatch *watch, GstRTSPResult result,
+                                         gpointer user_data);
+  GstRTSPStatusCode (*tunnel_start)     (GstRTSPWatch *watch, gpointer user_data);
+  GstRTSPResult     (*tunnel_complete)  (GstRTSPWatch *watch, gpointer user_data);
+  GstRTSPResult     (*error_full)       (GstRTSPWatch *watch, GstRTSPResult result,
+                                         GstRTSPMessage *message, guint id,
+                                         gpointer user_data);
+  GstRTSPResult     (*tunnel_lost)      (GstRTSPWatch *watch, gpointer user_data);
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING - 2];
+} GstRTSPWatchFuncs;
+
+GstRTSPWatch *     gst_rtsp_watch_new                (GstRTSPConnection *conn,
+                                                      GstRTSPWatchFuncs *funcs,
+						      gpointer user_data,
+						      GDestroyNotify notify);
+void               gst_rtsp_watch_reset              (GstRTSPWatch *watch);
+void               gst_rtsp_watch_unref              (GstRTSPWatch *watch);
+
+guint              gst_rtsp_watch_attach             (GstRTSPWatch *watch,
+                                                      GMainContext *context);
+
+GstRTSPResult      gst_rtsp_watch_write_data         (GstRTSPWatch *watch,
+                                                      const guint8 *data,
+                                                      guint size, guint *id);
+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__ */