gstreamer/gst/rtsp/gstrtspsrc.h

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
gst/rtsp/URLS: Added some test URLS. Original commit message from CVS: * gst/rtsp/URLS: Added some test URLS. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_loop), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: When creating streams, give access to the complete SDP. Fix some leaks. Collect and merge global stream properties in stream caps. Preliminary support for WMServer. * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_read), (read_body), (rtsp_connection_receive): * gst/rtsp/rtspconnection.h: Make connection interruptable. Refactor to make it reconnectable. Don't fail on short reads when reading data packets. * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port), (rtsp_url_get_port): * gst/rtsp/rtspurl.h: Add methods for getting/setting the port. * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_message_get_attribute_val), (sdp_media_get_attribute), (sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val), (sdp_media_get_format), (sdp_parse_line), (sdp_message_parse_buffer): Fix headers. Add methods for getting multiple attributes with the same name. Increase buffer size when parsing. Fix parsing of a=foo fields. * gst/rtsp/test.c: (main): Update to new connection API. * gst/rtsp/rtspmessage.c: (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/rtsptransport.c: (rtsp_transport_free): * gst/rtsp/rtsptransport.h: * gst/rtsp/sdp.h: * gst/rtsp/sdpmessage.h: * gst/rtsp/gstrtsp.c: * gst/rtsp/gstrtsp.h: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtpdec.h: * gst/rtsp/rtsp.h: * gst/rtsp/rtspdefs.c: * gst/rtsp/rtspdefs.h: Dual licensed under MIT and LGPL now.
2006-09-20 16:06:27 +00:00
* <2006> Wim Taymans <wim@fluendo.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.
*/
gst/rtsp/URLS: Added some test URLS. Original commit message from CVS: * gst/rtsp/URLS: Added some test URLS. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_loop), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: When creating streams, give access to the complete SDP. Fix some leaks. Collect and merge global stream properties in stream caps. Preliminary support for WMServer. * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_read), (read_body), (rtsp_connection_receive): * gst/rtsp/rtspconnection.h: Make connection interruptable. Refactor to make it reconnectable. Don't fail on short reads when reading data packets. * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port), (rtsp_url_get_port): * gst/rtsp/rtspurl.h: Add methods for getting/setting the port. * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_message_get_attribute_val), (sdp_media_get_attribute), (sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val), (sdp_media_get_format), (sdp_parse_line), (sdp_message_parse_buffer): Fix headers. Add methods for getting multiple attributes with the same name. Increase buffer size when parsing. Fix parsing of a=foo fields. * gst/rtsp/test.c: (main): Update to new connection API. * gst/rtsp/rtspmessage.c: (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/rtsptransport.c: (rtsp_transport_free): * gst/rtsp/rtsptransport.h: * gst/rtsp/sdp.h: * gst/rtsp/sdpmessage.h: * gst/rtsp/gstrtsp.c: * gst/rtsp/gstrtsp.h: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtpdec.h: * gst/rtsp/rtsp.h: * gst/rtsp/rtspdefs.c: * gst/rtsp/rtspdefs.h: Dual licensed under MIT and LGPL now.
2006-09-20 16:06:27 +00:00
/*
* 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_RTSPSRC_H__
#define __GST_RTSPSRC_H__
#include <gst/gst.h>
G_BEGIN_DECLS
#include "gstrtsp.h"
#include "rtsp.h"
#define GST_TYPE_RTSPSRC \
(gst_rtspsrc_get_type())
#define GST_RTSPSRC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSPSRC,GstRTSPSrc))
#define GST_RTSPSRC_CLASS(klass) \
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass Original commit message from CVS: * ext/aalib/gstaasink.h: * ext/annodex/gstcmmldec.h: * ext/cairo/gsttimeoverlay.h: * ext/dv/gstdvdec.h: * ext/dv/gstdvdemux.h: * ext/esd/esdmon.h: * ext/esd/esdsink.h: * ext/flac/gstflacenc.h: * ext/gconf/gstgconfaudiosink.h: * ext/gconf/gstgconfaudiosrc.h: * ext/gconf/gstgconfvideosink.h: * ext/gconf/gstgconfvideosrc.h: * ext/gdk_pixbuf/gstgdkanimation.h: * ext/gdk_pixbuf/pixbufscale.h: * ext/hal/gsthalaudiosink.h: * ext/hal/gsthalaudiosrc.h: * ext/jpeg/gstjpegenc.h: * ext/jpeg/gstsmokedec.h: * ext/jpeg/gstsmokeenc.h: * ext/libcaca/gstcacasink.h: * ext/libmng/gstmngdec.h: * ext/libmng/gstmngenc.h: * ext/libpng/gstpngdec.h: * ext/libpng/gstpngenc.h: * ext/raw1394/gstdv1394src.h: * ext/speex/gstspeexenc.h: * gst/autodetect/gstautoaudiosink.h: * gst/autodetect/gstautovideosink.h: * gst/avi/gstavidemux.h: * gst/cutter/gstcutter.h: * gst/debug/efence.h: * gst/debug/gstnavigationtest.h: * gst/debug/gstnavseek.h: * gst/flx/gstflxdec.h: * gst/goom/gstgoom.h: * gst/icydemux/gsticydemux.h: * gst/id3demux/gstid3demux.h: * gst/law/alaw-decode.h: * gst/law/alaw-encode.h: * gst/law/mulaw-decode.h: * gst/law/mulaw-encode.h: * gst/matroska/matroska-mux.h: * gst/median/gstmedian.h: * gst/oldcore/gstaggregator.h: * gst/oldcore/gstfdsink.h: * gst/oldcore/gstmd5sink.h: * gst/oldcore/gstmultifilesrc.h: * gst/oldcore/gstpipefilter.h: * gst/oldcore/gstshaper.h: * gst/oldcore/gststatistics.h: * gst/rtp/gstasteriskh263.h: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpdepay.h: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpapay.h: * gst/rtp/gstrtppcmadepay.h: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmudepay.h: * gst/rtp/gstrtppcmupay.h: * gst/rtp/gstrtpspeexdepay.h: * gst/rtp/gstrtpspeexpay.h: * gst/rtsp/gstrtpdec.h: * gst/rtsp/gstrtspsrc.h: * gst/smpte/gstsmpte.h: * gst/udp/gstdynudpsink.h: * gst/udp/gstmultiudpsink.h: * gst/udp/gstudpsink.h: * gst/udp/gstudpsrc.h: * gst/videofilter/gstvideobalance.h: * gst/videofilter/gstvideoflip.h: * sys/oss/gstossdmabuffer.h: * sys/oss/gstossmixerelement.h: * sys/oss/gstosssink.h: * sys/oss/gstosssrc.h: * sys/osxvideo/osxvideosink.h: * sys/sunaudio/gstsunaudiomixer.h: * sys/sunaudio/gstsunaudiosink.h: * sys/ximage/gstximagesrc.h: Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
2006-06-01 21:07:26 +00:00
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTSPSRC,GstRTSPSrcClass))
#define GST_IS_RTSPSRC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTSPSRC))
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass Original commit message from CVS: * ext/aalib/gstaasink.h: * ext/annodex/gstcmmldec.h: * ext/cairo/gsttimeoverlay.h: * ext/dv/gstdvdec.h: * ext/dv/gstdvdemux.h: * ext/esd/esdmon.h: * ext/esd/esdsink.h: * ext/flac/gstflacenc.h: * ext/gconf/gstgconfaudiosink.h: * ext/gconf/gstgconfaudiosrc.h: * ext/gconf/gstgconfvideosink.h: * ext/gconf/gstgconfvideosrc.h: * ext/gdk_pixbuf/gstgdkanimation.h: * ext/gdk_pixbuf/pixbufscale.h: * ext/hal/gsthalaudiosink.h: * ext/hal/gsthalaudiosrc.h: * ext/jpeg/gstjpegenc.h: * ext/jpeg/gstsmokedec.h: * ext/jpeg/gstsmokeenc.h: * ext/libcaca/gstcacasink.h: * ext/libmng/gstmngdec.h: * ext/libmng/gstmngenc.h: * ext/libpng/gstpngdec.h: * ext/libpng/gstpngenc.h: * ext/raw1394/gstdv1394src.h: * ext/speex/gstspeexenc.h: * gst/autodetect/gstautoaudiosink.h: * gst/autodetect/gstautovideosink.h: * gst/avi/gstavidemux.h: * gst/cutter/gstcutter.h: * gst/debug/efence.h: * gst/debug/gstnavigationtest.h: * gst/debug/gstnavseek.h: * gst/flx/gstflxdec.h: * gst/goom/gstgoom.h: * gst/icydemux/gsticydemux.h: * gst/id3demux/gstid3demux.h: * gst/law/alaw-decode.h: * gst/law/alaw-encode.h: * gst/law/mulaw-decode.h: * gst/law/mulaw-encode.h: * gst/matroska/matroska-mux.h: * gst/median/gstmedian.h: * gst/oldcore/gstaggregator.h: * gst/oldcore/gstfdsink.h: * gst/oldcore/gstmd5sink.h: * gst/oldcore/gstmultifilesrc.h: * gst/oldcore/gstpipefilter.h: * gst/oldcore/gstshaper.h: * gst/oldcore/gststatistics.h: * gst/rtp/gstasteriskh263.h: * gst/rtp/gstrtpL16depay.h: * gst/rtp/gstrtpL16pay.h: * gst/rtp/gstrtpamrdepay.h: * gst/rtp/gstrtpamrpay.h: * gst/rtp/gstrtpdepay.h: * gst/rtp/gstrtpgsmdepay.h: * gst/rtp/gstrtpgsmpay.h: * gst/rtp/gstrtph263pay.h: * gst/rtp/gstrtph263pdepay.h: * gst/rtp/gstrtph263ppay.h: * gst/rtp/gstrtpmp4gpay.h: * gst/rtp/gstrtpmp4vdepay.h: * gst/rtp/gstrtpmp4vpay.h: * gst/rtp/gstrtpmpadepay.h: * gst/rtp/gstrtpmpapay.h: * gst/rtp/gstrtppcmadepay.h: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmudepay.h: * gst/rtp/gstrtppcmupay.h: * gst/rtp/gstrtpspeexdepay.h: * gst/rtp/gstrtpspeexpay.h: * gst/rtsp/gstrtpdec.h: * gst/rtsp/gstrtspsrc.h: * gst/smpte/gstsmpte.h: * gst/udp/gstdynudpsink.h: * gst/udp/gstmultiudpsink.h: * gst/udp/gstudpsink.h: * gst/udp/gstudpsrc.h: * gst/videofilter/gstvideobalance.h: * gst/videofilter/gstvideoflip.h: * sys/oss/gstossdmabuffer.h: * sys/oss/gstossmixerelement.h: * sys/oss/gstosssink.h: * sys/oss/gstosssrc.h: * sys/osxvideo/osxvideosink.h: * sys/sunaudio/gstsunaudiomixer.h: * sys/sunaudio/gstsunaudiosink.h: * sys/ximage/gstximagesrc.h: Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
2006-06-01 21:07:26 +00:00
#define GST_IS_RTSPSRC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTSPSRC))
#define GST_RTSPSRC_CAST(obj) \
((GstRTSPSrc *)(obj))
typedef struct _GstRTSPSrc GstRTSPSrc;
typedef struct _GstRTSPSrcClass GstRTSPSrcClass;
#define GST_RTSP_LOOP_GET_COND(rtsp) (GST_RTSPSRC_CAST(rtsp)->loop_cond)
#define GST_RTSP_LOOP_WAIT(rtsp) (g_cond_wait(GST_RTSP_LOOP_GET_COND (rtsp), GST_OBJECT_GET_LOCK (rtsp)))
#define GST_RTSP_LOOP_SIGNAL(rtsp) (g_cond_signal(GST_RTSP_LOOP_GET_COND (rtsp)))
typedef struct _GstRTSPStream GstRTSPStream;
#include "rtspext.h"
struct _GstRTSPStream {
gint id;
gst/rtsp/: Allow url to be NULL to be able to use it for server connections. Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst/rtsp/COPYING.MIT: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup), (gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), (gst_rtspsrc_close): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_send), (read_line), (parse_request_line), (parse_line), (rtsp_connection_read), (rtsp_connection_close): * gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult), (rtsp_method_as_text), (rtsp_header_as_text), (rtsp_status_as_text), (rtsp_find_header_field), (rtsp_find_method): * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send), (rtsp_ext_wms_configure_stream): * gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init), (rtsp_message_new_request), (rtsp_message_init_request), (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_add_header), (rtsp_message_get_header), (rtsp_message_set_body), (rtsp_message_get_body), (dump_mem), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_media_get_attribute_val_n), (read_string), (read_string_del), (sdp_parse_line), (sdp_message_parse_buffer), (print_media), (sdp_message_dump): Allow url to be NULL to be able to use it for server connections. Can now send responses as well as requests. No longer hangs in an endless loop if EOF is received. Can now convert a status code to a text string. Return RTSP_HDR_INVALID for unknown headers. Return RTSP_INVALID for unknown methods. Copy CSeq and Session headers from the request. Only free memory corresponding to the currently set message type. Added const to function arguments as appropriate. Avoid a compiler warning when initializing nmedia. Use guint rather than gint to avoid compiler warnings. Fix crasher in wms extension. Factor out stream setup from open_connection. Delay activation of streams when actual data is received from the server, this prepares us to do proper protocol switching. Added new license. Fixes #380895.
2007-01-10 15:19:48 +00:00
GstRTSPSrc *parent; /* parent, no extra ref to parent is taken */
/* pad we expose or NULL when it does not have an actual pad */
GstPad *srcpad;
GstFlowReturn last_ret;
gst/rtsp/: Allow url to be NULL to be able to use it for server connections. Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst/rtsp/COPYING.MIT: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup), (gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), (gst_rtspsrc_close): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_send), (read_line), (parse_request_line), (parse_line), (rtsp_connection_read), (rtsp_connection_close): * gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult), (rtsp_method_as_text), (rtsp_header_as_text), (rtsp_status_as_text), (rtsp_find_header_field), (rtsp_find_method): * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send), (rtsp_ext_wms_configure_stream): * gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init), (rtsp_message_new_request), (rtsp_message_init_request), (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_add_header), (rtsp_message_get_header), (rtsp_message_set_body), (rtsp_message_get_body), (dump_mem), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_media_get_attribute_val_n), (read_string), (read_string_del), (sdp_parse_line), (sdp_message_parse_buffer), (print_media), (sdp_message_dump): Allow url to be NULL to be able to use it for server connections. Can now send responses as well as requests. No longer hangs in an endless loop if EOF is received. Can now convert a status code to a text string. Return RTSP_HDR_INVALID for unknown headers. Return RTSP_INVALID for unknown methods. Copy CSeq and Session headers from the request. Only free memory corresponding to the currently set message type. Added const to function arguments as appropriate. Avoid a compiler warning when initializing nmedia. Use guint rather than gint to avoid compiler warnings. Fix crasher in wms extension. Factor out stream setup from open_connection. Delay activation of streams when actual data is received from the server, this prepares us to do proper protocol switching. Added new license. Fixes #380895.
2007-01-10 15:19:48 +00:00
gboolean added;
/* for interleaved mode */
gint channel[2];
GstCaps *caps;
GstPad *channelpad[2];
/* our udp sources */
GstElement *udpsrc[2];
GstPad *blockedpad;
/* our udp sink back to the server */
GstElement *udpsink;
/* state */
gint pt;
gboolean container;
/* original control url */
gchar *control_url;
/* fully qualified control url */
gchar *setup_url;
guint32 ssrc;
guint32 seqbase;
guint64 timebase;
};
struct _GstRTSPSrc {
GstBin parent;
/* task and mutex for interleaved mode */
gboolean interleaved;
GstTask *task;
GStaticRecMutex *stream_rec_lock;
GstSegment segment;
gboolean running;
gint free_channel;
/* cond to signal loop */
GCond *loop_cond;
gint loop_cmd;
gint numstreams;
GList *streams;
gst/rtsp/URLS: Added some test URLS. Original commit message from CVS: * gst/rtsp/URLS: Added some test URLS. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_loop), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: When creating streams, give access to the complete SDP. Fix some leaks. Collect and merge global stream properties in stream caps. Preliminary support for WMServer. * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_read), (read_body), (rtsp_connection_receive): * gst/rtsp/rtspconnection.h: Make connection interruptable. Refactor to make it reconnectable. Don't fail on short reads when reading data packets. * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port), (rtsp_url_get_port): * gst/rtsp/rtspurl.h: Add methods for getting/setting the port. * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_message_get_attribute_val), (sdp_media_get_attribute), (sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val), (sdp_media_get_format), (sdp_parse_line), (sdp_message_parse_buffer): Fix headers. Add methods for getting multiple attributes with the same name. Increase buffer size when parsing. Fix parsing of a=foo fields. * gst/rtsp/test.c: (main): Update to new connection API. * gst/rtsp/rtspmessage.c: (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/rtsptransport.c: (rtsp_transport_free): * gst/rtsp/rtsptransport.h: * gst/rtsp/sdp.h: * gst/rtsp/sdpmessage.h: * gst/rtsp/gstrtsp.c: * gst/rtsp/gstrtsp.h: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtpdec.h: * gst/rtsp/rtsp.h: * gst/rtsp/rtspdefs.c: * gst/rtsp/rtspdefs.h: Dual licensed under MIT and LGPL now.
2006-09-20 16:06:27 +00:00
GstStructure *props;
gst/rtsp/: Allow url to be NULL to be able to use it for server connections. Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst/rtsp/COPYING.MIT: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup), (gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), (gst_rtspsrc_close): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_connect), (rtsp_connection_send), (read_line), (parse_request_line), (parse_line), (rtsp_connection_read), (rtsp_connection_close): * gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult), (rtsp_method_as_text), (rtsp_header_as_text), (rtsp_status_as_text), (rtsp_find_header_field), (rtsp_find_method): * gst/rtsp/rtspdefs.h: * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send), (rtsp_ext_wms_configure_stream): * gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init), (rtsp_message_new_request), (rtsp_message_init_request), (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_add_header), (rtsp_message_get_header), (rtsp_message_set_body), (rtsp_message_get_body), (dump_mem), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: * gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n), (sdp_media_get_attribute_val_n), (read_string), (read_string_del), (sdp_parse_line), (sdp_message_parse_buffer), (print_media), (sdp_message_dump): Allow url to be NULL to be able to use it for server connections. Can now send responses as well as requests. No longer hangs in an endless loop if EOF is received. Can now convert a status code to a text string. Return RTSP_HDR_INVALID for unknown headers. Return RTSP_INVALID for unknown methods. Copy CSeq and Session headers from the request. Only free memory corresponding to the currently set message type. Added const to function arguments as appropriate. Avoid a compiler warning when initializing nmedia. Use guint rather than gint to avoid compiler warnings. Fix crasher in wms extension. Factor out stream setup from open_connection. Delay activation of streams when actual data is received from the server, this prepares us to do proper protocol switching. Added new license. Fixes #380895.
2007-01-10 15:19:48 +00:00
gboolean need_activate;
/* properties */
gchar *location;
gchar *req_location; /* Sanitised URL to use in network requests */
RTSPUrl *url;
RTSPLowerTrans protocols;
gboolean debug;
guint retry;
guint64 timeout;
/* state */
gchar *content_base;
RTSPLowerTrans cur_protocols;
gboolean tried_url_auth;
gst/rtsp/gstrtspsrc.*: Small cleanups, added documentation. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_change_state), (gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri): * gst/rtsp/gstrtspsrc.h: Small cleanups, added documentation. Try to clean up the requests and responses. Refactor parsing the supported methods. * gst/rtsp/rtspconnection.c: (rtsp_connection_open), (rtsp_connection_create), (rtsp_connection_send), (parse_response_status), (parse_request_line), (rtsp_connection_receive), (rtsp_connection_close), (rtsp_connection_free): * gst/rtsp/rtsptransport.c: (rtsp_transport_new), (rtsp_transport_init), (rtsp_transport_parse), (rtsp_transport_free): * gst/rtsp/rtspurl.c: (rtsp_url_parse): * gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init), (sdp_message_clean), (sdp_message_free), (sdp_media_new), (sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump): Use g_return_val some more. * gst/rtsp/rtspdefs.h: Add more enum values to track initial states. * gst/rtsp/rtspmessage.c: (rtsp_message_new_request), (rtsp_message_init_request), (rtsp_message_new_response), (rtsp_message_init_response), (rtsp_message_init_data), (rtsp_message_unset), (rtsp_message_free), (rtsp_message_add_header), (rtsp_message_remove_header), (rtsp_message_get_header), (rtsp_message_set_body), (rtsp_message_take_body), (rtsp_message_get_body), (rtsp_message_steal_body), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: Reorder arguments, object goes as the first one. Use g_return_val some more.
2006-09-18 17:37:46 +00:00
/* supported methods */
gint methods;
gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback. Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session), (free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init), (gst_rtp_dec_init), (gst_rtp_dec_finalize), (gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp), (gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property), (gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock), (gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp), (create_rtcp), (gst_rtp_dec_request_new_pad), (gst_rtp_dec_release_pad): * gst/rtsp/gstrtpdec.h: * gst/rtsp/gstrtsp.c: (plugin_init): Morph RTPDec into something compatible with RTPBin as a fallback. Various other style fixes. * gst/rtsp/gstrtspsrc.c: (find_stream_by_id), (find_stream_by_udpsrc), (gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps), (new_session_pad), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Implement RTPBin session manager handling. Don't try to add empty properties to caps. Implement fallback session manager, handling. Don't combine errors from RTCP streams, just ignore them. * gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager): * gst/rtsp/rtsptransport.h: Implement fallback session manager. Make RTPBin the default one when available.
2007-04-06 12:54:16 +00:00
/* session management */
GstElement *session;
gulong session_sig_id;
gulong session_ptmap_id;
gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback. Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session), (free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init), (gst_rtp_dec_init), (gst_rtp_dec_finalize), (gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp), (gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property), (gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock), (gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp), (create_rtcp), (gst_rtp_dec_request_new_pad), (gst_rtp_dec_release_pad): * gst/rtsp/gstrtpdec.h: * gst/rtsp/gstrtsp.c: (plugin_init): Morph RTPDec into something compatible with RTPBin as a fallback. Various other style fixes. * gst/rtsp/gstrtspsrc.c: (find_stream_by_id), (find_stream_by_udpsrc), (gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps), (new_session_pad), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Implement RTPBin session manager handling. Don't try to add empty properties to caps. Implement fallback session manager, handling. Don't combine errors from RTCP streams, just ignore them. * gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager): * gst/rtsp/rtsptransport.h: Implement fallback session manager. Make RTPBin the default one when available.
2007-04-06 12:54:16 +00:00
RTSPConnection *connection;
RTSPExtensionCtx *extension;
};
struct _GstRTSPSrcClass {
GstBinClass parent_class;
};
GType gst_rtspsrc_get_type(void);
gboolean gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code);
G_END_DECLS
#endif /* __GST_RTSPSRC_H__ */