gstreamer/gst/rtsp/gstrtspsrc.c

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/* GStreamer
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
* Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
* <2006> Lutz Mueller <lutz at topfrose dot de>
*
* 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.
*/
/**
* SECTION:element-rtspsrc
*
* <refsect2>
* <para>
* Makes a connection to an RTSP server and read the data.
* rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
* RealMedia/Quicktime/Microsoft extensions.
* </para>
* <para>
* RTSP supports transport over TCP or UDP in unicast or multicast mode. By
* default rtspsrc will negotiate a connection in the following order:
* UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
* protocols can be controlled with the "protocols" property.
* </para>
* <para>
* rtspsrc currently understands SDP as the format of the session description.
* For each stream listed in the SDP a new rtp_stream%d pad will be created
* with caps derived from the SDP media description. This is a caps of mime type
* "application/x-rtp" that can be connected to any available RTP depayloader
* element.
* </para>
* <para>
* rtspsrc will internally instantiate an RTP session manager element
* that will handle the RTCP messages to and from the server, jitter removal,
* packet reordering along with providing a clock for the pipeline.
* This feature is however currently not yet implemented.
* </para>
* <para>
* rtspsrc acts like a live source and will therefore only generate data in the
* PLAYING state.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
* </programlisting>
* Establish a connection to an RTSP server and send the raw RTP packets to a fakesink.
* </para>
* </refsect2>
*
* Last reviewed on 2006-08-18 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <unistd.h>
#include <stdlib.h>
#include <string.h>
#include "gstrtspsrc.h"
#include "sdp.h"
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
/* define for experimental real support */
#undef WITH_EXT_REAL
#include "rtspextwms.h"
#ifdef WITH_EXT_REAL
#include "rtspextreal.h"
#endif
GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
#define GST_CAT_DEFAULT (rtspsrc_debug)
/* elementfactory information */
Define GstElementDetails as const and also static (when defined as global) Original commit message from CVS: * ext/aalib/gstaasink.c: * ext/annodex/gstcmmldec.c: * ext/annodex/gstcmmlenc.c: * ext/cairo/gsttextoverlay.c: * ext/cairo/gsttimeoverlay.c: * ext/cdio/gstcdiocddasrc.c: * ext/dv/gstdvdec.c: * ext/dv/gstdvdemux.c: * ext/esd/esdmon.c: * ext/esd/esdsink.c: * ext/flac/gstflacenc.c: * ext/flac/gstflactag.c: * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init): * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init): * ext/gdk_pixbuf/pixbufscale.c: * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init): * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init): * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstsmokedec.c: * ext/jpeg/gstsmokeenc.c: * ext/libcaca/gstcacasink.c: * ext/libmng/gstmngdec.c: * ext/libmng/gstmngenc.c: * ext/libpng/gstpngdec.c: * ext/libpng/gstpngenc.c: * ext/mikmod/gstmikmod.c: * ext/raw1394/gstdv1394src.c: * ext/shout2/gstshout2.c: (gst_shout2send_init): * ext/shout2/gstshout2.h: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * gst/alpha/gstalpha.c: * gst/alpha/gstalphacolor.c: * gst/apetag/gstapedemux.c: * gst/auparse/gstauparse.c: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_base_init): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_base_init): * gst/avi/gstavidemux.c: (gst_avi_demux_base_init): * gst/avi/gstavimux.c: (gst_avimux_base_init): * gst/cutter/gstcutter.c: * gst/debug/breakmydata.c: * gst/debug/efence.c: * gst/debug/gstnavigationtest.c: * gst/debug/gstnavseek.c: * gst/debug/negotiation.c: * gst/debug/progressreport.c: * gst/debug/testplugin.c: * gst/effectv/gstaging.c: * gst/effectv/gstdice.c: * gst/effectv/gstedge.c: * gst/effectv/gstquark.c: * gst/effectv/gstrev.c: * gst/effectv/gstshagadelic.c: * gst/effectv/gstvertigo.c: * gst/effectv/gstwarp.c: * gst/flx/gstflxdec.c: * gst/goom/gstgoom.c: * gst/icydemux/gsticydemux.c: * gst/id3demux/gstid3demux.c: * gst/interleave/deinterleave.c: * gst/interleave/interleave.c: * gst/law/alaw-decode.c: (gst_alawdec_base_init): * gst/law/alaw-encode.c: (gst_alawenc_base_init): * gst/law/mulaw-decode.c: (gst_mulawdec_base_init): * gst/law/mulaw-encode.c: (gst_mulawenc_base_init): * gst/level/gstlevel.c: * gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init): * gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init): * gst/median/gstmedian.c: * gst/monoscope/gstmonoscope.c: * gst/multipart/multipartdemux.c: * gst/multipart/multipartmux.c: * gst/oldcore/gstaggregator.c: * gst/oldcore/gstfdsink.c: * gst/oldcore/gstmd5sink.c: * gst/oldcore/gstmultifilesrc.c: * gst/oldcore/gstpipefilter.c: * gst/oldcore/gstshaper.c: * gst/oldcore/gststatistics.c: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtsp/gstrtpdec.c: * gst/rtsp/gstrtspsrc.c: * gst/smpte/gstsmpte.c: * gst/udp/gstdynudpsink.c: * gst/udp/gstmultiudpsink.c: * gst/udp/gstudpsink.c: * gst/udp/gstudpsrc.c: * gst/videobox/gstvideobox.c: * gst/videofilter/gstgamma.c: (gst_gamma_base_init): * gst/videofilter/gstvideobalance.c: * gst/videofilter/gstvideoflip.c: * gst/videofilter/gstvideotemplate.c: (gst_videotemplate_base_init): * gst/videomixer/videomixer.c: * gst/wavparse/gstwavparse.c: (gst_wavparse_base_init), (gst_wavparse_class_init), (gst_wavparse_dispose), (gst_wavparse_reset), (gst_wavparse_init), (gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info), (gst_wavparse_peek_chunk), (gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init), (gst_wavparse_send_event), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data), (gst_wavparse_chain), (gst_wavparse_srcpad_event), (gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull), (gst_wavparse_change_state): * gst/wavparse/gstwavparse.h: * sys/oss/gstossmixerelement.c: * sys/oss/gstosssink.c: * sys/oss/gstosssrc.c: * sys/osxaudio/gstosxaudioelement.c: * sys/osxaudio/gstosxaudiosink.c: * sys/osxaudio/gstosxaudiosrc.c: * sys/sunaudio/gstsunaudiomixer.c: * sys/sunaudio/gstsunaudiosink.c: Define GstElementDetails as const and also static (when defined as global)
2006-04-25 21:39:46 +00:00
static const GstElementDetails gst_rtspsrc_details =
GST_ELEMENT_DETAILS ("RTSP packet receiver",
"Source/Network",
"Receive data over the network via RTSP (RFC 2326)",
"Wim Taymans <wim@fluendo.com>\n"
"Thijs Vermeir <thijs.vermeir@barco.com>\n"
"Lutz Mueller <lutz@topfrose.de>");
static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream%d",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_LOCATION NULL
#define DEFAULT_PROTOCOLS RTSP_LOWER_TRANS_UDP | RTSP_LOWER_TRANS_UDP_MCAST | RTSP_LOWER_TRANS_TCP
#define DEFAULT_DEBUG FALSE
#define DEFAULT_RETRY 20
#define DEFAULT_TIMEOUT 5000000
enum
{
PROP_0,
PROP_LOCATION,
PROP_PROTOCOLS,
PROP_DEBUG,
PROP_RETRY,
PROP_TIMEOUT,
};
#define GST_TYPE_RTSP_LOWER_TRANS (gst_rtsp_lower_trans_get_type())
static GType
gst_rtsp_lower_trans_get_type (void)
{
static GType rtsp_lower_trans_type = 0;
static const GFlagsValue rtsp_lower_trans[] = {
{RTSP_LOWER_TRANS_UDP, "UDP Unicast Mode", "udp-unicast"},
{RTSP_LOWER_TRANS_UDP_MCAST, "UDP Multicast Mode", "udp-multicast"},
{RTSP_LOWER_TRANS_TCP, "TCP interleaved mode", "tcp"},
{0, NULL, NULL},
};
if (!rtsp_lower_trans_type) {
rtsp_lower_trans_type =
g_flags_register_static ("GstRTSPLowerTrans", rtsp_lower_trans);
}
return rtsp_lower_trans_type;
}
static void gst_rtspsrc_base_init (gpointer g_class);
static void gst_rtspsrc_finalize (GObject * object);
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
static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
gpointer iface_data);
static GstCaps *gst_rtspsrc_media_to_caps (gint pt, SDPMedia * media);
static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
GstStateChange transition);
static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
RTSPMessage * response);
static gboolean gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code);
static gboolean gst_rtspsrc_open (GstRTSPSrc * src);
static gboolean gst_rtspsrc_play (GstRTSPSrc * src);
static gboolean gst_rtspsrc_pause (GstRTSPSrc * src);
static gboolean gst_rtspsrc_close (GstRTSPSrc * src);
static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
const gchar * uri);
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
static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
static void gst_rtspsrc_loop (GstRTSPSrc * src);
/* commands we send to out loop to notify it of events */
#define CMD_WAIT 0
#define CMD_RECONNECT 1
#define CMD_STOP 2
/*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */
static void
_do_init (GType rtspsrc_type)
{
static const GInterfaceInfo urihandler_info = {
gst_rtspsrc_uri_handler_init,
NULL,
NULL
};
GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
g_type_add_interface_static (rtspsrc_type, GST_TYPE_URI_HANDLER,
&urihandler_info);
}
GST_BOILERPLATE_FULL (GstRTSPSrc, gst_rtspsrc, GstBin, GST_TYPE_BIN, _do_init);
static void
gst_rtspsrc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtptemplate));
gst_element_class_set_details (element_class, &gst_rtspsrc_details);
}
static void
gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBinClass *gstbin_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbin_class = (GstBinClass *) klass;
gobject_class->set_property = gst_rtspsrc_set_property;
gobject_class->get_property = gst_rtspsrc_get_property;
gobject_class->finalize = gst_rtspsrc_finalize;
g_object_class_install_property (gobject_class, PROP_LOCATION,
g_param_spec_string ("location", "RTSP Location",
"Location of the RTSP url to read",
DEFAULT_LOCATION, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
g_param_spec_flags ("protocols", "Protocols",
"Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
g_object_class_install_property (gobject_class, PROP_DEBUG,
g_param_spec_boolean ("debug", "Debug",
"Dump request and response messages to stdout",
DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
g_object_class_install_property (gobject_class, PROP_RETRY,
g_param_spec_uint ("retry", "Retry",
"Max number of retries when allocating RTP ports.",
0, G_MAXUINT16, DEFAULT_RETRY,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
g_object_class_install_property (gobject_class, PROP_TIMEOUT,
g_param_spec_uint64 ("timeout", "Timeout",
"Retry TCP transport after timeout microseconds (0 = disabled)",
0, G_MAXUINT64, DEFAULT_TIMEOUT,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
gstelement_class->change_state = gst_rtspsrc_change_state;
gstbin_class->handle_message = gst_rtspsrc_handle_message;
}
static void
gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
{
src->stream_rec_lock = g_new (GStaticRecMutex, 1);
g_static_rec_mutex_init (src->stream_rec_lock);
src->loop_cond = g_cond_new ();
src->location = g_strdup (DEFAULT_LOCATION);
src->url = NULL;
#ifdef WITH_EXT_REAL
src->extension = rtsp_ext_real_get_context ();
Fix a bunch of leaks shown by the newly-added states test. Original commit message from CVS: * ext/flac/gstflacenc.c: (gst_flac_enc_finalize): * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_class_init), (gst_gconf_audio_sink_dispose), (gst_gconf_audio_sink_finalize): * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init), (gst_gconf_audio_src_class_init), (gst_gconf_audio_src_dispose), (gst_gconf_audio_src_finalize), (do_toggle_element): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init), (gst_gconf_video_sink_class_init), (gst_gconf_video_sink_finalize), (do_toggle_element): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init), (gst_gconf_video_src_class_init), (gst_gconf_video_src_dispose), (gst_gconf_video_src_finalize), (do_toggle_element): * ext/gconf/gstswitchsink.c: (gst_switch_sink_class_init), (gst_switch_sink_reset), (gst_switch_sink_set_child): * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init): * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init): * ext/shout2/gstshout2.c: (gst_shout2send_class_init), (gst_shout2send_init), (gst_shout2send_finalize): * gst/debug/testplugin.c: (gst_test_class_init), (gst_test_finalize): * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (gst_flxdec_dispose): * gst/multipart/multipartmux.c: (gst_multipart_mux_finalize): * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize): * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_free_context): * gst/rtsp/rtspextwms.h: * gst/smpte/gstsmpte.c: (gst_smpte_class_init), (gst_smpte_finalize): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_finalize): * gst/udp/gstudpsink.c: (gst_udpsink_class_init), (gst_udpsink_finalize): * gst/wavparse/gstwavparse.c: (gst_wavparse_dispose), (gst_wavparse_sink_activate): * sys/oss/gstosssink.c: (gst_oss_sink_finalise): * sys/oss/gstosssrc.c: (gst_oss_src_class_init), (gst_oss_src_finalize): * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_destroy): * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init), (gst_v4l2src_finalize): * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get): Fix a bunch of leaks shown by the newly-added states test.
2007-03-04 13:52:03 +00:00
#else
/* install WMS extension by default */
src->extension = rtsp_ext_wms_get_context ();
#endif
src->extension->src = (gpointer) src;
}
static void
gst_rtspsrc_finalize (GObject * object)
{
GstRTSPSrc *rtspsrc;
rtspsrc = GST_RTSPSRC (object);
g_static_rec_mutex_free (rtspsrc->stream_rec_lock);
g_free (rtspsrc->stream_rec_lock);
g_cond_free (rtspsrc->loop_cond);
g_free (rtspsrc->location);
g_free (rtspsrc->req_location);
g_free (rtspsrc->content_base);
rtsp_url_free (rtspsrc->url);
Fix a bunch of leaks shown by the newly-added states test. Original commit message from CVS: * ext/flac/gstflacenc.c: (gst_flac_enc_finalize): * ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_class_init), (gst_gconf_audio_sink_dispose), (gst_gconf_audio_sink_finalize): * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init), (gst_gconf_audio_src_class_init), (gst_gconf_audio_src_dispose), (gst_gconf_audio_src_finalize), (do_toggle_element): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init), (gst_gconf_video_sink_class_init), (gst_gconf_video_sink_finalize), (do_toggle_element): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init), (gst_gconf_video_src_class_init), (gst_gconf_video_src_dispose), (gst_gconf_video_src_finalize), (do_toggle_element): * ext/gconf/gstswitchsink.c: (gst_switch_sink_class_init), (gst_switch_sink_reset), (gst_switch_sink_set_child): * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init): * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init): * ext/shout2/gstshout2.c: (gst_shout2send_class_init), (gst_shout2send_init), (gst_shout2send_finalize): * gst/debug/testplugin.c: (gst_test_class_init), (gst_test_finalize): * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (gst_flxdec_dispose): * gst/multipart/multipartmux.c: (gst_multipart_mux_finalize): * gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize): * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_free_context): * gst/rtsp/rtspextwms.h: * gst/smpte/gstsmpte.c: (gst_smpte_class_init), (gst_smpte_finalize): * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_finalize): * gst/udp/gstudpsink.c: (gst_udpsink_class_init), (gst_udpsink_finalize): * gst/wavparse/gstwavparse.c: (gst_wavparse_dispose), (gst_wavparse_sink_activate): * sys/oss/gstosssink.c: (gst_oss_sink_finalise): * sys/oss/gstosssrc.c: (gst_oss_src_class_init), (gst_oss_src_finalize): * sys/v4l2/gstv4l2object.c: (gst_v4l2_object_destroy): * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init), (gst_v4l2src_finalize): * sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get): Fix a bunch of leaks shown by the newly-added states test.
2007-03-04 13:52:03 +00:00
if (rtspsrc->extension) {
#ifdef WITH_EXT_REAL
rtsp_ext_real_free_context (rtspsrc->extension);
#else
rtsp_ext_wms_free_context (rtspsrc->extension);
#endif
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstRTSPSrc *rtspsrc;
rtspsrc = GST_RTSPSRC (object);
switch (prop_id) {
case PROP_LOCATION:
gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
g_value_get_string (value));
break;
case PROP_PROTOCOLS:
rtspsrc->protocols = g_value_get_flags (value);
break;
case PROP_DEBUG:
rtspsrc->debug = g_value_get_boolean (value);
break;
case PROP_RETRY:
rtspsrc->retry = g_value_get_uint (value);
break;
case PROP_TIMEOUT:
rtspsrc->timeout = g_value_get_uint64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstRTSPSrc *rtspsrc;
rtspsrc = GST_RTSPSRC (object);
switch (prop_id) {
case PROP_LOCATION:
g_value_set_string (value, rtspsrc->location);
break;
case PROP_PROTOCOLS:
g_value_set_flags (value, rtspsrc->protocols);
break;
case PROP_DEBUG:
g_value_set_boolean (value, rtspsrc->debug);
break;
case PROP_RETRY:
g_value_set_uint (value, rtspsrc->retry);
break;
case PROP_TIMEOUT:
g_value_set_uint64 (value, rtspsrc->timeout);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gint
find_stream_by_channel (GstRTSPStream * stream, gconstpointer a)
{
gint channel = GPOINTER_TO_INT (a);
if (stream->channel[0] == channel || stream->channel[1] == channel)
return 0;
return -1;
}
static gint
find_stream_by_pt (GstRTSPStream * stream, gconstpointer a)
{
gint pt = GPOINTER_TO_INT (a);
if (stream->pt == pt)
return 0;
return -1;
}
static gint
find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
{
GstElement *src = (GstElement *) a;
if (stream->udpsrc[0] == src)
return 0;
return -1;
}
static GstRTSPStream *
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
gst_rtspsrc_create_stream (GstRTSPSrc * src, SDPMessage * sdp, gint idx)
{
GstRTSPStream *stream;
gchar *control_url;
gchar *payload;
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
SDPMedia *media;
/* get media, should not return NULL */
media = sdp_message_get_media (sdp, idx);
if (media == NULL)
return NULL;
stream = g_new0 (GstRTSPStream, 1);
stream->parent = src;
/* we mark the pad as not linked, we will mark it as OK when we add the pad to
* the element. */
stream->last_ret = GST_FLOW_NOT_LINKED;
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
stream->added = FALSE;
stream->id = src->numstreams++;
/* we must have a payload. No payload means we cannot create caps */
/* FIXME, handle multiple formats. */
if ((payload = sdp_media_get_format (media, 0))) {
stream->pt = atoi (payload);
/* convert caps */
stream->caps = gst_rtspsrc_media_to_caps (stream->pt, media);
if (stream->pt >= 96) {
/* If we have a dynamic payload type, see if we have a stream with the
* same payload number. If there is one, they are part of the same
* container and we only need to add one pad. */
if (g_list_find_custom (src->streams, GINT_TO_POINTER (stream->pt),
(GCompareFunc) find_stream_by_pt)) {
stream->container = TRUE;
}
}
}
/* get control url to construct the setup url. The setup url is used to
* configure the transport of the stream and is used to identity the stream in
* the RTP-Info header field returned from PLAY. */
control_url = sdp_media_get_attribute_val (media, "control");
GST_DEBUG_OBJECT (src, "stream %d", stream->id);
GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
GST_DEBUG_OBJECT (src, " container: %d", stream->container);
GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps);
GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
if (control_url != NULL) {
/* If the control_url starts with a '/' or a non rtsp: protocol we will most
* likely build a URL that the server will fail to understand, this is ok,
* we will fail then. */
if (g_str_has_prefix (control_url, "rtsp://"))
stream->setup_url = g_strdup (control_url);
else if (src->content_base)
stream->setup_url =
g_strdup_printf ("%s%s", src->content_base, control_url);
else
stream->setup_url =
g_strdup_printf ("%s/%s", src->req_location, control_url);
}
GST_DEBUG_OBJECT (src, " setup: %s", GST_STR_NULL (stream->setup_url));
/* we keep track of all streams */
src->streams = g_list_append (src->streams, stream);
return stream;
}
static void
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
gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
{
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
gint i;
GST_DEBUG_OBJECT (src, "free stream %p", stream);
if (stream->caps)
gst_caps_unref (stream->caps);
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
g_free (stream->setup_url);
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
for (i = 0; i < 2; i++) {
GstElement *udpsrc = stream->udpsrc[i];
if (udpsrc) {
GstPad *pad;
/* unlink the pad */
pad = gst_element_get_pad (udpsrc, "src");
if (stream->channelpad[i]) {
gst_pad_unlink (pad, stream->channelpad[i]);
gst_object_unref (stream->channelpad[i]);
stream->channelpad[i] = NULL;
}
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
gst_element_set_state (udpsrc, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), udpsrc);
gst_object_unref (stream->udpsrc[i]);
stream->udpsrc[i] = NULL;
}
}
if (stream->sess) {
gst_element_set_state (stream->sess, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), stream->sess);
stream->sess = NULL;
}
if (stream->srcpad) {
gst_pad_set_active (stream->srcpad, FALSE);
if (stream->added) {
gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
stream->added = FALSE;
}
stream->srcpad = NULL;
}
g_free (stream);
}
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
static void
gst_rtspsrc_cleanup (GstRTSPSrc * src)
{
GList *walk;
GST_DEBUG_OBJECT (src, "cleanup");
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
gst_rtspsrc_stream_free (src, stream);
}
g_list_free (src->streams);
src->streams = NULL;
src->numstreams = 0;
if (src->props)
gst_structure_free (src->props);
src->props = NULL;
}
#define PARSE_INT(p, del, res) \
G_STMT_START { \
gchar *t = p; \
p = strstr (p, del); \
if (p == NULL) \
res = -1; \
else { \
*p = '\0'; \
p++; \
res = atoi (t); \
} \
} G_STMT_END
#define PARSE_STRING(p, del, res) \
G_STMT_START { \
gchar *t = p; \
p = strstr (p, del); \
if (p == NULL) \
res = NULL; \
else { \
*p = '\0'; \
p++; \
res = t; \
} \
} G_STMT_END
#define SKIP_SPACES(p) \
while (*p && g_ascii_isspace (*p)) \
p++;
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
/* rtpmap contains:
*
* <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
*/
static gboolean
gst_rtspsrc_parse_rtpmap (gchar * rtpmap, gint * payload, gchar ** name,
gint * rate, gchar ** params)
{
gchar *p, *t;
t = p = rtpmap;
PARSE_INT (p, " ", *payload);
if (*payload == -1)
return FALSE;
SKIP_SPACES (p);
if (*p == '\0')
return FALSE;
PARSE_STRING (p, "/", *name);
if (*name == NULL) {
/* no rate, assume 0 then */
*name = p;
*rate = -1;
return TRUE;
}
t = p;
p = strstr (p, "/");
if (p == NULL) {
*rate = atoi (t);
return TRUE;
}
*p = '\0';
p++;
*rate = atoi (t);
t = p;
if (*p == '\0')
return TRUE;
*params = t;
return TRUE;
}
/*
* Mapping of caps to and from SDP fields:
*
* m=<media> <UDP port> RTP/AVP <payload>
* a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
* a=fmtp:<payload> <param>[=<value>];...
*/
static GstCaps *
gst_rtspsrc_media_to_caps (gint pt, SDPMedia * media)
{
GstCaps *caps;
gchar *rtpmap;
gchar *fmtp;
gchar *name = NULL;
gint rate = -1;
gchar *params = NULL;
gchar *tmp;
GstStructure *s;
/* dynamic payloads need rtpmap */
if (pt >= 96) {
gint payload = 0;
gboolean ret;
if ((rtpmap = sdp_media_get_attribute_val (media, "rtpmap"))) {
ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, &params);
if (ret) {
if (payload != pt) {
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
/* FIXME, not fatal? */
g_warning ("rtpmap of wrong payload type");
name = NULL;
rate = -1;
params = NULL;
}
} else {
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
/* FIXME, not fatal? */
g_warning ("error parsing rtpmap");
}
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
} else
goto no_rtpmap;
}
tmp = g_ascii_strdown (media->media, -1);
caps = gst_caps_new_simple ("application/x-unknown",
"media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
g_free (tmp);
s = gst_caps_get_structure (caps, 0);
if (rate != -1)
gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
/* encoding name must be upper case */
if (name != NULL) {
tmp = g_ascii_strup (name, -1);
gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
g_free (tmp);
}
/* params must be lower case */
if (params != NULL) {
tmp = g_ascii_strdown (params, -1);
gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
g_free (tmp);
}
/* parse optional fmtp: field */
if ((fmtp = sdp_media_get_attribute_val (media, "fmtp"))) {
gchar *p;
gint payload = 0;
p = fmtp;
/* p is now of the format <payload> <param>[=<value>];... */
PARSE_INT (p, " ", payload);
if (payload != -1 && payload == pt) {
gchar **pairs;
gint i;
/* <param>[=<value>] are separated with ';' */
pairs = g_strsplit (p, ";", 0);
for (i = 0; pairs[i]; i++) {
gchar *valpos;
gchar *val, *key;
/* the key may not have a '=', the value can have other '='s */
valpos = strstr (pairs[i], "=");
if (valpos) {
/* we have a '=' and thus a value, remove the '=' with \0 */
*valpos = '\0';
/* value is everything between '=' and ';'. FIXME, strip? */
val = g_strstrip (valpos + 1);
} else {
/* simple <param>;.. is translated into <param>=1;... */
val = "1";
}
/* strip the key of spaces, convert key to lowercase but not the value. */
key = g_strstrip (pairs[i]);
tmp = g_ascii_strdown (key, -1);
gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
g_free (tmp);
}
g_strfreev (pairs);
}
}
return caps;
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
/* ERRORS */
no_rtpmap:
{
g_warning ("rtpmap type not given for dynamic payload %d", pt);
return NULL;
}
}
static gboolean
gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
gint * rtpport, gint * rtcpport)
{
GstRTSPSrc *src;
GstStateChangeReturn ret;
GstElement *tmp, *udpsrc0, *udpsrc1;
gint tmp_rtp, tmp_rtcp;
guint count;
src = stream->parent;
tmp = NULL;
udpsrc0 = NULL;
udpsrc1 = NULL;
count = 0;
/* try to allocate 2 UDP ports, the RTP port should be an even
* number and the RTCP port should be the next (uneven) port */
again:
udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL);
if (udpsrc0 == NULL)
goto no_udp_protocol;
ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_udp_failure;
g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
/* check if port is even */
if ((tmp_rtp & 0x01) != 0) {
/* port not even, close and allocate another */
count++;
if (count > src->retry)
goto no_ports;
GST_DEBUG_OBJECT (src, "RTP port not even, retry %d", count);
/* have to keep port allocated so we can get a new one */
if (tmp != NULL) {
GST_DEBUG_OBJECT (src, "free temp");
gst_element_set_state (tmp, GST_STATE_NULL);
gst_object_unref (tmp);
}
tmp = udpsrc0;
GST_DEBUG_OBJECT (src, "retry %d", count);
goto again;
}
/* free leftover temp element/port */
if (tmp) {
gst_element_set_state (tmp, GST_STATE_NULL);
gst_object_unref (tmp);
tmp = NULL;
}
/* allocate port+1 for RTCP now */
udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
if (udpsrc1 == NULL)
goto no_udp_rtcp_protocol;
/* set port */
tmp_rtcp = tmp_rtp + 1;
g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
/* FIXME, this could fail if the next port is not free, we
* should retry with another port then */
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_rtcp_failure;
/* all fine, do port check */
g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
/* this should not happen */
if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
goto port_error;
/* we keep these elements, we configure all in configure_transport when the
* server told us to really use the UDP ports. */
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
stream->udpsrc[0] = gst_object_ref (udpsrc0);
stream->udpsrc[1] = gst_object_ref (udpsrc1);
/* they are ours now */
gst_object_sink (udpsrc0);
gst_object_sink (udpsrc1);
return TRUE;
/* ERRORS */
no_udp_protocol:
{
GST_DEBUG_OBJECT (src, "could not get UDP source");
goto cleanup;
}
start_udp_failure:
{
GST_DEBUG_OBJECT (src, "could not start UDP source");
goto cleanup;
}
no_ports:
{
GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
count);
goto cleanup;
}
no_udp_rtcp_protocol:
{
GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
goto cleanup;
}
start_rtcp_failure:
{
GST_DEBUG_OBJECT (src, "could not start UDP source for RTCP");
goto cleanup;
}
port_error:
{
GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
goto cleanup;
}
cleanup:
{
if (tmp) {
gst_element_set_state (tmp, GST_STATE_NULL);
gst_object_unref (tmp);
}
if (udpsrc0) {
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
}
if (udpsrc1) {
gst_element_set_state (udpsrc1, GST_STATE_NULL);
gst_object_unref (udpsrc1);
}
return FALSE;
}
}
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
static void
pad_unblocked (GstPad * pad, gboolean blocked, GstRTSPSrc * src)
{
GST_DEBUG_OBJECT (src, "pad %s:%s unblocked", GST_DEBUG_PAD_NAME (pad));
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
}
static void
pad_blocked (GstPad * pad, gboolean blocked, GstRTSPSrc * src)
{
GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
GST_DEBUG_PAD_NAME (pad));
/* activate the streams */
GST_OBJECT_LOCK (src);
if (!src->need_activate)
goto was_ok;
src->need_activate = FALSE;
GST_OBJECT_UNLOCK (src);
gst_rtspsrc_activate_streams (src);
return;
was_ok:
{
GST_OBJECT_UNLOCK (src);
return;
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
}
}
/* sets up all elements needed for streaming over the specified transport.
* Does not yet expose the element pads, this will be done when there is actuall
* dataflow detected, which might never happen when UDP is blocked in a
* firewall, for example.
*/
static gboolean
gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
RTSPTransport * transport)
{
GstRTSPSrc *src;
GstPad *outpad = NULL;
GstPadTemplate *template;
GstStateChangeReturn ret;
gchar *name;
GstStructure *s;
const gchar *mime, *manager;
RTSPResult res;
src = stream->parent;
GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
s = gst_caps_get_structure (stream->caps, 0);
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
/* get the proper mime type for this stream now */
if ((res = rtsp_transport_get_mime (transport->trans, &mime)) < 0)
goto no_mime;
if (!mime)
goto no_mime;
/* configure the final mime type */
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
GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
gst_structure_set_name (s, mime);
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
/* find a manager */
if ((res = rtsp_transport_get_manager (transport->trans, &manager)) < 0)
goto no_manager;
if (manager) {
GST_DEBUG_OBJECT (src, "using manager %s", manager);
/* FIXME, the session manager needs to be shared with all the streams */
if (!(stream->sess = gst_element_factory_make (manager, NULL)))
goto no_element;
/* we manage this element */
gst_bin_add (GST_BIN_CAST (src), stream->sess);
ret = gst_element_set_state (stream->sess, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_session_failure;
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
/* we stream directly to the manager, FIXME, pad names should not be
* hardcoded. */
stream->channelpad[0] = gst_element_get_pad (stream->sess, "sinkrtp");
stream->channelpad[1] = gst_element_get_pad (stream->sess, "sinkrtcp");
}
if (transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
gint i;
/* configure for interleaved delivery, nothing needs to be done
* here, the loop function will call the chain functions of the
* session manager. */
stream->channel[0] = transport->interleaved.min;
stream->channel[1] = transport->interleaved.max;
GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
stream->channel[0], stream->channel[1]);
/* we can remove the allocated UDP ports now */
for (i = 0; i < 2; i++) {
if (stream->udpsrc[i]) {
gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
gst_object_unref (stream->udpsrc[i]);
stream->udpsrc[i] = NULL;
}
}
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
/* no session manager, send data to srcpad directly */
if (!stream->channelpad[0]) {
GST_DEBUG_OBJECT (src, "no manager, creating pad");
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
/* create a new pad we will use to stream to */
name = g_strdup_printf ("stream%d", stream->id);
template = gst_static_pad_template_get (&rtptemplate);
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
stream->channelpad[0] = gst_pad_new_from_template (template, name);
gst_object_unref (template);
g_free (name);
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
/* set caps and activate */
gst_pad_use_fixed_caps (stream->channelpad[0]);
gst_pad_set_caps (stream->channelpad[0], stream->caps);
gst_pad_set_active (stream->channelpad[0], TRUE);
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
outpad = gst_object_ref (stream->channelpad[0]);
} else {
GST_DEBUG_OBJECT (src, "using manager source pad");
outpad = gst_element_get_pad (stream->sess, "srcrtp");
}
} else {
/* multicast was selected, create UDP sources and join the multicast
* group. */
if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST) {
gchar *uri;
GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
/* creating UDP source */
if (transport->port.min != -1) {
uri = g_strdup_printf ("udp://%s:%d", transport->destination,
transport->port.min);
stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
g_free (uri);
if (stream->udpsrc[0] == NULL)
goto no_element;
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
/* take ownership */
gst_object_ref (stream->udpsrc[0]);
gst_object_sink (stream->udpsrc[0]);
/* change state */
gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
}
/* creating another UDP source */
if (transport->port.max != -1) {
uri = g_strdup_printf ("udp://%s:%d", transport->destination,
transport->port.max);
stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
g_free (uri);
if (stream->udpsrc[1] == NULL)
goto no_element;
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
/* take ownership */
gst_object_ref (stream->udpsrc[0]);
gst_object_sink (stream->udpsrc[0]);
gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
}
}
/* we manage the UDP elements now. For unicast, the UDP sources where
* allocated in the stream when we suggested a transport. */
if (stream->udpsrc[0]) {
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
GST_DEBUG_OBJECT (src, "setting up UDP source");
/* set caps */
g_object_set (G_OBJECT (stream->udpsrc[0]), "caps", stream->caps, NULL);
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
/* configure a timeout on the UDP port. When the timeout message is
* posted, we assume UDP transport is not possible. We reconnect using TCP
* if we can. */
g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->timeout,
NULL);
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
/* get output pad of the UDP source. */
outpad = gst_element_get_pad (stream->udpsrc[0], "src");
/* save it so we can unblock */
stream->blockedpad = outpad;
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
/* configure pad block on the pad. As soon as there is dataflow on the
* UDP source, we know that UDP is not blocked by a firewall and we can
* configure all the streams to let the application autoplug decoders. */
gst_pad_set_blocked_async (outpad, TRUE,
(GstPadBlockCallback) pad_blocked, src);
if (stream->channelpad[0]) {
GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
/* configure for UDP delivery, we need to connect the UDP pads to
* the session plugin. */
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
gst_pad_link (outpad, stream->channelpad[0]);
gst_object_unref (outpad);
/* the real output pad is that of the session manager */
outpad = gst_element_get_pad (stream->sess, "srcrtp");
} else {
GST_DEBUG_OBJECT (src, "using UDP src pad as output");
}
}
if (stream->udpsrc[1]) {
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
if (stream->channelpad[1]) {
GstPad *pad;
GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
pad = gst_element_get_pad (stream->udpsrc[1], "src");
gst_pad_link (pad, stream->channelpad[1]);
gst_object_unref (pad);
}
}
}
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
if (outpad) {
GST_DEBUG_OBJECT (src, "creating ghostpad");
gst_pad_use_fixed_caps (outpad);
gst_pad_set_caps (outpad, stream->caps);
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
/* create ghostpad, don't add just yet, this will be done when we activate
* the stream. */
name = g_strdup_printf ("stream%d", stream->id);
template = gst_static_pad_template_get (&rtptemplate);
stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
gst_object_unref (template);
g_free (name);
gst_object_unref (outpad);
}
/* mark pad as ok */
stream->last_ret = GST_FLOW_OK;
return TRUE;
/* ERRORS */
no_mime:
{
GST_DEBUG_OBJECT (src, "unknown transport");
return FALSE;
}
no_manager:
{
GST_DEBUG_OBJECT (src, "cannot get a session manager");
return FALSE;
}
no_element:
{
GST_DEBUG_OBJECT (src, "no rtpdec element found");
return FALSE;
}
start_session_failure:
{
GST_DEBUG_OBJECT (src, "could not start session");
return FALSE;
}
}
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
/* Adds the source pads of all configured streams to the element.
* This code is performed when we detected dataflow.
*
* We detect dataflow from either the _loop function or with pad probes on the
* udp sources.
*/
static gboolean
gst_rtspsrc_activate_streams (GstRTSPSrc * src)
{
GList *walk;
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
if (stream->srcpad) {
gst_pad_set_active (stream->srcpad, TRUE);
/* add the pad */
if (!stream->added) {
gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
stream->added = TRUE;
}
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
}
}
/* if we got here all was configured. We have dynamic pads so we notify that
* we are done */
gst_element_no_more_pads (GST_ELEMENT_CAST (src));
/* unblock all pads */
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
if (stream->blockedpad) {
gst_pad_set_blocked_async (stream->blockedpad, FALSE,
(GstPadBlockCallback) pad_unblocked, src);
stream->blockedpad = NULL;
}
}
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
return TRUE;
}
static GstFlowReturn
gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
GstFlowReturn ret)
{
GList *streams;
/* store the value */
stream->last_ret = ret;
/* if it's success we can return the value right away */
if (GST_FLOW_IS_SUCCESS (ret))
goto done;
/* any other error that is not-linked can be returned right
* away */
if (ret != GST_FLOW_NOT_LINKED)
goto done;
/* only return NOT_LINKED if all other pads returned NOT_LINKED */
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
ret = ostream->last_ret;
/* some other return value (must be SUCCESS but we can return
* other values as well) */
if (ret != GST_FLOW_NOT_LINKED)
goto done;
}
/* if we get here, all other pads were unlinked and we return
* NOT_LINKED then */
done:
return ret;
}
static void
gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
{
GList *streams;
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
/* only pads that have a connection to the outside world */
if (ostream->srcpad == NULL)
continue;
if (ostream->channelpad[0]) {
gst_event_ref (event);
if (GST_PAD_IS_SRC (ostream->channelpad[0]))
gst_pad_push_event (ostream->channelpad[0], event);
else
gst_pad_send_event (ostream->channelpad[0], event);
}
if (ostream->channelpad[1]) {
gst_event_ref (event);
if (GST_PAD_IS_SRC (ostream->channelpad[1]))
gst_pad_push_event (ostream->channelpad[1], event);
else
gst_pad_send_event (ostream->channelpad[1], event);
}
}
gst_event_unref (event);
}
static void
gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
{
RTSPMessage response = { 0 };
RTSPResult res;
gint channel;
GList *lstream;
GstRTSPStream *stream;
GstPad *outpad = NULL;
guint8 *data;
guint size;
GstFlowReturn ret = GST_FLOW_OK;
GstCaps *caps = NULL;
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
GstBuffer *buf;
do {
GST_DEBUG_OBJECT (src, "doing receive");
if ((res = rtsp_connection_receive (src->connection, &response)) < 0)
goto receive_error;
GST_DEBUG_OBJECT (src, "got packet type %d", response.type);
}
while (response.type != RTSP_MESSAGE_DATA);
channel = response.type_data.data.channel;
lstream = g_list_find_custom (src->streams, GINT_TO_POINTER (channel),
(GCompareFunc) find_stream_by_channel);
if (!lstream)
goto unknown_stream;
stream = (GstRTSPStream *) lstream->data;
if (channel == stream->channel[0]) {
outpad = stream->channelpad[0];
caps = stream->caps;
} else if (channel == stream->channel[1]) {
outpad = stream->channelpad[1];
}
/* take a look at the body to figure out what we have */
rtsp_message_get_body (&response, &data, &size);
if (size < 2)
goto invalid_length;
/* channels are not correct on some servers, do extra check */
if (data[1] >= 200 && data[1] <= 204) {
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
/* hmm RTCP message switch to the RTCP pad of the same stream. */
outpad = stream->channelpad[1];
}
/* we have no clue what this is, just ignore then. */
if (outpad == NULL)
goto unknown_stream;
/* and chain buffer to internal element */
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
rtsp_message_steal_body (&response, &data, &size);
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
/* strip the trailing \0 */
size -= 1;
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
buf = gst_buffer_new ();
GST_BUFFER_DATA (buf) = data;
GST_BUFFER_MALLOCDATA (buf) = data;
GST_BUFFER_SIZE (buf) = size;
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
/* don't need message anymore */
rtsp_message_unset (&response);
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
if (caps)
gst_buffer_set_caps (buf, caps);
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
GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
channel);
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
if (src->need_activate) {
gst_rtspsrc_activate_streams (src);
src->need_activate = FALSE;
}
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
/* chain to the peer pad */
if (GST_PAD_IS_SINK (outpad))
ret = gst_pad_chain (outpad, buf);
else
ret = gst_pad_push (outpad, buf);
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
/* combine all stream flows */
ret = gst_rtspsrc_combine_flows (src, stream, ret);
if (ret != GST_FLOW_OK)
goto need_pause;
return;
/* ERRORS */
unknown_stream:
{
GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
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
rtsp_message_unset (&response);
return;
}
receive_error:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
g_free (str);
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
if (src->debug)
rtsp_message_dump (&response);
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
rtsp_message_unset (&response);
ret = GST_FLOW_UNEXPECTED;
goto need_pause;
}
invalid_length:
{
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("Short message received."));
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
rtsp_message_unset (&response);
return;
}
need_pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
src->running = FALSE;
gst_task_pause (src->task);
if (GST_FLOW_IS_FATAL (ret) || ret == GST_FLOW_NOT_LINKED) {
if (ret == GST_FLOW_UNEXPECTED) {
/* perform EOS logic */
if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (src),
gst_message_new_segment_done (GST_OBJECT_CAST (src),
src->segment.format, src->segment.last_stop));
} else {
gst_rtspsrc_push_event (src, gst_event_new_eos ());
}
} else {
/* for fatal errors we post an error message, post the error
* first so the app knows about the error first. */
GST_ELEMENT_ERROR (src, STREAM, FAILED,
("Internal data flow error."),
("streaming task paused, reason %s (%d)", reason, ret));
gst_rtspsrc_push_event (src, gst_event_new_eos ());
}
}
return;
}
}
static void
gst_rtspsrc_loop_udp (GstRTSPSrc * src)
{
gboolean restart = FALSE;
GST_OBJECT_LOCK (src);
if (src->loop_cmd == CMD_STOP)
goto stopping;
while (src->loop_cmd == CMD_WAIT) {
GST_DEBUG_OBJECT (src, "waiting");
GST_RTSP_LOOP_WAIT (src);
GST_DEBUG_OBJECT (src, "waiting done");
if (src->loop_cmd == CMD_STOP)
goto stopping;
}
if (src->loop_cmd == CMD_RECONNECT) {
/* FIXME, when we get here we have to reconnect using tcp */
src->loop_cmd = CMD_WAIT;
/* only restart when the pads were not yet activated, else we were
* streaming over UDP */
restart = src->need_activate;
}
GST_OBJECT_UNLOCK (src);
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
/* no need to restart, we're done */
if (!restart)
goto done;
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
/* We post a warning message now to inform the user
* that nothing happened. It's most likely a firewall thing. */
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("Could not receive any UDP packets for %.4f seconds, maybe your "
"firewall is blocking it. Retrying using a TCP connection.",
gst_guint64_to_gdouble (src->timeout / 1000000)));
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
/* we can try only TCP now */
src->cur_protocols = RTSP_LOWER_TRANS_TCP;
/* pause to prepare for a restart */
gst_rtspsrc_pause (src);
if (src->task) {
/* stop task, we cannot join as this would deadlock */
gst_task_stop (src->task);
/* and free the task so that _close will not stop/join it again. */
gst_object_unref (GST_OBJECT (src->task));
src->task = NULL;
}
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
/* close and cleanup our state */
gst_rtspsrc_close (src);
/* see if we have TCP left to try */
if (!(src->cur_protocols & RTSP_LOWER_TRANS_TCP))
goto no_protocols;
/* open new connection using tcp */
if (!gst_rtspsrc_open (src))
goto open_failed;
/* start playback */
if (!gst_rtspsrc_play (src))
goto play_failed;
done:
return;
/* ERRORS */
stopping:
{
GST_OBJECT_UNLOCK (src);
src->running = FALSE;
gst_task_pause (src->task);
return;
}
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
no_protocols:
{
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
src->cur_protocols = 0;
/* no transport possible, post an error and stop */
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
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
("Could not connect to server, no protocols left"));
return;
}
open_failed:
{
GST_DEBUG_OBJECT (src, "open failed");
return;
}
play_failed:
{
GST_DEBUG_OBJECT (src, "play failed");
return;
}
}
static void
gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd)
{
GST_OBJECT_LOCK (src);
src->loop_cmd = cmd;
GST_RTSP_LOOP_SIGNAL (src);
GST_OBJECT_UNLOCK (src);
}
static void
gst_rtspsrc_loop (GstRTSPSrc * src)
{
if (src->interleaved)
gst_rtspsrc_loop_interleaved (src);
else
gst_rtspsrc_loop_udp (src);
}
static RTSPResult
gst_rtspsrc_handle_request (GstRTSPSrc * src, RTSPMessage * request)
{
RTSPMessage response = { 0 };
RTSPResult res;
res = rtsp_message_init_response (&response, RTSP_STS_OK, "OK", request);
if (res < 0)
goto send_error;
if (src->debug)
rtsp_message_dump (&response);
if ((res = rtsp_connection_send (src->connection, &response)) < 0)
goto send_error;
return RTSP_OK;
/* ERRORS */
send_error:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
g_free (str);
return res;
}
}
#ifndef GST_DISABLE_GST_DEBUG
const gchar *
rtsp_auth_method_to_string (RTSPAuthMethod method)
{
gint index = 0;
while (method != 0) {
index++;
method >>= 1;
}
switch (index) {
case 0:
return "None";
case 1:
return "Basic";
case 2:
return "Digest";
}
return "Unknown";
}
#endif
/* Parse a WWW-Authenticate Response header and determine the
* available authentication methods
* FIXME: To implement digest or other auth types, we should extract
* the authentication tokens that the server provided for each method
* into an array of structures and give those to the connection object.
*
* This code should also cope with the fact that each WWW-Authenticate
* header can contain multiple challenge methods + tokens
*
* At the moment, we just do a minimal check for Basic auth and don't
* even parse out the realm */
static void
gst_rtspsrc_parse_auth_hdr (gchar * hdr, RTSPAuthMethod * methods)
{
gchar *start;
g_return_if_fail (hdr != NULL);
g_return_if_fail (methods != NULL);
/* Skip whitespace at the start of the string */
for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
if (g_ascii_strncasecmp (start, "basic", 5) == 0)
*methods |= RTSP_AUTH_BASIC;
}
/**
* gst_rtspsrc_setup_auth:
* @src: the rtsp source
*
* Configure a username and password and auth method on the
* connection object based on a response we received from the
* peer.
*
* Currently, this requires that a username and password were supplied
* in the uri. In the future, they may be requested on demand by sending
* a message up the bus.
*
* Returns: TRUE if authentication information could be set up correctly.
*/
static gboolean
gst_rtspsrc_setup_auth (GstRTSPSrc * src, RTSPMessage * response)
{
gchar *user = NULL;
gchar *pass = NULL;
RTSPAuthMethod avail_methods = RTSP_AUTH_NONE;
RTSPAuthMethod method;
RTSPResult auth_result;
gchar *hdr;
/* Identify the available auth methods and see if any are supported */
if (rtsp_message_get_header (response, RTSP_HDR_WWW_AUTHENTICATE, &hdr) ==
RTSP_OK) {
gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods);
}
if (avail_methods == RTSP_AUTH_NONE)
goto no_auth_available;
/* FIXME: For digest auth, if the response indicates that the session
* data are stale, we just update them in the connection object and
* return TRUE to retry the request */
/* Do we have username and password available? */
if (src->url != NULL && !src->tried_url_auth) {
user = src->url->user;
pass = src->url->passwd;
src->tried_url_auth = TRUE;
GST_DEBUG_OBJECT (src,
"Attempting authentication using credentials from the URL");
}
/* FIXME: If the url didn't contain username and password or we tried them
* already, request a username and passwd from the application via some kind
* of credentials request message */
/* If we don't have a username and passwd at this point, bail out. */
if (user == NULL || pass == NULL)
goto no_user_pass;
/* Try to configure for each available authentication method, strongest to
* weakest */
for (method = RTSP_AUTH_MAX; method != RTSP_AUTH_NONE; method >>= 1) {
/* Check if this method is available on the server */
if ((method & avail_methods) == 0)
continue;
/* Pass the credentials to the connection to try on the next request */
auth_result =
rtsp_connection_set_auth (src->connection, method, user, pass);
/* INVAL indicates an invalid username/passwd were supplied, so we'll just
* ignore it and end up retrying later */
if (auth_result == RTSP_OK || auth_result == RTSP_EINVAL) {
GST_DEBUG_OBJECT (src, "Attempting %s authentication",
rtsp_auth_method_to_string (method));
break;
}
}
if (method == RTSP_AUTH_NONE)
goto no_auth_available;
return TRUE;
no_auth_available:
/* Output an error indicating that we couldn't connect because there were
* no supported authentication protocols */
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("No supported authentication protocol was found"));
return FALSE;
no_user_pass:
/* We don't fire an error message, we just return FALSE and let the
* normal NOT_AUTHORIZED error be propagated */
return FALSE;
}
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
/**
* gst_rtspsrc_send:
* @src: the rtsp source
* @request: must point to a valid request
* @response: must point to an empty #RTSPMessage
*
* send @request and retrieve the response in @response. optionally @code can be
* non-NULL in which case it will contain the status code of the response.
*
* If This function returns TRUE, @response will contain a valid response
* message that should be cleaned with rtsp_message_unset() after usage.
*
* If @code is NULL, this function will return FALSE (with an invalid @response
* message) if the response code was not 200 (OK).
*
* If the attempt results in an authentication failure, then this will attempt
* to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
* the request.
*
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
* Returns: TRUE if the processing was successful.
*/
gboolean
gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code)
{
RTSPStatusCode int_code = RTSP_STS_OK;
gboolean res;
gboolean retry;
do {
retry = FALSE;
res = gst_rtspsrc_try_send (src, request, response, &int_code);
if (int_code == RTSP_STS_UNAUTHORIZED) {
if (gst_rtspsrc_setup_auth (src, response)) {
/* Try the request/response again after configuring the auth info
* and loop again */
retry = TRUE;
}
}
} while (retry == TRUE);
/* If the user requested the code, let them handle errors, otherwise
* post an error below */
if (code != NULL)
*code = int_code;
else if (int_code != RTSP_STS_OK)
goto error_response;
return res;
error_response:
{
switch (response->type_data.response.code) {
case RTSP_STS_NOT_FOUND:
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
response->type_data.response.reason));
break;
default:
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Got error response: %d (%s).", response->type_data.response.code,
response->type_data.response.reason));
break;
}
/* we return FALSE so we should unset the response ourselves */
rtsp_message_unset (response);
return FALSE;
}
}
static gboolean
gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code)
{
RTSPResult res;
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
RTSPStatusCode thecode;
gchar *content_base = NULL;
if (src->extension && src->extension->before_send)
src->extension->before_send (src->extension, request);
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
if (src->debug)
rtsp_message_dump (request);
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
if ((res = rtsp_connection_send (src->connection, request)) < 0)
goto send_error;
next:
if ((res = rtsp_connection_receive (src->connection, response)) < 0)
goto receive_error;
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
if (src->debug)
rtsp_message_dump (response);
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.
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switch (response->type) {
case RTSP_MESSAGE_REQUEST:
/* FIXME, handle server request, reply with OK, for now */
if ((res = gst_rtspsrc_handle_request (src, response)) < 0)
goto handle_request_failed;
goto next;
case RTSP_MESSAGE_RESPONSE:
/* ok, a response is good */
break;
default:
case RTSP_MESSAGE_DATA:
/* get next response */
goto next;
}
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
thecode = response->type_data.response.code;
/* if the caller wanted the result code, we store it. */
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
if (code)
*code = thecode;
/* If the request didn't succeed, bail out before doing any more */
if (thecode != RTSP_STS_OK)
return FALSE;
/* store new content base if any */
rtsp_message_get_header (response, RTSP_HDR_CONTENT_BASE, &content_base);
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
g_free (src->content_base);
src->content_base = g_strdup (content_base);
if (src->extension && src->extension->after_send)
src->extension->after_send (src->extension, request, response);
return TRUE;
/* ERRORS */
send_error:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
g_free (str);
return FALSE;
}
receive_error:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
g_free (str);
return FALSE;
}
handle_request_failed:
{
/* ERROR was posted */
return FALSE;
}
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.
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}
/* parse the response and collect all the supported methods. We need this
* information so that we don't try to send an unsupported request to the
* server.
*/
static gboolean
gst_rtspsrc_parse_methods (GstRTSPSrc * src, RTSPMessage * response)
{
gchar *respoptions = NULL;
gchar **options;
gint i;
/* clear supported methods */
src->methods = 0;
/* Try Allow Header first */
rtsp_message_get_header (response, RTSP_HDR_ALLOW, &respoptions);
if (!respoptions)
/* Then maybe Public Header... */
rtsp_message_get_header (response, RTSP_HDR_PUBLIC, &respoptions);
if (!respoptions) {
/* this field is not required, assume the server supports
* DESCRIBE, SETUP and PLAY */
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
GST_DEBUG_OBJECT (src, "could not get OPTIONS");
src->methods = RTSP_DESCRIBE | RTSP_SETUP | RTSP_PLAY | RTSP_PAUSE;
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
goto done;
}
/* If we get here, the server gave a list of supported methods, parse
* them here. The string is like:
*
* OPTIONS, DESCRIBE, ANNOUNCE, PLAY, SETUP, ...
*/
options = g_strsplit (respoptions, ",", 0);
for (i = 0; options[i]; i++) {
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
gchar *stripped;
gint method;
stripped = g_strstrip (options[i]);
method = rtsp_find_method (stripped);
/* keep bitfield of supported methods */
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
if (method != RTSP_INVALID)
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
src->methods |= method;
}
g_strfreev (options);
/* we need describe and setup */
if (!(src->methods & RTSP_DESCRIBE))
goto no_describe;
if (!(src->methods & RTSP_SETUP))
goto no_setup;
done:
return TRUE;
/* ERRORS */
no_describe:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Server does not support DESCRIBE."));
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
return FALSE;
}
no_setup:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Server does not support SETUP."));
return FALSE;
}
}
static RTSPResult
gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
RTSPLowerTrans protocols, gchar ** transports)
{
gchar *result;
RTSPResult res;
*transports = NULL;
if (src->extension && src->extension->get_transports)
if ((res =
src->extension->get_transports (src->extension, protocols,
transports)) < 0)
goto failed;
/* extension listed transports, use those */
if (*transports != NULL)
return RTSP_OK;
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
/* the default RTSP transports */
result = g_strdup ("");
if (protocols & RTSP_LOWER_TRANS_UDP) {
gchar *new;
GST_DEBUG_OBJECT (src, "adding UDP unicast");
new =
g_strconcat (result, "RTP/AVP/UDP;unicast;client_port=%%u1-%%u2", NULL);
g_free (result);
result = new;
}
if (protocols & RTSP_LOWER_TRANS_UDP_MCAST) {
gchar *new;
GST_DEBUG_OBJECT (src, "adding UDP multicast");
/* we don't have to allocate any UDP ports yet, if the selected transport
* turns out to be multicast we can create them and join the multicast
* group indicated in the transport reply */
new = g_strconcat (result, result[0] ? "," : "",
"RTP/AVP/UDP;multicast", NULL);
g_free (result);
result = new;
}
if (protocols & RTSP_LOWER_TRANS_TCP) {
gchar *new;
GST_DEBUG_OBJECT (src, "adding TCP");
new = g_strconcat (result, result[0] ? "," : "",
"RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2", NULL);
g_free (result);
result = new;
}
*transports = result;
return RTSP_OK;
/* ERRORS */
failed:
{
return res;
}
}
static RTSPResult
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
gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports)
{
GstRTSPSrc *src;
gint nr_udp, nr_int;
gchar *next, *p;
gint rtpport = 0, rtcpport = 0;
GString *str;
src = stream->parent;
/* find number of placeholders first */
if (strstr (*transports, "%%i2"))
nr_int = 2;
else if (strstr (*transports, "%%i1"))
nr_int = 1;
else
nr_int = 0;
if (strstr (*transports, "%%u2"))
nr_udp = 2;
else if (strstr (*transports, "%%u1"))
nr_udp = 1;
else
nr_udp = 0;
if (nr_udp == 0 && nr_int == 0)
goto done;
if (nr_udp > 0) {
if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
goto failed;
}
str = g_string_new ("");
p = *transports;
while ((next = strstr (p, "%%"))) {
g_string_append_len (str, p, next - p);
if (next[2] == 'u') {
if (next[3] == '1')
g_string_append_printf (str, "%d", rtpport);
else if (next[3] == '2')
g_string_append_printf (str, "%d", rtcpport);
}
if (next[2] == 'i') {
if (next[3] == '1')
g_string_append_printf (str, "%d", src->free_channel);
else if (next[3] == '2')
g_string_append_printf (str, "%d", src->free_channel + 1);
}
p = next + 4;
}
g_free (*transports);
*transports = g_string_free (str, FALSE);
done:
return RTSP_OK;
/* ERRORS */
failed:
{
return RTSP_ERROR;
}
}
static gboolean
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
gst_rtspsrc_setup_streams (GstRTSPSrc * src)
{
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
GList *walk;
RTSPResult res;
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
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
GstRTSPStream *stream = NULL;
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
RTSPLowerTrans protocols;
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
/* we initially allow all configured lower transports. based on the URL
* transports and the replies from the server we narrow them down. */
protocols = src->url->transports & src->cur_protocols;
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
/* reset some state */
src->free_channel = 0;
src->interleaved = FALSE;
for (walk = src->streams; walk; walk = g_list_next (walk)) {
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
gchar *transports;
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
stream = (GstRTSPStream *) walk->data;
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
/* see if we need to configure this stream */
if (src->extension && src->extension->configure_stream) {
if (!src->extension->configure_stream (src->extension, stream)) {
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
GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
stream);
continue;
}
}
/* merge/overwrite global caps */
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
if (stream->caps) {
guint j, num;
GstStructure *s;
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
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
s = gst_caps_get_structure (stream->caps, 0);
num = gst_structure_n_fields (src->props);
for (j = 0; j < num; j++) {
const gchar *name;
const GValue *val;
name = gst_structure_nth_field_name (src->props, j);
val = gst_structure_get_value (src->props, name);
gst_structure_set_value (s, name, val);
GST_DEBUG_OBJECT (src, "copied %s", name);
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
}
}
/* skip setup if we have no URL for it */
if (stream->setup_url == NULL) {
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
GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
continue;
}
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
GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
stream->setup_url);
/* create a string with all the transports */
res = gst_rtspsrc_create_transports_string (src, protocols, &transports);
if (res < 0)
goto setup_transport_failed;
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
/* replace placeholders with real values, this function will optionally
* allocate UDP ports and other info needed to execute the setup request */
res = gst_rtspsrc_prepare_transports (stream, &transports);
if (res < 0)
goto setup_transport_failed;
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
/* create SETUP request */
res = rtsp_message_init_request (&request, RTSP_SETUP, stream->setup_url);
if (res < 0)
goto create_request_failed;
/* select transport, copy is made when adding to header so we can free it. */
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
rtsp_message_add_header (&request, RTSP_HDR_TRANSPORT, transports);
g_free (transports);
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
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
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
/* parse response transport */
{
gchar *resptrans = NULL;
RTSPTransport transport = { 0 };
rtsp_message_get_header (&response, RTSP_HDR_TRANSPORT, &resptrans);
if (!resptrans)
goto no_transport;
/* parse transport */
if (rtsp_transport_parse (resptrans, &transport) != RTSP_OK)
continue;
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
/* update allowed transports for other streams. once the transport of
* one stream has been determined, we make sure that all other streams
* are configured in the same way */
switch (transport.lower_transport) {
case RTSP_LOWER_TRANS_TCP:
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
GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
protocols = RTSP_LOWER_TRANS_TCP;
src->interleaved = TRUE;
/* update free channels */
src->free_channel =
MAX (transport.interleaved.min, src->free_channel);
src->free_channel =
MAX (transport.interleaved.max, src->free_channel);
src->free_channel++;
break;
case RTSP_LOWER_TRANS_UDP_MCAST:
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
/* only allow multicast for other streams */
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
GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
protocols = RTSP_LOWER_TRANS_UDP_MCAST;
break;
case RTSP_LOWER_TRANS_UDP:
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
/* only allow unicast for other streams */
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
GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
protocols = RTSP_LOWER_TRANS_UDP;
break;
default:
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
GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
transport.lower_transport);
break;
}
if (!stream->container || !src->interleaved) {
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
/* now configure the stream with the selected transport */
if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
GST_DEBUG_OBJECT (src,
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
"could not configure stream %p transport, skipping stream",
stream);
}
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
}
/* clean up our transport struct */
rtsp_transport_init (&transport);
}
}
if (src->extension && src->extension->stream_select)
src->extension->stream_select (src->extension);
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
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
/* we need to activate the streams when we detect activity */
src->need_activate = TRUE;
return TRUE;
/* ERRORS */
create_request_failed:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
goto cleanup_error;
}
setup_transport_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Could not setup transport."));
goto cleanup_error;
}
send_error:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
g_free (str);
goto cleanup_error;
}
no_transport:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server did not select transport."));
goto cleanup_error;
}
cleanup_error:
{
rtsp_message_unset (&request);
rtsp_message_unset (&response);
return FALSE;
}
}
static gboolean
gst_rtspsrc_open (GstRTSPSrc * src)
{
RTSPResult res;
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
guint8 *data;
guint size;
gint i, n_streams;
SDPMessage sdp = { 0 };
GstRTSPStream *stream = NULL;
gchar *respcont = NULL;
/* reset our state */
gst_segment_init (&src->segment, GST_FORMAT_TIME);
/* can't continue without a valid url */
if (G_UNLIKELY (src->url == NULL))
goto no_url;
src->tried_url_auth = FALSE;
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
/* create connection */
GST_DEBUG_OBJECT (src, "creating connection (%s)...", src->req_location);
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
if ((res = rtsp_connection_create (src->url, &src->connection)) < 0)
goto could_not_create;
/* connect */
GST_DEBUG_OBJECT (src, "connecting (%s)...", src->req_location);
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
if ((res = rtsp_connection_connect (src->connection)) < 0)
goto could_not_connect;
/* create OPTIONS */
GST_DEBUG_OBJECT (src, "create options...");
res = rtsp_message_init_request (&request, RTSP_OPTIONS, src->req_location);
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
if (res < 0)
goto create_request_failed;
/* send OPTIONS */
GST_DEBUG_OBJECT (src, "send options...");
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
/* parse OPTIONS */
if (!gst_rtspsrc_parse_methods (src, &response))
goto methods_error;
/* create DESCRIBE */
GST_DEBUG_OBJECT (src, "create describe...");
res = rtsp_message_init_request (&request, RTSP_DESCRIBE, src->req_location);
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
if (res < 0)
goto create_request_failed;
/* we only accept SDP for now */
rtsp_message_add_header (&request, RTSP_HDR_ACCEPT, "application/sdp");
/* prepare global stream caps properties */
if (src->props)
gst_structure_remove_all_fields (src->props);
else
src->props = gst_structure_empty_new ("RTSP Properties");
/* send DESCRIBE */
GST_DEBUG_OBJECT (src, "send describe...");
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
/* check if reply is SDP */
rtsp_message_get_header (&response, RTSP_HDR_CONTENT_TYPE, &respcont);
/* could not be set but since the request returned OK, we assume it
* was SDP, else check it. */
if (respcont) {
if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
goto wrong_content_type;
}
/* get message body and parse as SDP */
rtsp_message_get_body (&response, &data, &size);
GST_DEBUG_OBJECT (src, "parse SDP...");
sdp_message_init (&sdp);
sdp_message_parse_buffer (data, size, &sdp);
if (src->debug)
sdp_message_dump (&sdp);
if (src->extension && src->extension->parse_sdp)
src->extension->parse_sdp (src->extension, &sdp);
/* create streams */
n_streams = sdp_message_medias_len (&sdp);
for (i = 0; i < n_streams; i++) {
stream = gst_rtspsrc_create_stream (src, &sdp, i);
}
/* setup streams */
gst_rtspsrc_setup_streams (src);
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
/* clean up any messages */
rtsp_message_unset (&request);
rtsp_message_unset (&response);
return TRUE;
/* ERRORS */
no_url:
{
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
("No valid RTSP URL was provided"));
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
goto cleanup_error;
}
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
could_not_create:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
("Could not create connection. (%s)", str));
g_free (str);
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
goto cleanup_error;
}
could_not_connect:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
("Could not connect to server. (%s)", str));
g_free (str);
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
goto cleanup_error;
}
create_request_failed:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
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
goto cleanup_error;
}
send_error:
{
/* Don't post a message - the rtsp_send method will have
* taken care of it because we passed NULL for the response code */
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
goto cleanup_error;
}
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
methods_error:
{
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
/* error was posted */
goto cleanup_error;
}
wrong_content_type:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server does not support SDP, got %s.", respcont));
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
goto cleanup_error;
}
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
cleanup_error:
{
rtsp_message_unset (&request);
rtsp_message_unset (&response);
return FALSE;
}
}
static gboolean
gst_rtspsrc_close (GstRTSPSrc * src)
{
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
RTSPResult res;
GST_DEBUG_OBJECT (src, "TEARDOWN...");
gst_rtspsrc_loop_send_cmd (src, CMD_STOP);
/* stop task if any */
if (src->task) {
gst_task_stop (src->task);
/* make sure it is not running */
g_static_rec_mutex_lock (src->stream_rec_lock);
g_static_rec_mutex_unlock (src->stream_rec_lock);
/* no wait for the task to finish */
gst_task_join (src->task);
/* and free the task */
gst_object_unref (GST_OBJECT (src->task));
src->task = NULL;
}
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
if (src->methods & RTSP_PLAY) {
/* do TEARDOWN */
res =
rtsp_message_init_request (&request, RTSP_TEARDOWN, src->req_location);
if (res < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
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
/* FIXME, parse result? */
rtsp_message_unset (&request);
rtsp_message_unset (&response);
}
/* close connection */
GST_DEBUG_OBJECT (src, "closing connection...");
if ((res = rtsp_connection_close (src->connection)) < 0)
goto close_failed;
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
/* free connection */
rtsp_connection_free (src->connection);
src->connection = NULL;
/* cleanup */
gst_rtspsrc_cleanup (src);
return TRUE;
/* ERRORS */
create_request_failed:
{
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request."));
return FALSE;
}
send_error:
{
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
rtsp_message_unset (&request);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message."));
return FALSE;
}
close_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, CLOSE, (NULL), ("Close failed."));
return FALSE;
}
}
/* RTP-Info is of the format:
*
* url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
*/
static gboolean
gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
{
gchar **infos;
gint i;
infos = g_strsplit (rtpinfo, ",", 0);
for (i = 0; infos[i]; i++) {
/* FIXME, do something here:
* parse url, find stream for url.
* parse seq and rtptime. The seq number should be configured in the rtp
* depayloader or session manager to detect gaps. Same for the rtptime, it
* should be used to create an initial time newsegment.
*/
}
g_strfreev (infos);
return TRUE;
}
static gboolean
gst_rtspsrc_play (GstRTSPSrc * src)
{
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
RTSPResult res;
gchar *rtpinfo;
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
if (!(src->methods & RTSP_PLAY))
return TRUE;
GST_DEBUG_OBJECT (src, "PLAY...");
/* do play */
res = rtsp_message_init_request (&request, RTSP_PLAY, src->req_location);
if (res < 0)
goto create_request_failed;
rtsp_message_add_header (&request, RTSP_HDR_RANGE, "npt=0-");
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
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
rtsp_message_unset (&request);
/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
* for the RTP packets. If this is not present, we assume all starts from 0...
* FIXME, this is info for the RTP session manager ideally. */
rtsp_message_get_header (&response, RTSP_HDR_RTP_INFO, &rtpinfo);
if (rtpinfo)
gst_rtspsrc_parse_rtpinfo (src, rtpinfo);
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
rtsp_message_unset (&response);
/* for interleaved transport, we receive the data on the RTSP connection
* instead of UDP. We start a task to select and read from that connection.
* For UDP we start the task as well to look for server info and UDP timeouts. */
if (src->task == NULL) {
src->task = gst_task_create ((GstTaskFunction) gst_rtspsrc_loop, src);
gst_task_set_lock (src->task, src->stream_rec_lock);
}
src->running = TRUE;
gst_rtspsrc_loop_send_cmd (src, CMD_WAIT);
gst_task_start (src->task);
return TRUE;
/* ERRORS */
create_request_failed:
{
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request."));
return FALSE;
}
send_error:
{
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
rtsp_message_unset (&request);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message."));
return FALSE;
}
}
static gboolean
gst_rtspsrc_pause (GstRTSPSrc * src)
{
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
RTSPResult res;
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
if (!(src->methods & RTSP_PAUSE))
return TRUE;
GST_DEBUG_OBJECT (src, "PAUSE...");
/* do pause */
res = rtsp_message_init_request (&request, RTSP_PAUSE, src->req_location);
if (res < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
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
rtsp_message_unset (&request);
rtsp_message_unset (&response);
return TRUE;
/* ERRORS */
create_request_failed:
{
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request."));
return FALSE;
}
send_error:
{
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
rtsp_message_unset (&request);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message."));
return FALSE;
}
}
static void
gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
{
GstRTSPSrc *rtspsrc;
rtspsrc = GST_RTSPSRC (bin);
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ELEMENT:
{
const GstStructure *s = gst_message_get_structure (message);
if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
GST_DEBUG_OBJECT (bin, "timeout on UDP port");
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT);
return;
}
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
case GST_MESSAGE_ERROR:
{
GstObject *udpsrc;
GList *lstream;
GstRTSPStream *stream;
GstFlowReturn ret;
udpsrc = GST_MESSAGE_SRC (message);
lstream = g_list_find_custom (rtspsrc->streams, udpsrc,
(GCompareFunc) find_stream_by_udpsrc);
if (!lstream)
goto forward;
stream = (GstRTSPStream *) lstream->data;
/* if we get error messages from the udp sources, that's not a problem as
* long as not all of them error out. We also don't really know what the
* problem is, the message does not give enough detail... */
ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
if (ret != GST_FLOW_OK)
goto forward;
gst_message_unref (message);
break;
/* fatal our not our message, forward */
forward:
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
default:
{
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
}
}
static GstStateChangeReturn
gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
{
GstRTSPSrc *rtspsrc;
GstStateChangeReturn ret;
rtspsrc = GST_RTSPSRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
rtspsrc->cur_protocols = rtspsrc->protocols;
if (!gst_rtspsrc_open (rtspsrc))
goto open_failed;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
rtsp_connection_flush (rtspsrc->connection, FALSE);
gst_rtspsrc_play (rtspsrc);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_READY:
rtsp_connection_flush (rtspsrc->connection, TRUE);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
goto done;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
gst_rtspsrc_pause (rtspsrc);
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtspsrc_close (rtspsrc);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
done:
return ret;
open_failed:
{
return GST_STATE_CHANGE_FAILURE;
}
}
/*** GSTURIHANDLER INTERFACE *************************************************/
static GstURIType
gst_rtspsrc_uri_get_type (void)
{
return GST_URI_SRC;
}
static gchar **
gst_rtspsrc_uri_get_protocols (void)
{
static gchar *protocols[] = { "rtsp", "rtspu", "rtspt", NULL };
return protocols;
}
static const gchar *
gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
{
GstRTSPSrc *src = GST_RTSPSRC (handler);
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
/* should not dup */
return src->location;
}
static gboolean
gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri)
{
GstRTSPSrc *src;
RTSPResult res;
RTSPUrl *newurl;
src = GST_RTSPSRC (handler);
/* same URI, we're fine */
if (src->location && uri && !strcmp (uri, src->location))
goto was_ok;
/* try to parse */
if ((res = rtsp_url_parse (uri, &newurl)) < 0)
goto parse_error;
/* if worked, free previous and store new url object along with the original
* location. */
rtsp_url_free (src->url);
src->url = newurl;
g_free (src->location);
g_free (src->req_location);
src->location = g_strdup (uri);
src->req_location = rtsp_url_get_request_uri (src->url);
GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
GST_DEBUG_OBJECT (src, "request uri is: %s",
GST_STR_NULL (src->req_location));
return TRUE;
/* Special cases */
was_ok:
{
GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
return TRUE;
}
parse_error:
{
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
GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
GST_STR_NULL (uri), res);
return FALSE;
}
}
static void
gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
{
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
iface->get_type = gst_rtspsrc_uri_get_type;
iface->get_protocols = gst_rtspsrc_uri_get_protocols;
iface->get_uri = gst_rtspsrc_uri_get_uri;
iface->set_uri = gst_rtspsrc_uri_set_uri;
}