avcodec/avcodec: Store whether AVCodec->close needs to be called

[ffmpeg] / libavformat / rtsp.h

diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h
index c38b90432d86eea9fd154c077dfe5105d88cc45b..1310dd9c08ab1b7bca3f0bfb8700acf0636b5011 100644 (file)
--- a/libavformat/rtsp.h
+++ b/libavformat/rtsp.h
@@ -2,20 +2,20 @@
  * RTSP definitions
  * Copyright (c) 2002 Fabrice Bellard
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg 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
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 #ifndef AVFORMAT_RTSP_H
@@ -27,6 +27,7 @@
 #include "rtpdec.h"
 #include "network.h"
 #include "httpauth.h"
+#include "internal.h"
 
 #include "libavutil/log.h"
 #include "libavutil/opt.h"
@@ -42,6 +43,7 @@ enum RTSPLowerTransport {
     RTSP_LOWER_TRANSPORT_HTTP = 8,          /**< HTTP tunneled - not a proper
                                                  transport mode as such,
                                                  only for use via AVOptions */
+    RTSP_LOWER_TRANSPORT_HTTPS,             /**< HTTPS tunneled */
     RTSP_LOWER_TRANSPORT_CUSTOM = 16,       /**< Custom IO - not a public
                                                  option for lower_transport_mask,
                                                  but set in the SDP demuxer based
@@ -76,7 +78,8 @@ enum RTSPControlTransport {
 #define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
 #define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
 #define RTSP_RTP_PORT_MIN 5000
-#define RTSP_RTP_PORT_MAX 10000
+#define RTSP_RTP_PORT_MAX 65000
+#define SDP_MAX_SIZE 16384
 
 /**
  * This describes a single item in the "Transport:" line of one stream as
@@ -185,6 +188,11 @@ typedef struct RTSPMessageHeader {
      * Content type header
      */
     char content_type[64];
+
+    /**
+     * SAT>IP com.ses.streamID header
+     */
+    char stream_id[64];
 } RTSPMessageHeader;
 
 /**
@@ -207,6 +215,7 @@ enum RTSPServerType {
     RTSP_SERVER_RTP,  /**< Standards-compliant RTP-server */
     RTSP_SERVER_REAL, /**< Realmedia-style server */
     RTSP_SERVER_WMS,  /**< Windows Media server */
+    RTSP_SERVER_SATIP,/**< SAT>IP server */
     RTSP_SERVER_NB
 };
 
@@ -314,7 +323,7 @@ typedef struct RTSPState {
     /** some MS RTSP streams contain a URL in the SDP that we need to use
      * for all subsequent RTSP requests, rather than the input URI; in
      * other cases, this is a copy of AVFormatContext->filename. */
-    char control_uri[1024];
+    char control_uri[MAX_URL_SIZE];
 
     /** The following are used for parsing raw mpegts in udp */
     //@{
@@ -392,16 +401,24 @@ typedef struct RTSPState {
      */
     int initial_timeout;
 
+    /**
+     * timeout of socket i/o operations.
+     */
+    int stimeout;
+
     /**
      * Size of RTP packet reordering queue.
      */
     int reordering_queue_size;
 
+    /**
+     * User-Agent string
+     */
+    char *user_agent;
+
     char default_lang[4];
     int buffer_size;
     int pkt_size;
-
-    const URLProtocol **protocols;
 } RTSPState;
 
 #define RTSP_FLAG_FILTER_SRC  0x1    /**< Filter incoming UDP packets -
@@ -411,6 +428,8 @@ typedef struct RTSPState {
 #define RTSP_FLAG_CUSTOM_IO   0x4    /**< Do all IO via the AVIOContext. */
 #define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
                                           address of received packets. */
+#define RTSP_FLAG_PREFER_TCP  0x10   /**< Try RTP via TCP first if possible. */
+#define RTSP_FLAG_SATIP_RAW   0x20   /**< Export SAT>IP stream as raw MPEG-TS */
 
 typedef struct RTSPSource {
     char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
@@ -433,7 +452,7 @@ typedef struct RTSPStream {
      * for the selected transport. Only used for TCP. */
     int interleaved_min, interleaved_max;
 
-    char control_url[1024];   /**< url for this stream (from SDP) */
+    char control_url[MAX_URL_SIZE];   /**< url for this stream (from SDP) */
 
     /** The following are used only in SDP, not RTSP */
     //@{
@@ -450,7 +469,7 @@ typedef struct RTSPStream {
     /** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
     //@{
     /** handler structure */
-    RTPDynamicProtocolHandler *dynamic_handler;
+    const RTPDynamicProtocolHandler *dynamic_handler;
 
     /** private data associated with the dynamic protocol */
     PayloadContext *dynamic_protocol_context;