2 * Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
3 * <2006> Lutz Mueller <lutz at topfrose dot de>
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
18 * Boston, MA 02111-1307, USA.
21 * Unless otherwise indicated, Source Code is licensed under MIT license.
22 * See further explanation attached in License Statement (distributed in the file
25 * Permission is hereby granted, free of charge, to any person obtaining a copy of
26 * this software and associated documentation files (the "Software"), to deal in
27 * the Software without restriction, including without limitation the rights to
28 * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
29 * of the Software, and to permit persons to whom the Software is furnished to do
30 * so, subject to the following conditions:
32 * The above copyright notice and this permission notice shall be included in all
33 * copies or substantial portions of the Software.
35 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
36 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
37 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
38 * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
39 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
40 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
44 * SECTION:element-rtspsrc
46 * Makes a connection to an RTSP server and read the data.
47 * rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
48 * RealMedia/Quicktime/Microsoft extensions.
50 * RTSP supports transport over TCP or UDP in unicast or multicast mode. By
51 * default rtspsrc will negotiate a connection in the following order:
52 * UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
53 * protocols can be controlled with the #GstRTSPSrc:protocols property.
55 * rtspsrc currently understands SDP as the format of the session description.
56 * For each stream listed in the SDP a new rtp_stream%d pad will be created
57 * with caps derived from the SDP media description. This is a caps of mime type
58 * "application/x-rtp" that can be connected to any available RTP depayloader
61 * rtspsrc will internally instantiate an RTP session manager element
62 * that will handle the RTCP messages to and from the server, jitter removal,
63 * packet reordering along with providing a clock for the pipeline.
64 * This feature is implemented using the gstrtpbin element.
66 * rtspsrc acts like a live source and will therefore only generate data in the
70 * <title>Example launch line</title>
72 * gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
73 * ]| Establish a connection to an RTSP server and send the raw RTP packets to a
77 * Last reviewed on 2006-08-18 (0.10.5)
86 #endif /* HAVE_UNISTD_H */
93 #include <gst/sdp/gstsdpmessage.h>
94 #include <gst/rtp/gstrtppayloads.h>
96 #include "gst/gst-i18n-plugin.h"
98 #include "gstrtspsrc.h"
100 GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
101 #define GST_CAT_DEFAULT (rtspsrc_debug)
103 static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
106 GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
108 /* templates used internally */
109 static GstStaticPadTemplate anysrctemplate =
110 GST_STATIC_PAD_TEMPLATE ("internalsrc_%u",
113 GST_STATIC_CAPS_ANY);
115 static GstStaticPadTemplate anysinktemplate =
116 GST_STATIC_PAD_TEMPLATE ("internalsink_%u",
119 GST_STATIC_CAPS_ANY);
127 enum _GstRtspSrcRtcpSyncMode
134 enum _GstRtspSrcBufferMode
142 #define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
144 gst_rtsp_src_buffer_mode_get_type (void)
146 static GType buffer_mode_type = 0;
147 static const GEnumValue buffer_modes[] = {
148 {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
149 {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
150 {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
151 {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
155 if (!buffer_mode_type) {
157 g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
159 return buffer_mode_type;
162 #define DEFAULT_LOCATION NULL
163 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
164 #define DEFAULT_DEBUG FALSE
165 #define DEFAULT_RETRY 20
166 #define DEFAULT_TIMEOUT 5000000
167 #define DEFAULT_UDP_BUFFER_SIZE 0x80000
168 #define DEFAULT_TCP_TIMEOUT 20000000
169 #define DEFAULT_LATENCY_MS 2000
170 #define DEFAULT_CONNECTION_SPEED 0
171 #define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
172 #define DEFAULT_DO_RTCP TRUE
173 #define DEFAULT_PROXY NULL
174 #define DEFAULT_RTP_BLOCKSIZE 0
175 #define DEFAULT_USER_ID NULL
176 #define DEFAULT_USER_PW NULL
177 #define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO
178 #define DEFAULT_PORT_RANGE NULL
179 #define DEFAULT_SHORT_HEADER FALSE
191 PROP_CONNECTION_SPEED,
200 PROP_UDP_BUFFER_SIZE,
205 #define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
207 gst_rtsp_nat_method_get_type (void)
209 static GType rtsp_nat_method_type = 0;
210 static const GEnumValue rtsp_nat_method[] = {
211 {GST_RTSP_NAT_NONE, "None", "none"},
212 {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
216 if (!rtsp_nat_method_type) {
217 rtsp_nat_method_type =
218 g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
220 return rtsp_nat_method_type;
223 static void gst_rtspsrc_finalize (GObject * object);
225 static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
226 const GValue * value, GParamSpec * pspec);
227 static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
228 GValue * value, GParamSpec * pspec);
230 static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
231 gpointer iface_data);
233 static void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
236 static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
237 static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
239 static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
241 static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
242 GstStateChange transition);
243 static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
244 static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
246 static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
247 GstRTSPMessage * response);
249 static void gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd);
250 static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
251 GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
253 static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
254 static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
256 static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle,
258 static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
259 gboolean only_close);
261 static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
262 const gchar * uri, GError ** error);
264 static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
265 static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
266 static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
267 GstRTSPStream * stream, GstEvent * event, gboolean source);
268 static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event,
271 /* commands we send to out loop to notify it of events */
277 #define CMD_RECONNECT 5
280 #define GST_ELEMENT_PROGRESS(el, type, code, text) \
282 gchar *__txt = _gst_element_error_printf text; \
283 gst_element_post_message (GST_ELEMENT_CAST (el), \
284 gst_message_new_progress (GST_OBJECT_CAST (el), \
285 GST_PROGRESS_TYPE_ ##type, code, __txt)); \
289 /*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */
290 #define gst_rtspsrc_parent_class parent_class
291 G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
292 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
295 gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
297 GObjectClass *gobject_class;
298 GstElementClass *gstelement_class;
299 GstBinClass *gstbin_class;
301 gobject_class = (GObjectClass *) klass;
302 gstelement_class = (GstElementClass *) klass;
303 gstbin_class = (GstBinClass *) klass;
305 GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
307 gobject_class->set_property = gst_rtspsrc_set_property;
308 gobject_class->get_property = gst_rtspsrc_get_property;
310 gobject_class->finalize = gst_rtspsrc_finalize;
312 g_object_class_install_property (gobject_class, PROP_LOCATION,
313 g_param_spec_string ("location", "RTSP Location",
314 "Location of the RTSP url to read",
315 DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
317 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
318 g_param_spec_flags ("protocols", "Protocols",
319 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
320 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
322 g_object_class_install_property (gobject_class, PROP_DEBUG,
323 g_param_spec_boolean ("debug", "Debug",
324 "Dump request and response messages to stdout",
325 DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
327 g_object_class_install_property (gobject_class, PROP_RETRY,
328 g_param_spec_uint ("retry", "Retry",
329 "Max number of retries when allocating RTP ports.",
330 0, G_MAXUINT16, DEFAULT_RETRY,
331 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
333 g_object_class_install_property (gobject_class, PROP_TIMEOUT,
334 g_param_spec_uint64 ("timeout", "Timeout",
335 "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
336 0, G_MAXUINT64, DEFAULT_TIMEOUT,
337 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
339 g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
340 g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
341 "Fail after timeout microseconds on TCP connections (0 = disabled)",
342 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
343 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
345 g_object_class_install_property (gobject_class, PROP_LATENCY,
346 g_param_spec_uint ("latency", "Buffer latency in ms",
347 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
348 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
350 g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
351 g_param_spec_uint64 ("connection-speed", "Connection Speed",
352 "Network connection speed in kbps (0 = unknown)",
353 0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
354 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
356 g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
357 g_param_spec_enum ("nat-method", "NAT Method",
358 "Method to use for traversing firewalls and NAT",
359 GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
360 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
363 * GstRTSPSrc::do-rtcp
365 * Enable RTCP support. Some old server don't like RTCP and then this property
366 * needs to be set to FALSE.
370 g_object_class_install_property (gobject_class, PROP_DO_RTCP,
371 g_param_spec_boolean ("do-rtcp", "Do RTCP",
372 "Send RTCP packets, disable for old incompatible server.",
373 DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
378 * Set the proxy parameters. This has to be a string of the format
379 * [http://][user:passwd@]host[:port].
383 g_object_class_install_property (gobject_class, PROP_PROXY,
384 g_param_spec_string ("proxy", "Proxy",
385 "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
386 DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
389 * GstRTSPSrc::rtp_blocksize
391 * RTP package size to suggest to server.
395 g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
396 g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
397 "RTP package size to suggest to server (0 = disabled)",
398 0, 65536, DEFAULT_RTP_BLOCKSIZE,
399 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
401 g_object_class_install_property (gobject_class,
403 g_param_spec_string ("user-id", "user-id",
404 "RTSP location URI user id for authentication", DEFAULT_USER_ID,
405 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
406 g_object_class_install_property (gobject_class, PROP_USER_PW,
407 g_param_spec_string ("user-pw", "user-pw",
408 "RTSP location URI user password for authentication", DEFAULT_USER_PW,
409 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
412 * GstRTSPSrc::buffer-mode:
414 * Control the buffering and timestamping mode used by the jitterbuffer.
418 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
419 g_param_spec_enum ("buffer-mode", "Buffer Mode",
420 "Control the buffering algorithm in use",
421 GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
422 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
425 * GstRTSPSrc::port-range:
427 * Configure the client port numbers that can be used to recieve RTP and
432 g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
433 g_param_spec_string ("port-range", "Port range",
434 "Client port range that can be used to receive RTP and RTCP data, "
435 "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
436 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
439 * GstRTSPSrc::udp-buffer-size:
441 * Size of the kernel UDP receive buffer in bytes.
445 g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
446 g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
447 "Size of the kernel UDP receive buffer in bytes, 0=default",
448 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
449 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
452 * GstRTSPSrc::short-header:
454 * Only send the basic RTSP headers for broken encoders.
458 g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
459 g_param_spec_boolean ("short-header", "Short Header",
460 "Only send the basic RTSP headers for broken encoders",
461 DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
463 gstelement_class->send_event = gst_rtspsrc_send_event;
464 gstelement_class->change_state = gst_rtspsrc_change_state;
466 gst_element_class_add_pad_template (gstelement_class,
467 gst_static_pad_template_get (&rtptemplate));
469 gst_element_class_set_details_simple (gstelement_class,
470 "RTSP packet receiver", "Source/Network",
471 "Receive data over the network via RTSP (RFC 2326)",
472 "Wim Taymans <wim@fluendo.com>, "
473 "Thijs Vermeir <thijs.vermeir@barco.com>, "
474 "Lutz Mueller <lutz@topfrose.de>");
476 gstbin_class->handle_message = gst_rtspsrc_handle_message;
478 gst_rtsp_ext_list_init ();
483 gst_rtspsrc_init (GstRTSPSrc * src)
485 src->conninfo.location = g_strdup (DEFAULT_LOCATION);
486 src->protocols = DEFAULT_PROTOCOLS;
487 src->debug = DEFAULT_DEBUG;
488 src->retry = DEFAULT_RETRY;
489 src->udp_timeout = DEFAULT_TIMEOUT;
490 gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
491 src->latency = DEFAULT_LATENCY_MS;
492 src->connection_speed = DEFAULT_CONNECTION_SPEED;
493 src->nat_method = DEFAULT_NAT_METHOD;
494 src->do_rtcp = DEFAULT_DO_RTCP;
495 gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
496 src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
497 src->user_id = g_strdup (DEFAULT_USER_ID);
498 src->user_pw = g_strdup (DEFAULT_USER_PW);
499 src->buffer_mode = DEFAULT_BUFFER_MODE;
500 src->client_port_range.min = 0;
501 src->client_port_range.max = 0;
502 src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
503 src->short_header = DEFAULT_SHORT_HEADER;
505 /* get a list of all extensions */
506 src->extensions = gst_rtsp_ext_list_get ();
508 /* connect to send signal */
509 gst_rtsp_ext_list_connect (src->extensions, "send",
510 (GCallback) gst_rtspsrc_send_cb, src);
512 /* protects the streaming thread in interleaved mode or the polling
513 * thread in UDP mode. */
514 g_rec_mutex_init (&src->stream_rec_lock);
516 /* protects our state changes from multiple invocations */
517 g_rec_mutex_init (&src->state_rec_lock);
519 src->state = GST_RTSP_STATE_INVALID;
521 GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
525 gst_rtspsrc_finalize (GObject * object)
529 rtspsrc = GST_RTSPSRC (object);
531 gst_rtsp_ext_list_free (rtspsrc->extensions);
532 g_free (rtspsrc->conninfo.location);
533 gst_rtsp_url_free (rtspsrc->conninfo.url);
534 g_free (rtspsrc->conninfo.url_str);
535 g_free (rtspsrc->user_id);
536 g_free (rtspsrc->user_pw);
539 gst_sdp_message_free (rtspsrc->sdp);
544 g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
545 g_rec_mutex_clear (&rtspsrc->state_rec_lock);
547 G_OBJECT_CLASS (parent_class)->finalize (object);
550 /* a proxy string of the format [user:passwd@]host[:port] */
552 gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
556 g_free (rtsp->proxy_user);
557 rtsp->proxy_user = NULL;
558 g_free (rtsp->proxy_passwd);
559 rtsp->proxy_passwd = NULL;
560 g_free (rtsp->proxy_host);
561 rtsp->proxy_host = NULL;
562 rtsp->proxy_port = 0;
569 /* we allow http:// in front but ignore it */
570 if (g_str_has_prefix (p, "http://"))
573 at = strchr (p, '@');
575 /* look for user:passwd */
576 col = strchr (proxy, ':');
577 if (col == NULL || col > at)
580 rtsp->proxy_user = g_strndup (p, col - p);
582 rtsp->proxy_passwd = g_strndup (col, at - col);
587 col = strchr (p, ':');
590 /* everything before the colon is the hostname */
591 rtsp->proxy_host = g_strndup (p, col - p);
593 rtsp->proxy_port = strtoul (p, (char **) &p, 10);
595 rtsp->proxy_host = g_strdup (p);
596 rtsp->proxy_port = 8080;
602 gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
604 rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
605 rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
608 rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
610 rtspsrc->ptcp_timeout = NULL;
614 gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
619 rtspsrc = GST_RTSPSRC (object);
623 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
624 g_value_get_string (value), NULL);
627 rtspsrc->protocols = g_value_get_flags (value);
630 rtspsrc->debug = g_value_get_boolean (value);
633 rtspsrc->retry = g_value_get_uint (value);
636 rtspsrc->udp_timeout = g_value_get_uint64 (value);
638 case PROP_TCP_TIMEOUT:
639 gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
642 rtspsrc->latency = g_value_get_uint (value);
644 case PROP_CONNECTION_SPEED:
645 rtspsrc->connection_speed = g_value_get_uint64 (value);
647 case PROP_NAT_METHOD:
648 rtspsrc->nat_method = g_value_get_enum (value);
651 rtspsrc->do_rtcp = g_value_get_boolean (value);
654 gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
656 case PROP_RTP_BLOCKSIZE:
657 rtspsrc->rtp_blocksize = g_value_get_uint (value);
660 if (rtspsrc->user_id)
661 g_free (rtspsrc->user_id);
662 rtspsrc->user_id = g_value_dup_string (value);
665 if (rtspsrc->user_pw)
666 g_free (rtspsrc->user_pw);
667 rtspsrc->user_pw = g_value_dup_string (value);
669 case PROP_BUFFER_MODE:
670 rtspsrc->buffer_mode = g_value_get_enum (value);
672 case PROP_PORT_RANGE:
676 str = g_value_get_string (value);
678 sscanf (str, "%u-%u",
679 &rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
681 rtspsrc->client_port_range.min = 0;
682 rtspsrc->client_port_range.max = 0;
686 case PROP_UDP_BUFFER_SIZE:
687 rtspsrc->udp_buffer_size = g_value_get_int (value);
689 case PROP_SHORT_HEADER:
690 rtspsrc->short_header = g_value_get_boolean (value);
693 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
699 gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
704 rtspsrc = GST_RTSPSRC (object);
708 g_value_set_string (value, rtspsrc->conninfo.location);
711 g_value_set_flags (value, rtspsrc->protocols);
714 g_value_set_boolean (value, rtspsrc->debug);
717 g_value_set_uint (value, rtspsrc->retry);
720 g_value_set_uint64 (value, rtspsrc->udp_timeout);
722 case PROP_TCP_TIMEOUT:
726 timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
727 rtspsrc->tcp_timeout.tv_usec;
728 g_value_set_uint64 (value, timeout);
732 g_value_set_uint (value, rtspsrc->latency);
734 case PROP_CONNECTION_SPEED:
735 g_value_set_uint64 (value, rtspsrc->connection_speed);
737 case PROP_NAT_METHOD:
738 g_value_set_enum (value, rtspsrc->nat_method);
741 g_value_set_boolean (value, rtspsrc->do_rtcp);
747 if (rtspsrc->proxy_host) {
749 g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
753 g_value_take_string (value, str);
756 case PROP_RTP_BLOCKSIZE:
757 g_value_set_uint (value, rtspsrc->rtp_blocksize);
760 g_value_set_string (value, rtspsrc->user_id);
763 g_value_set_string (value, rtspsrc->user_pw);
765 case PROP_BUFFER_MODE:
766 g_value_set_enum (value, rtspsrc->buffer_mode);
768 case PROP_PORT_RANGE:
772 if (rtspsrc->client_port_range.min != 0) {
773 str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
774 rtspsrc->client_port_range.max);
778 g_value_take_string (value, str);
781 case PROP_UDP_BUFFER_SIZE:
782 g_value_set_int (value, rtspsrc->udp_buffer_size);
784 case PROP_SHORT_HEADER:
785 g_value_set_boolean (value, rtspsrc->short_header);
788 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
794 find_stream_by_id (GstRTSPStream * stream, gint * id)
796 if (stream->id == *id)
803 find_stream_by_channel (GstRTSPStream * stream, gint * channel)
805 if (stream->channel[0] == *channel || stream->channel[1] == *channel)
812 find_stream_by_pt (GstRTSPStream * stream, gint * pt)
814 if (stream->pt == *pt)
821 find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
823 GstElement *src = (GstElement *) a;
825 if (stream->udpsrc[0] == src)
827 if (stream->udpsrc[1] == src)
834 find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
836 /* check qualified setup_url */
837 if (!strcmp (stream->conninfo.location, (gchar *) a))
839 /* check original control_url */
840 if (!strcmp (stream->control_url, (gchar *) a))
843 /* check if qualified setup_url ends with string */
844 if (g_str_has_suffix (stream->control_url, (gchar *) a))
850 static GstRTSPStream *
851 find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
855 /* find and get stream */
856 if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
857 return (GstRTSPStream *) lstream->data;
862 static const GstSDPBandwidth *
863 gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
864 const GstSDPMedia * media, const gchar * type)
868 /* first look in the media specific section */
869 len = gst_sdp_media_bandwidths_len (media);
870 for (i = 0; i < len; i++) {
871 const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
873 if (strcmp (bw->bwtype, type) == 0)
876 /* then look in the message specific section */
877 len = gst_sdp_message_bandwidths_len (sdp);
878 for (i = 0; i < len; i++) {
879 const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
881 if (strcmp (bw->bwtype, type) == 0)
888 gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
889 const GstSDPMedia * media, GstRTSPStream * stream)
891 const GstSDPBandwidth *bw;
893 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
894 stream->as_bandwidth = bw->bandwidth;
896 stream->as_bandwidth = -1;
898 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
899 stream->rr_bandwidth = bw->bandwidth;
901 stream->rr_bandwidth = -1;
903 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
904 stream->rs_bandwidth = bw->bandwidth;
906 stream->rs_bandwidth = -1;
910 gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
911 const GstSDPConnection * conn)
913 if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
916 if (conn->addrtype == NULL)
920 if (strcmp (conn->addrtype, "IP4") == 0)
921 stream->is_ipv6 = FALSE;
922 else if (strcmp (conn->addrtype, "IP6") == 0)
923 stream->is_ipv6 = TRUE;
928 g_free (stream->destination);
929 stream->destination = g_strdup (conn->address);
931 /* check for multicast */
932 stream->is_multicast =
933 gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
935 stream->ttl = conn->ttl;
938 /* Go over the connections for a stream.
939 * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
941 * - If we are dealing with a localhost address, we disable multicast
944 gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
945 const GstSDPMedia * media, GstRTSPStream * stream)
947 const GstSDPConnection *conn;
950 /* first look in the media specific section */
951 len = gst_sdp_media_connections_len (media);
952 for (i = 0; i < len; i++) {
953 conn = gst_sdp_media_get_connection (media, i);
955 gst_rtspsrc_do_stream_connection (src, stream, conn);
957 /* then look in the message specific section */
958 if ((conn = gst_sdp_message_get_connection (sdp))) {
959 gst_rtspsrc_do_stream_connection (src, stream, conn);
963 static GstRTSPStream *
964 gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
966 GstRTSPStream *stream;
967 const gchar *control_url;
968 const gchar *payload;
969 const GstSDPMedia *media;
971 /* get media, should not return NULL */
972 media = gst_sdp_message_get_media (sdp, idx);
976 stream = g_new0 (GstRTSPStream, 1);
977 stream->parent = src;
978 /* we mark the pad as not linked, we will mark it as OK when we add the pad to
980 stream->last_ret = GST_FLOW_NOT_LINKED;
981 stream->added = FALSE;
982 stream->disabled = FALSE;
983 stream->id = src->numstreams++;
985 stream->discont = TRUE;
986 stream->seqbase = -1;
987 stream->timebase = -1;
989 /* collect bandwidth information for this steam. FIXME, configure in the RTP
990 * session manager to scale RTCP. */
991 gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
993 /* collect connection info */
994 gst_rtspsrc_collect_connections (src, sdp, media, stream);
996 /* we must have a payload. No payload means we cannot create caps */
997 /* FIXME, handle multiple formats. The problem here is that we just want to
998 * take the first available format that we can handle but in order to do that
999 * we need to scan for depayloader plugins. Scanning for payloader plugins is
1000 * also suboptimal because the user maybe just wants to save the raw stream
1001 * and then we don't care. */
1002 if ((payload = gst_sdp_media_get_format (media, 0))) {
1003 stream->pt = atoi (payload);
1005 stream->caps = gst_rtspsrc_media_to_caps (stream->pt, media);
1007 GST_DEBUG ("mapping sdp session level attributes to caps");
1008 gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, stream->caps);
1009 GST_DEBUG ("mapping sdp media level attributes to caps");
1010 gst_rtspsrc_sdp_attributes_to_caps (media->attributes, stream->caps);
1012 if (stream->pt >= 96) {
1013 /* If we have a dynamic payload type, see if we have a stream with the
1014 * same payload number. If there is one, they are part of the same
1015 * container and we only need to add one pad. */
1016 if (find_stream (src, &stream->pt, (gpointer) find_stream_by_pt)) {
1017 stream->container = TRUE;
1018 GST_DEBUG ("found another stream with pt %d, marking as container",
1023 /* collect port number */
1024 stream->port = gst_sdp_media_get_port (media);
1026 /* get control url to construct the setup url. The setup url is used to
1027 * configure the transport of the stream and is used to identity the stream in
1028 * the RTP-Info header field returned from PLAY. */
1029 control_url = gst_sdp_media_get_attribute_val (media, "control");
1030 if (control_url == NULL)
1031 control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
1033 GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
1034 GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
1035 GST_DEBUG_OBJECT (src, " port: %d", stream->port);
1036 GST_DEBUG_OBJECT (src, " container: %d", stream->container);
1037 GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps);
1038 GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
1040 if (control_url != NULL) {
1041 stream->control_url = g_strdup (control_url);
1042 /* Build a fully qualified url using the content_base if any or by prefixing
1043 * the original request.
1044 * If the control_url starts with a '/' or a non rtsp: protocol we will most
1045 * likely build a URL that the server will fail to understand, this is ok,
1046 * we will fail then. */
1047 if (g_str_has_prefix (control_url, "rtsp://"))
1048 stream->conninfo.location = g_strdup (control_url);
1053 if (g_strcmp0 (control_url, "*") == 0)
1057 base = src->control;
1058 else if (src->content_base)
1059 base = src->content_base;
1060 else if (src->conninfo.url_str)
1061 base = src->conninfo.url_str;
1065 /* check if the base ends or control starts with / */
1066 has_slash = g_str_has_prefix (control_url, "/");
1067 has_slash = has_slash || g_str_has_suffix (base, "/");
1069 /* concatenate the two strings, insert / when not present */
1070 stream->conninfo.location =
1071 g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
1074 GST_DEBUG_OBJECT (src, " setup: %s",
1075 GST_STR_NULL (stream->conninfo.location));
1077 /* we keep track of all streams */
1078 src->streams = g_list_append (src->streams, stream);
1086 gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
1090 GST_DEBUG_OBJECT (src, "free stream %p", stream);
1093 gst_caps_unref (stream->caps);
1095 g_free (stream->destination);
1096 g_free (stream->control_url);
1097 g_free (stream->conninfo.location);
1099 for (i = 0; i < 2; i++) {
1100 if (stream->udpsrc[i]) {
1101 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
1102 gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
1103 gst_object_unref (stream->udpsrc[i]);
1104 stream->udpsrc[i] = NULL;
1106 if (stream->channelpad[i]) {
1107 gst_object_unref (stream->channelpad[i]);
1108 stream->channelpad[i] = NULL;
1110 if (stream->udpsink[i]) {
1111 gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
1112 gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
1113 gst_object_unref (stream->udpsink[i]);
1114 stream->udpsink[i] = NULL;
1117 if (stream->fakesrc) {
1118 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
1119 gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
1120 gst_object_unref (stream->fakesrc);
1121 stream->fakesrc = NULL;
1123 if (stream->srcpad) {
1124 gst_pad_set_active (stream->srcpad, FALSE);
1125 if (stream->added) {
1126 gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
1127 stream->added = FALSE;
1129 stream->srcpad = NULL;
1131 if (stream->rtcppad) {
1132 gst_object_unref (stream->rtcppad);
1133 stream->rtcppad = NULL;
1135 if (stream->session) {
1136 g_object_unref (stream->session);
1137 stream->session = NULL;
1143 gst_rtspsrc_cleanup (GstRTSPSrc * src)
1147 GST_DEBUG_OBJECT (src, "cleanup");
1149 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1150 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1152 gst_rtspsrc_stream_free (src, stream);
1154 g_list_free (src->streams);
1155 src->streams = NULL;
1157 if (src->manager_sig_id) {
1158 g_signal_handler_disconnect (src->manager, src->manager_sig_id);
1159 src->manager_sig_id = 0;
1161 gst_element_set_state (src->manager, GST_STATE_NULL);
1162 gst_bin_remove (GST_BIN_CAST (src), src->manager);
1163 src->manager = NULL;
1165 src->numstreams = 0;
1167 gst_structure_free (src->props);
1170 g_free (src->content_base);
1171 src->content_base = NULL;
1173 g_free (src->control);
1174 src->control = NULL;
1177 gst_rtsp_range_free (src->range);
1180 /* don't clear the SDP when it was used in the url */
1181 if (src->sdp && !src->from_sdp) {
1182 gst_sdp_message_free (src->sdp);
1187 #define PARSE_INT(p, del, res) \
1190 p = strstr (p, del); \
1200 #define PARSE_STRING(p, del, res) \
1203 p = strstr (p, del); \
1215 #define SKIP_SPACES(p) \
1216 while (*p && g_ascii_isspace (*p)) \
1221 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1224 gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
1225 gint * rate, gchar ** params)
1229 p = (gchar *) rtpmap;
1231 PARSE_INT (p, " ", *payload);
1239 PARSE_STRING (p, "/", *name);
1240 if (*name == NULL) {
1241 GST_DEBUG ("no rate, name %s", p);
1242 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
1243 * streams seem to omit the rate. */
1250 p = strstr (p, "/");
1268 * Mapping SDP attributes to caps
1270 * prepend 'a-' to IANA registered sdp attributes names
1271 * (ie: not prefixed with 'x-') in order to avoid
1272 * collision with gstreamer standard caps properties names
1275 gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
1277 if (attributes->len > 0) {
1281 s = gst_caps_get_structure (caps, 0);
1283 for (i = 0; i < attributes->len; i++) {
1284 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
1285 gchar *tofree, *key;
1289 /* skip some of the attribute we already handle */
1290 if (!strcmp (key, "fmtp"))
1292 if (!strcmp (key, "rtpmap"))
1294 if (!strcmp (key, "control"))
1296 if (!strcmp (key, "range"))
1299 /* string must be valid UTF8 */
1300 if (!g_utf8_validate (attr->value, -1, NULL))
1303 if (!g_str_has_prefix (key, "x-"))
1304 tofree = key = g_strdup_printf ("a-%s", key);
1308 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
1309 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
1316 * Mapping of caps to and from SDP fields:
1318 * m=<media> <UDP port> RTP/AVP <payload>
1319 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1320 * a=fmtp:<payload> <param>[=<value>];...
1323 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
1326 const gchar *rtpmap;
1330 gchar *params = NULL;
1336 /* get and parse rtpmap */
1337 if ((rtpmap = gst_sdp_media_get_attribute_val (media, "rtpmap"))) {
1338 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
1340 if (payload != pt) {
1341 /* we ignore the rtpmap if the payload type is different. */
1342 g_warning ("rtpmap of wrong payload type, ignoring");
1348 /* if we failed to parse the rtpmap for a dynamic payload type, we have an
1352 /* else we can ignore */
1353 g_warning ("error parsing rtpmap, ignoring");
1356 /* dynamic payloads need rtpmap or we fail */
1360 /* check if we have a rate, if not, we need to look up the rate from the
1361 * default rates based on the payload types. */
1363 const GstRTPPayloadInfo *info;
1365 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
1366 /* dynamic types, use media and encoding_name */
1367 tmp = g_ascii_strdown (media->media, -1);
1368 info = gst_rtp_payload_info_for_name (tmp, name);
1371 /* static types, use payload type */
1372 info = gst_rtp_payload_info_for_pt (pt);
1376 if ((rate = info->clock_rate) == 0)
1379 /* we fail if we cannot find one */
1384 tmp = g_ascii_strdown (media->media, -1);
1385 caps = gst_caps_new_simple ("application/x-unknown",
1386 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
1388 s = gst_caps_get_structure (caps, 0);
1390 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
1392 /* encoding name must be upper case */
1394 tmp = g_ascii_strup (name, -1);
1395 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
1399 /* params must be lower case */
1400 if (params != NULL) {
1401 tmp = g_ascii_strdown (params, -1);
1402 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
1406 /* parse optional fmtp: field */
1407 if ((fmtp = gst_sdp_media_get_attribute_val (media, "fmtp"))) {
1413 /* p is now of the format <payload> <param>[=<value>];... */
1414 PARSE_INT (p, " ", payload);
1415 if (payload != -1 && payload == pt) {
1419 /* <param>[=<value>] are separated with ';' */
1420 pairs = g_strsplit (p, ";", 0);
1421 for (i = 0; pairs[i]; i++) {
1423 const gchar *val, *key;
1425 /* the key may not have a '=', the value can have other '='s */
1426 valpos = strstr (pairs[i], "=");
1428 /* we have a '=' and thus a value, remove the '=' with \0 */
1430 /* value is everything between '=' and ';'. We split the pairs at ;
1431 * boundaries so we can take the remainder of the value. Some servers
1432 * put spaces around the value which we strip off here. Alternatively
1433 * we could strip those spaces in the depayloaders should these spaces
1434 * actually carry any meaning in the future. */
1435 val = g_strstrip (valpos + 1);
1437 /* simple <param>;.. is translated into <param>=1;... */
1440 /* strip the key of spaces, convert key to lowercase but not the value. */
1441 key = g_strstrip (pairs[i]);
1442 if (strlen (key) > 1) {
1443 tmp = g_ascii_strdown (key, -1);
1444 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
1456 g_warning ("rtpmap type not given for dynamic payload %d", pt);
1461 g_warning ("rate unknown for payload type %d", pt);
1467 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
1468 gint * rtpport, gint * rtcpport)
1471 GstStateChangeReturn ret;
1472 GstElement *udpsrc0, *udpsrc1;
1473 gint tmp_rtp, tmp_rtcp;
1477 src = stream->parent;
1483 /* Start at next port */
1484 tmp_rtp = src->next_port_num;
1486 if (stream->is_ipv6)
1487 host = "udp://[::0]";
1489 host = "udp://0.0.0.0";
1491 /* try to allocate 2 UDP ports, the RTP port should be an even
1492 * number and the RTCP port should be the next (uneven) port */
1495 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
1496 tmp_rtp >= src->client_port_range.max)
1499 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
1500 if (udpsrc0 == NULL)
1501 goto no_udp_protocol;
1502 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
1504 if (src->udp_buffer_size != 0)
1505 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
1508 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
1509 if (ret == GST_STATE_CHANGE_FAILURE) {
1511 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
1514 if (++count > src->retry)
1517 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1518 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1519 gst_object_unref (udpsrc0);
1521 GST_DEBUG_OBJECT (src, "retry %d", count);
1524 goto no_udp_protocol;
1527 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
1528 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
1530 /* check if port is even */
1531 if ((tmp_rtp & 0x01) != 0) {
1532 /* port not even, close and allocate another */
1533 if (++count > src->retry)
1536 GST_DEBUG_OBJECT (src, "RTP port not even");
1538 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1539 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1540 gst_object_unref (udpsrc0);
1542 GST_DEBUG_OBJECT (src, "retry %d", count);
1547 /* allocate port+1 for RTCP now */
1548 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
1549 if (udpsrc1 == NULL)
1550 goto no_udp_rtcp_protocol;
1553 tmp_rtcp = tmp_rtp + 1;
1554 if (src->client_port_range.max > 0 && tmp_rtcp >= src->client_port_range.max)
1557 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
1559 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
1560 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
1561 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
1562 if (ret == GST_STATE_CHANGE_FAILURE) {
1563 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
1565 if (++count > src->retry)
1568 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1569 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1570 gst_object_unref (udpsrc0);
1572 GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
1573 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1574 gst_object_unref (udpsrc1);
1578 GST_DEBUG_OBJECT (src, "retry %d", count);
1582 /* all fine, do port check */
1583 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
1584 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
1586 /* this should not happen... */
1587 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
1590 /* we keep these elements, we configure all in configure_transport when the
1591 * server told us to really use the UDP ports. */
1592 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
1593 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
1595 /* keep track of next available port number when we have a range
1597 if (src->next_port_num != 0)
1598 src->next_port_num = tmp_rtcp + 1;
1605 GST_DEBUG_OBJECT (src, "could not get UDP source");
1610 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
1614 no_udp_rtcp_protocol:
1616 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
1621 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
1622 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
1628 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1629 gst_object_unref (udpsrc0);
1632 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1633 gst_object_unref (udpsrc1);
1640 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
1647 GstClockTime base_time = GST_CLOCK_TIME_NONE;
1650 event = gst_event_new_flush_start ();
1651 GST_DEBUG_OBJECT (src, "start flush");
1653 state = GST_STATE_PAUSED;
1655 event = gst_event_new_flush_stop (TRUE);
1656 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
1659 state = GST_STATE_PLAYING;
1661 state = GST_STATE_PAUSED;
1662 clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
1664 base_time = gst_clock_get_time (clock);
1665 gst_object_unref (clock);
1668 gst_rtspsrc_push_event (src, event, FALSE);
1669 gst_rtspsrc_loop_send_cmd (src, cmd);
1671 /* set up manager before data-flow resumes */
1672 /* to manage jitterbuffer buffer mode */
1674 gst_element_set_base_time (GST_ELEMENT_CAST (src->manager), base_time);
1675 /* and to have base_time trickle further down,
1676 * e.g. to jitterbuffer for its timeout handling */
1677 if (base_time != -1)
1678 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
1681 /* make running time start start at 0 again */
1682 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1683 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1685 for (i = 0; i < 2; i++) {
1687 if (stream->udpsrc[i]) {
1688 if (base_time != -1)
1689 gst_element_set_base_time (stream->udpsrc[i], base_time);
1690 gst_element_set_state (stream->udpsrc[i], state);
1694 /* for tcp interleaved case */
1695 if (base_time != -1)
1696 gst_element_set_base_time (GST_ELEMENT_CAST (src), base_time);
1699 static GstRTSPResult
1700 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
1701 GstRTSPMessage * message, GTimeVal * timeout)
1706 ret = gst_rtsp_connection_send (conn, message, timeout);
1708 ret = GST_RTSP_ERROR;
1713 static GstRTSPResult
1714 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
1715 GstRTSPMessage * message, GTimeVal * timeout)
1720 ret = gst_rtsp_connection_receive (conn, message, timeout);
1722 ret = GST_RTSP_ERROR;
1728 gst_rtspsrc_get_position (GstRTSPSrc * src)
1733 query = gst_query_new_position (GST_FORMAT_TIME);
1734 /* should be known somewhere down the stream (e.g. jitterbuffer) */
1735 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1736 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1740 if (stream->srcpad) {
1741 if (gst_pad_query (stream->srcpad, query)) {
1742 gst_query_parse_position (query, &fmt, &pos);
1743 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
1744 GST_TIME_ARGS (pos));
1745 src->last_pos = pos;
1755 gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
1757 src->state = GST_RTSP_STATE_SEEKING;
1758 /* PLAY will add the range header now. */
1759 src->need_range = TRUE;
1765 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
1770 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
1772 gboolean flush, skip;
1775 GstSegment seeksegment = { 0, };
1779 GST_DEBUG_OBJECT (src, "doing seek with event");
1781 gst_event_parse_seek (event, &rate, &format, &flags,
1782 &cur_type, &cur, &stop_type, &stop);
1784 /* no negative rates yet */
1788 /* we need TIME format */
1789 if (format != src->segment.format)
1792 GST_DEBUG_OBJECT (src, "doing seek without event");
1794 cur_type = GST_SEEK_TYPE_SET;
1795 stop_type = GST_SEEK_TYPE_SET;
1798 /* get flush flag */
1799 flush = flags & GST_SEEK_FLAG_FLUSH;
1800 skip = flags & GST_SEEK_FLAG_SKIP;
1802 /* now we need to make sure the streaming thread is stopped. We do this by
1803 * either sending a FLUSH_START event downstream which will cause the
1804 * streaming thread to stop with a WRONG_STATE.
1805 * For a non-flushing seek we simply pause the task, which will happen as soon
1806 * as it completes one iteration (and thus might block when the sink is
1807 * blocking in preroll). */
1809 GST_DEBUG_OBJECT (src, "starting flush");
1810 gst_rtspsrc_flush (src, TRUE, FALSE);
1813 gst_task_pause (src->task);
1817 /* we should now be able to grab the streaming thread because we stopped it
1818 * with the above flush/pause code */
1819 GST_RTSP_STREAM_LOCK (src);
1821 GST_DEBUG_OBJECT (src, "stopped streaming");
1823 /* copy segment, we need this because we still need the old
1824 * segment when we close the current segment. */
1825 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
1827 /* configure the seek parameters in the seeksegment. We will then have the
1828 * right values in the segment to perform the seek */
1830 GST_DEBUG_OBJECT (src, "configuring seek");
1831 gst_segment_do_seek (&seeksegment, rate, format, flags,
1832 cur_type, cur, stop_type, stop, &update);
1835 /* figure out the last position we need to play. If it's configured (stop !=
1836 * -1), use that, else we play until the total duration of the file */
1837 if ((stop = seeksegment.stop) == -1)
1838 stop = seeksegment.duration;
1840 playing = (src->state == GST_RTSP_STATE_PLAYING);
1842 /* if we were playing, pause first */
1844 /* obtain current position in case seek fails */
1845 gst_rtspsrc_get_position (src);
1846 gst_rtspsrc_pause (src, FALSE, FALSE);
1849 gst_rtspsrc_do_seek (src, &seeksegment);
1851 /* and continue playing */
1853 gst_rtspsrc_play (src, &seeksegment, FALSE);
1855 /* prepare for streaming again */
1857 /* if we started flush, we stop now */
1858 GST_DEBUG_OBJECT (src, "stopping flush");
1859 gst_rtspsrc_flush (src, FALSE, playing);
1862 /* now we did the seek and can activate the new segment values */
1863 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
1865 /* if we're doing a segment seek, post a SEGMENT_START message */
1866 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
1867 gst_element_post_message (GST_ELEMENT_CAST (src),
1868 gst_message_new_segment_start (GST_OBJECT_CAST (src),
1869 src->segment.format, src->segment.position));
1872 /* now create the newsegment */
1873 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
1874 " to %" G_GINT64_FORMAT, src->segment.position, stop);
1876 /* store the newsegment event so it can be sent from the streaming thread. */
1877 if (src->start_segment)
1878 gst_event_unref (src->start_segment);
1879 src->start_segment = gst_event_new_segment (&src->segment);
1882 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
1883 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1884 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1885 stream->discont = TRUE;
1889 GST_RTSP_STREAM_UNLOCK (src);
1896 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
1901 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
1907 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
1911 gboolean res = TRUE;
1914 src = GST_RTSPSRC_CAST (parent);
1916 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
1917 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
1919 switch (GST_EVENT_TYPE (event)) {
1920 case GST_EVENT_SEEK:
1921 res = gst_rtspsrc_perform_seek (src, event);
1925 case GST_EVENT_NAVIGATION:
1926 case GST_EVENT_LATENCY:
1934 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
1935 res = gst_pad_send_event (target, event);
1936 gst_object_unref (target);
1938 gst_event_unref (event);
1941 gst_event_unref (event);
1947 /* this is the final event function we receive on the internal source pad when
1948 * we deal with TCP connections */
1950 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
1955 GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
1957 switch (GST_EVENT_TYPE (event)) {
1958 case GST_EVENT_SEEK:
1960 case GST_EVENT_NAVIGATION:
1961 case GST_EVENT_LATENCY:
1963 gst_event_unref (event);
1970 /* this is the final query function we receive on the internal source pad when
1971 * we deal with TCP connections */
1973 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
1977 gboolean res = TRUE;
1979 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
1981 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
1982 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
1984 switch (GST_QUERY_TYPE (query)) {
1985 case GST_QUERY_POSITION:
1990 case GST_QUERY_DURATION:
1994 gst_query_parse_duration (query, &format, NULL);
1997 case GST_FORMAT_TIME:
1998 gst_query_set_duration (query, format, src->segment.duration);
2006 case GST_QUERY_LATENCY:
2008 /* we are live with a min latency of 0 and unlimited max latency, this
2009 * result will be updated by the session manager if there is any. */
2010 gst_query_set_latency (query, TRUE, 0, -1);
2020 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2022 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2026 gboolean res = FALSE;
2028 src = GST_RTSPSRC_CAST (parent);
2030 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2031 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2033 switch (GST_QUERY_TYPE (query)) {
2034 case GST_QUERY_DURATION:
2038 gst_query_parse_duration (query, &format, NULL);
2041 case GST_FORMAT_TIME:
2042 gst_query_set_duration (query, format, src->segment.duration);
2050 case GST_QUERY_SEEKING:
2054 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2055 if (format == GST_FORMAT_TIME) {
2057 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2059 /* seeking without duration is unlikely */
2060 seekable = seekable && src->seekable && src->segment.duration &&
2061 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2063 /* FIXME ?? should we have 0 and segment.duration here; see demuxers */
2064 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
2065 src->segment.start, src->segment.stop);
2072 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2074 /* forward the query to the proxy target pad */
2076 res = gst_pad_query (target, query);
2077 gst_object_unref (target);
2086 /* callback for RTCP messages to be sent to the server when operating in TCP
2088 static GstFlowReturn
2089 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2092 GstRTSPStream *stream;
2093 GstFlowReturn res = GST_FLOW_OK;
2098 GstRTSPMessage message = { 0 };
2099 GstRTSPConnection *conn;
2101 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2102 src = stream->parent;
2104 data = gst_buffer_map (buffer, &bsize, NULL, GST_MAP_READ);
2107 gst_rtsp_message_init_data (&message, stream->channel[1]);
2109 /* lend the body data to the message */
2110 gst_rtsp_message_take_body (&message, data, size);
2112 if (stream->conninfo.connection)
2113 conn = stream->conninfo.connection;
2115 conn = src->conninfo.connection;
2117 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2118 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2119 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2121 /* and steal it away again because we will free it when unreffing the
2123 gst_rtsp_message_steal_body (&message, &data, &size);
2124 gst_rtsp_message_unset (&message);
2126 gst_buffer_unmap (buffer, data, size);
2127 gst_buffer_unref (buffer);
2132 static GstPadProbeReturn
2133 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2135 GstRTSPSrc *src = user_data;
2137 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2138 GST_DEBUG_PAD_NAME (pad));
2140 /* activate the streams */
2141 GST_OBJECT_LOCK (src);
2142 if (!src->need_activate)
2145 src->need_activate = FALSE;
2146 GST_OBJECT_UNLOCK (src);
2148 gst_rtspsrc_activate_streams (src);
2150 return GST_PAD_PROBE_OK;
2154 GST_OBJECT_UNLOCK (src);
2155 return GST_PAD_PROBE_OK;
2159 /* this callback is called when the session manager generated a new src pad with
2160 * payloaded RTP packets. We simply ghost the pad here. */
2162 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2165 GstPadTemplate *template;
2168 GstRTSPStream *stream;
2171 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2173 GST_RTSP_STATE_LOCK (src);
2175 name = gst_object_get_name (GST_OBJECT_CAST (pad));
2176 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
2177 goto unknown_stream;
2179 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %d, PT %d", id, ssrc, pt);
2181 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
2183 goto unknown_stream;
2185 /* create a new pad we will use to stream to */
2186 template = gst_static_pad_template_get (&rtptemplate);
2187 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
2188 gst_object_unref (template);
2191 stream->added = TRUE;
2192 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
2193 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
2194 gst_pad_set_active (stream->srcpad, TRUE);
2195 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
2197 /* check if we added all streams */
2199 for (lstream = src->streams; lstream; lstream = g_list_next (lstream)) {
2200 stream = (GstRTSPStream *) lstream->data;
2202 GST_DEBUG_OBJECT (src, "stream %p, container %d, disabled %d, added %d",
2203 stream, stream->container, stream->disabled, stream->added);
2205 /* a container stream only needs one pad added. Also disabled streams don't
2207 if (!stream->container && !stream->disabled && !stream->added) {
2212 GST_RTSP_STATE_UNLOCK (src);
2215 GST_DEBUG_OBJECT (src, "We added all streams");
2216 /* when we get here, all stream are added and we can fire the no-more-pads
2218 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
2226 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
2227 GST_RTSP_STATE_UNLOCK (src);
2234 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
2236 GstRTSPStream *stream;
2239 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
2241 GST_RTSP_STATE_LOCK (src);
2242 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2244 goto unknown_stream;
2246 caps = stream->caps;
2248 gst_caps_ref (caps);
2249 GST_RTSP_STATE_UNLOCK (src);
2255 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
2256 GST_RTSP_STATE_UNLOCK (src);
2262 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
2264 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
2270 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos (), TRUE);
2276 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
2282 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2284 GstRTSPSrc *src = stream->parent;
2286 GST_DEBUG_OBJECT (src, "source in session %u received BYE", stream->id);
2288 gst_rtspsrc_do_stream_eos (src, stream);
2292 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2294 GstRTSPSrc *src = stream->parent;
2296 GST_DEBUG_OBJECT (src, "source in session %u timed out", stream->id);
2298 gst_rtspsrc_do_stream_eos (src, stream);
2302 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
2304 GstRTSPStream *stream;
2306 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
2308 /* get stream for session */
2309 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2311 gst_rtspsrc_do_stream_eos (src, stream);
2316 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
2318 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
2322 /* try to get and configure a manager */
2324 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
2325 GstRTSPTransport * transport)
2327 const gchar *manager;
2329 GstStateChangeReturn ret;
2331 /* find a manager */
2332 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
2336 GST_DEBUG_OBJECT (src, "using manager %s", manager);
2338 /* configure the manager */
2339 if (src->manager == NULL) {
2340 GObjectClass *klass;
2343 if (!(src->manager = gst_element_factory_make (manager, NULL))) {
2345 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
2349 goto use_no_manager;
2351 if (!(src->manager = gst_element_factory_make (manager, NULL)))
2352 goto manager_failed;
2355 /* we manage this element */
2356 gst_bin_add (GST_BIN_CAST (src), src->manager);
2358 GST_OBJECT_LOCK (src);
2359 target = GST_STATE_TARGET (src);
2360 GST_OBJECT_UNLOCK (src);
2362 ret = gst_element_set_state (src->manager, target);
2363 if (ret == GST_STATE_CHANGE_FAILURE)
2364 goto start_manager_failure;
2366 g_object_set (src->manager, "latency", src->latency, NULL);
2368 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
2369 if (g_object_class_find_property (klass, "buffer-mode")) {
2370 if (src->buffer_mode != BUFFER_MODE_AUTO) {
2371 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
2373 gboolean need_slave;
2375 const gchar *encoding;
2377 /* buffer mode pauses are handled by adding offsets to buffer times,
2378 * but some depayloaders may have a hard time syncing output times
2379 * with such input times, e.g. container ones, most notably ASF */
2380 /* TODO alternatives are having an event that indicates these shifts,
2381 * or having rtsp extensions provide suggestion on buffer mode */
2382 need_slave = stream->container;
2383 if (stream->caps && (s = gst_caps_get_structure (stream->caps, 0)) &&
2384 (encoding = gst_structure_get_string (s, "encoding-name")))
2385 need_slave = need_slave || (strcmp (encoding, "X-ASF-PF") == 0);
2386 GST_DEBUG_OBJECT (src, "auto buffering mode, need_slave %d",
2388 /* valid duration implies not likely live pipeline,
2389 * so slaving in jitterbuffer does not make much sense
2390 * (and might mess things up due to bursts) */
2391 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
2392 src->segment.duration && !need_slave) {
2393 GST_DEBUG_OBJECT (src, "selected buffer");
2394 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER,
2397 GST_DEBUG_OBJECT (src, "selected slave");
2398 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
2403 /* connect to signals if we did not already do so */
2404 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
2406 src->manager_sig_id =
2407 g_signal_connect (src->manager, "pad-added",
2408 (GCallback) new_manager_pad, src);
2409 src->manager_ptmap_id =
2410 g_signal_connect (src->manager, "request-pt-map",
2411 (GCallback) request_pt_map, src);
2413 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
2417 /* we stream directly to the manager, get some pads. Each RTSP stream goes
2418 * into a separate RTP session. */
2419 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
2420 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
2422 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
2423 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
2426 /* now configure the bandwidth in the manager */
2427 if (g_signal_lookup ("get-internal-session",
2428 G_OBJECT_TYPE (src->manager)) != 0) {
2429 GObject *rtpsession;
2431 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
2434 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
2436 stream->session = rtpsession;
2438 if (stream->as_bandwidth != -1) {
2439 GST_INFO_OBJECT (src, "setting AS: %f",
2440 (gdouble) (stream->as_bandwidth * 1000));
2441 g_object_set (rtpsession, "bandwidth",
2442 (gdouble) (stream->as_bandwidth * 1000), NULL);
2444 if (stream->rr_bandwidth != -1) {
2445 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
2446 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
2449 if (stream->rs_bandwidth != -1) {
2450 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
2451 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
2454 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2456 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
2458 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
2460 g_signal_connect (rtpsession, "on-ssrc-active",
2461 (GCallback) on_ssrc_active, stream);
2472 GST_DEBUG_OBJECT (src, "cannot get a session manager");
2477 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
2480 start_manager_failure:
2482 GST_DEBUG_OBJECT (src, "could not start session manager");
2487 /* free the UDP sources allocated when negotiating a transport.
2488 * This function is called when the server negotiated to a transport where the
2489 * UDP sources are not needed anymore, such as TCP or multicast. */
2491 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
2495 for (i = 0; i < 2; i++) {
2496 if (stream->udpsrc[i]) {
2497 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
2498 gst_object_unref (stream->udpsrc[i]);
2499 stream->udpsrc[i] = NULL;
2504 /* for TCP, create pads to send and receive data to and from the manager and to
2505 * intercept various events and queries
2508 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
2509 GstRTSPTransport * transport, GstPad ** outpad)
2512 GstPadTemplate *template;
2513 GstPad *pad0, *pad1;
2515 /* configure for interleaved delivery, nothing needs to be done
2516 * here, the loop function will call the chain functions of the
2517 * session manager. */
2518 stream->channel[0] = transport->interleaved.min;
2519 stream->channel[1] = transport->interleaved.max;
2520 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
2521 stream->channel[0], stream->channel[1]);
2523 /* we can remove the allocated UDP ports now */
2524 gst_rtspsrc_stream_free_udp (stream);
2526 /* no session manager, send data to srcpad directly */
2527 if (!stream->channelpad[0]) {
2528 GST_DEBUG_OBJECT (src, "no manager, creating pad");
2530 /* create a new pad we will use to stream to */
2531 name = g_strdup_printf ("stream_%u", stream->id);
2532 template = gst_static_pad_template_get (&rtptemplate);
2533 stream->channelpad[0] = gst_pad_new_from_template (template, name);
2534 gst_object_unref (template);
2537 /* set caps and activate */
2538 gst_pad_use_fixed_caps (stream->channelpad[0]);
2539 gst_pad_set_active (stream->channelpad[0], TRUE);
2541 *outpad = gst_object_ref (stream->channelpad[0]);
2543 GST_DEBUG_OBJECT (src, "using manager source pad");
2545 template = gst_static_pad_template_get (&anysrctemplate);
2547 /* allocate pads for sending the channel data into the manager */
2548 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
2549 gst_pad_link (pad0, stream->channelpad[0]);
2550 gst_object_unref (stream->channelpad[0]);
2551 stream->channelpad[0] = pad0;
2552 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
2553 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
2554 gst_pad_set_element_private (pad0, src);
2555 gst_pad_set_active (pad0, TRUE);
2557 if (stream->channelpad[1]) {
2558 /* if we have a sinkpad for the other channel, create a pad and link to the
2560 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
2561 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
2562 gst_pad_link (pad1, stream->channelpad[1]);
2563 gst_object_unref (stream->channelpad[1]);
2564 stream->channelpad[1] = pad1;
2565 gst_pad_set_active (pad1, TRUE);
2567 gst_object_unref (template);
2569 /* setup RTCP transport back to the server if we have to. */
2570 if (src->manager && src->do_rtcp) {
2573 template = gst_static_pad_template_get (&anysinktemplate);
2575 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
2576 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
2577 gst_pad_set_element_private (stream->rtcppad, stream);
2578 gst_pad_set_active (stream->rtcppad, TRUE);
2580 /* get session RTCP pad */
2581 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
2582 pad = gst_element_get_request_pad (src->manager, name);
2587 gst_pad_link (pad, stream->rtcppad);
2588 gst_object_unref (pad);
2591 gst_object_unref (template);
2597 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
2598 GstRTSPTransport * transport, const gchar ** destination, gint * min,
2599 gint * max, guint * ttl)
2601 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
2603 if (!(*destination = transport->destination))
2604 *destination = stream->destination;
2607 /* transport first */
2608 *min = transport->port.min;
2609 *max = transport->port.max;
2610 if (*min == -1 && *max == -1) {
2611 /* then try from SDP */
2612 if (stream->port != 0) {
2613 *min = stream->port;
2614 *max = stream->port + 1;
2620 if (!(*ttl = transport->ttl))
2625 /* first take the source, then the endpoint to figure out where to send
2627 if (!(*destination = transport->source)) {
2628 if (src->conninfo.connection)
2629 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
2630 else if (stream->conninfo.connection)
2632 gst_rtsp_connection_get_ip (stream->conninfo.connection);
2636 /* for unicast we only expect the ports here */
2637 *min = transport->server_port.min;
2638 *max = transport->server_port.max;
2643 /* For multicast create UDP sources and join the multicast group. */
2645 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
2646 GstRTSPTransport * transport, GstPad ** outpad)
2649 const gchar *destination;
2652 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
2654 /* we can remove the allocated UDP ports now */
2655 gst_rtspsrc_stream_free_udp (stream);
2657 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
2660 /* we need a destination now */
2661 if (destination == NULL)
2662 goto no_destination;
2664 /* we really need ports now or we won't be able to receive anything at all */
2665 if (min == -1 && max == -1)
2668 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
2669 destination, min, max);
2671 /* creating UDP source for RTP */
2673 uri = g_strdup_printf ("udp://%s:%d", destination, min);
2674 stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
2676 if (stream->udpsrc[0] == NULL)
2679 /* take ownership */
2680 gst_object_ref_sink (stream->udpsrc[0]);
2683 gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED);
2686 /* creating another UDP source for RTCP */
2688 uri = g_strdup_printf ("udp://%s:%d", destination, max);
2689 stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
2691 if (stream->udpsrc[1] == NULL)
2694 /* take ownership */
2695 gst_object_ref_sink (stream->udpsrc[1]);
2697 gst_element_set_state (stream->udpsrc[1], GST_STATE_PAUSED);
2704 GST_DEBUG_OBJECT (src, "no UDP source element found");
2709 GST_DEBUG_OBJECT (src, "no destination found");
2714 GST_DEBUG_OBJECT (src, "no ports found");
2719 /* configure the remainder of the UDP ports */
2721 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
2722 GstRTSPTransport * transport, GstPad ** outpad)
2724 /* we manage the UDP elements now. For unicast, the UDP sources where
2725 * allocated in the stream when we suggested a transport. */
2726 if (stream->udpsrc[0]) {
2727 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
2729 GST_DEBUG_OBJECT (src, "setting up UDP source");
2731 /* configure a timeout on the UDP port. When the timeout message is
2732 * posted, we assume UDP transport is not possible. We reconnect using TCP
2734 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->udp_timeout,
2737 /* get output pad of the UDP source. */
2738 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
2740 /* save it so we can unblock */
2741 stream->blockedpad = *outpad;
2743 /* configure pad block on the pad. As soon as there is dataflow on the
2744 * UDP source, we know that UDP is not blocked by a firewall and we can
2745 * configure all the streams to let the application autoplug decoders. */
2747 gst_pad_add_probe (stream->blockedpad,
2748 GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, pad_blocked, src, NULL);
2750 if (stream->channelpad[0]) {
2751 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
2752 /* configure for UDP delivery, we need to connect the UDP pads to
2753 * the session plugin. */
2754 gst_pad_link (*outpad, stream->channelpad[0]);
2755 gst_object_unref (*outpad);
2757 /* we connected to pad-added signal to get pads from the manager */
2759 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
2764 if (stream->udpsrc[1]) {
2765 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
2767 if (stream->channelpad[1]) {
2770 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
2772 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
2773 gst_pad_link (pad, stream->channelpad[1]);
2774 gst_object_unref (pad);
2776 /* leave unlinked */
2782 /* configure the UDP sink back to the server for status reports */
2784 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
2785 GstRTSPStream * stream, GstRTSPTransport * transport)
2788 gint rtp_port, rtcp_port;
2789 gboolean do_rtp, do_rtcp;
2790 const gchar *destination;
2795 /* get transport info */
2796 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
2797 &rtp_port, &rtcp_port, &ttl);
2799 /* see what we need to do */
2800 do_rtp = (rtp_port != -1);
2801 /* it's possible that the server does not want us to send RTCP in which case
2803 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
2805 /* we need a destination when we have RTP or RTCP ports */
2806 if (destination == NULL && (do_rtp || do_rtcp))
2807 goto no_destination;
2809 /* try to construct the fakesrc to the RTP port of the server to open up any
2812 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
2815 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
2816 stream->udpsink[0] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
2818 if (stream->udpsink[0] == NULL)
2819 goto no_sink_element;
2821 /* don't join multicast group, we will have the source socket do that */
2822 /* no sync or async state changes needed */
2823 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
2824 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
2826 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
2828 if (stream->udpsrc[0]) {
2829 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
2830 * so that NAT firewalls will open a hole for us */
2831 g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
2832 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
2833 /* configure socket and make sure udpsink does not close it when shutting
2834 * down, it belongs to udpsrc after all. */
2835 g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
2836 "close-socket", FALSE, NULL);
2837 g_object_unref (socket);
2840 /* the source for the dummy packets to open up NAT */
2841 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
2842 if (stream->fakesrc == NULL)
2843 goto no_fakesrc_element;
2845 /* random data in 5 buffers, a size of 200 bytes should be fine */
2846 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
2847 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
2849 /* we don't want to consider this a sink */
2850 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
2852 /* keep everything locked */
2853 gst_element_set_locked_state (stream->udpsink[0], TRUE);
2854 gst_element_set_locked_state (stream->fakesrc, TRUE);
2856 gst_object_ref (stream->udpsink[0]);
2857 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
2858 gst_object_ref (stream->fakesrc);
2859 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
2861 gst_element_link (stream->fakesrc, stream->udpsink[0]);
2864 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
2867 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
2868 stream->udpsink[1] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
2870 if (stream->udpsink[1] == NULL)
2871 goto no_sink_element;
2873 /* don't join multicast group, we will have the source socket do that */
2874 /* no sync or async state changes needed */
2875 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
2876 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
2878 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
2880 if (stream->udpsrc[1]) {
2881 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
2882 * because some servers check the port number of where it sends RTCP to identify
2883 * the RTCP packets it receives */
2884 g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
2885 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
2886 /* configure socket and make sure udpsink does not close it when shutting
2887 * down, it belongs to udpsrc after all. */
2888 g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
2889 "close-socket", FALSE, NULL);
2890 g_object_unref (socket);
2893 /* we don't want to consider this a sink */
2894 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
2896 /* we keep this playing always */
2897 gst_element_set_locked_state (stream->udpsink[1], TRUE);
2898 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
2900 gst_object_ref (stream->udpsink[1]);
2901 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
2903 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
2905 /* get session RTCP pad */
2906 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
2907 pad = gst_element_get_request_pad (src->manager, name);
2912 gst_pad_link (pad, stream->rtcppad);
2913 gst_object_unref (pad);
2922 GST_DEBUG_OBJECT (src, "no destination address specified");
2927 GST_DEBUG_OBJECT (src, "no UDP sink element found");
2932 GST_DEBUG_OBJECT (src, "no fakesrc element found");
2937 /* sets up all elements needed for streaming over the specified transport.
2938 * Does not yet expose the element pads, this will be done when there is actuall
2939 * dataflow detected, which might never happen when UDP is blocked in a
2940 * firewall, for example.
2943 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
2944 GstRTSPTransport * transport)
2947 GstPad *outpad = NULL;
2948 GstPadTemplate *template;
2953 src = stream->parent;
2955 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
2957 s = gst_caps_get_structure (stream->caps, 0);
2959 /* get the proper mime type for this stream now */
2960 if (gst_rtsp_transport_get_mime (transport->trans, &mime) < 0)
2961 goto unknown_transport;
2963 goto unknown_transport;
2965 /* configure the final mime type */
2966 GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
2967 gst_structure_set_name (s, mime);
2969 /* try to get and configure a manager, channelpad[0-1] will be configured with
2970 * the pads for the manager, or NULL when no manager is needed. */
2971 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
2974 switch (transport->lower_transport) {
2975 case GST_RTSP_LOWER_TRANS_TCP:
2976 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
2977 goto transport_failed;
2979 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
2980 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
2981 goto transport_failed;
2982 /* fallthrough, the rest is the same for UDP and MCAST */
2983 case GST_RTSP_LOWER_TRANS_UDP:
2984 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
2985 goto transport_failed;
2986 /* configure udpsinks back to the server for RTCP messages and for the
2987 * dummy RTP messages to open NAT. */
2988 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
2989 goto transport_failed;
2992 goto unknown_transport;
2996 GST_DEBUG_OBJECT (src, "creating ghostpad");
2998 gst_pad_use_fixed_caps (outpad);
3000 /* create ghostpad, don't add just yet, this will be done when we activate
3002 name = g_strdup_printf ("stream_%u", stream->id);
3003 template = gst_static_pad_template_get (&rtptemplate);
3004 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
3005 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
3006 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
3007 gst_object_unref (template);
3010 gst_object_unref (outpad);
3012 /* mark pad as ok */
3013 stream->last_ret = GST_FLOW_OK;
3020 GST_DEBUG_OBJECT (src, "failed to configure transport");
3025 GST_DEBUG_OBJECT (src, "unknown transport");
3030 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3035 /* send a couple of dummy random packets on the receiver RTP port to the server,
3036 * this should make a firewall think we initiated the data transfer and
3037 * hopefully allow packets to go from the sender port to our RTP receiver port */
3039 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
3043 if (src->nat_method != GST_RTSP_NAT_DUMMY)
3046 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3047 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3049 if (stream->fakesrc && stream->udpsink[0]) {
3050 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
3051 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
3052 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
3053 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
3054 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
3060 /* Adds the source pads of all configured streams to the element.
3061 * This code is performed when we detected dataflow.
3063 * We detect dataflow from either the _loop function or with pad probes on the
3067 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
3071 GST_DEBUG_OBJECT (src, "activating streams");
3073 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3074 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3076 if (stream->udpsrc[0]) {
3077 /* remove timeout, we are streaming now and timeouts will be handled by
3078 * the session manager and jitter buffer */
3079 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
3081 if (stream->srcpad) {
3082 /* if we don't have a session manager, set the caps now. If we have a
3083 * session, we will get a notification of the pad and the caps. */
3084 if (!src->manager) {
3085 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
3086 gst_pad_set_caps (stream->srcpad, stream->caps);
3089 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
3090 gst_pad_set_active (stream->srcpad, TRUE);
3092 if (!stream->added) {
3093 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
3094 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
3095 stream->added = TRUE;
3100 /* unblock all pads */
3101 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3102 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3104 if (stream->blockid) {
3105 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
3106 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
3107 stream->blockid = 0;
3115 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment)
3118 guint64 start, stop;
3119 gdouble play_speed, play_scale;
3121 GST_DEBUG_OBJECT (src, "configuring stream caps");
3123 start = segment->position;
3124 stop = segment->duration;
3125 play_speed = segment->rate;
3126 play_scale = segment->applied_rate;
3128 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3129 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3132 if ((caps = stream->caps)) {
3133 caps = gst_caps_make_writable (caps);
3135 if (stream->timebase != -1)
3136 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
3137 (guint) stream->timebase, NULL);
3138 if (stream->seqbase != -1)
3139 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
3140 (guint) stream->seqbase, NULL);
3141 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
3143 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
3144 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
3145 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
3147 stream->caps = caps;
3149 GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps);
3152 GST_DEBUG_OBJECT (src, "clear session");
3153 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
3157 static GstFlowReturn
3158 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
3163 /* store the value */
3164 stream->last_ret = ret;
3166 /* if it's success we can return the value right away */
3167 if (ret == GST_FLOW_OK)
3170 /* any other error that is not-linked can be returned right
3172 if (ret != GST_FLOW_NOT_LINKED)
3175 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
3176 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3177 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3179 ret = ostream->last_ret;
3180 /* some other return value (must be SUCCESS but we can return
3181 * other values as well) */
3182 if (ret != GST_FLOW_NOT_LINKED)
3185 /* if we get here, all other pads were unlinked and we return
3186 * NOT_LINKED then */
3192 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
3193 GstEvent * event, gboolean source)
3195 gboolean res = TRUE;
3197 /* only streams that have a connection to the outside world */
3198 if (stream->srcpad == NULL)
3201 if (source && stream->udpsrc[0]) {
3202 gst_event_ref (event);
3203 res = gst_element_send_event (stream->udpsrc[0], event);
3204 } else if (stream->channelpad[0]) {
3205 gst_event_ref (event);
3206 if (GST_PAD_IS_SRC (stream->channelpad[0]))
3207 res = gst_pad_push_event (stream->channelpad[0], event);
3209 res = gst_pad_send_event (stream->channelpad[0], event);
3212 if (source && stream->udpsrc[1]) {
3213 gst_event_ref (event);
3214 res &= gst_element_send_event (stream->udpsrc[1], event);
3215 } else if (stream->channelpad[1]) {
3216 gst_event_ref (event);
3217 if (GST_PAD_IS_SRC (stream->channelpad[1]))
3218 res &= gst_pad_push_event (stream->channelpad[1], event);
3220 res &= gst_pad_send_event (stream->channelpad[1], event);
3224 gst_event_unref (event);
3230 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event, gboolean source)
3233 gboolean res = TRUE;
3235 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3236 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3238 gst_event_ref (event);
3239 res &= gst_rtspsrc_stream_push_event (src, ostream, event, source);
3241 gst_event_unref (event);
3246 static GstRTSPResult
3247 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3252 if (info->connection == NULL) {
3253 if (info->url == NULL) {
3254 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
3255 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
3259 /* create connection */
3260 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
3261 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
3262 goto could_not_create;
3265 g_free (info->url_str);
3266 info->url_str = gst_rtsp_url_get_request_uri (info->url);
3268 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
3270 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
3271 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
3273 if (src->proxy_host) {
3274 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
3276 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
3281 if (!info->connected) {
3284 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
3285 ("Connecting to %s", info->location));
3286 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
3288 gst_rtsp_connection_connect (info->connection,
3289 src->ptcp_timeout)) < 0)
3290 goto could_not_connect;
3292 info->connected = TRUE;
3299 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
3304 gchar *str = gst_rtsp_strresult (res);
3305 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
3311 gchar *str = gst_rtsp_strresult (res);
3312 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
3318 static GstRTSPResult
3319 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
3322 if (info->connected) {
3323 GST_DEBUG_OBJECT (src, "closing connection...");
3324 gst_rtsp_connection_close (info->connection);
3325 info->connected = FALSE;
3327 if (free && info->connection) {
3328 /* free connection */
3329 GST_DEBUG_OBJECT (src, "freeing connection...");
3330 gst_rtsp_connection_free (info->connection);
3331 info->connection = NULL;
3336 static GstRTSPResult
3337 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3342 GST_DEBUG_OBJECT (src, "reconnecting connection...");
3343 gst_rtsp_conninfo_close (src, info, FALSE);
3344 res = gst_rtsp_conninfo_connect (src, info, async);
3350 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
3354 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
3355 if (src->conninfo.connection) {
3356 GST_DEBUG_OBJECT (src, "connection flush");
3357 gst_rtsp_connection_flush (src->conninfo.connection, flush);
3359 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3360 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3361 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
3362 if (stream->conninfo.connection)
3363 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
3367 /* FIXME, handle server request, reply with OK, for now */
3368 static GstRTSPResult
3369 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
3370 GstRTSPMessage * request)
3372 GstRTSPMessage response = { 0 };
3375 GST_DEBUG_OBJECT (src, "got server request message");
3378 gst_rtsp_message_dump (request);
3380 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
3382 if (res == GST_RTSP_ENOTIMPL) {
3383 /* default implementation, send OK */
3385 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
3390 GST_DEBUG_OBJECT (src, "replying with OK");
3393 gst_rtsp_message_dump (&response);
3395 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
3399 gst_rtsp_message_unset (&response);
3400 } else if (res == GST_RTSP_EEOF)
3408 gst_rtsp_message_unset (&response);
3413 /* send server keep-alive */
3414 static GstRTSPResult
3415 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
3417 GstRTSPMessage request = { 0 };
3419 GstRTSPMethod method;
3422 GST_DEBUG_OBJECT (src, "creating server keep-alive");
3424 /* find a method to use for keep-alive */
3425 if (src->methods & GST_RTSP_GET_PARAMETER)
3426 method = GST_RTSP_GET_PARAMETER;
3428 method = GST_RTSP_OPTIONS;
3431 control = src->control;
3433 control = src->conninfo.url_str;
3435 if (control == NULL)
3438 res = gst_rtsp_message_init_request (&request, method, control);
3443 gst_rtsp_message_dump (&request);
3446 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
3451 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
3452 gst_rtsp_message_unset (&request);
3459 GST_WARNING_OBJECT (src, "no control url to send keepalive");
3464 gchar *str = gst_rtsp_strresult (res);
3466 gst_rtsp_message_unset (&request);
3467 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
3468 ("Could not send keep-alive. (%s)", str));
3474 static GstFlowReturn
3475 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
3477 GstRTSPMessage message = { 0 };
3480 GstRTSPStream *stream;
3481 GstPad *outpad = NULL;
3484 GstFlowReturn ret = GST_FLOW_OK;
3486 gboolean is_rtcp, have_data;
3488 /* here we are only interested in data messages */
3491 GTimeVal tv_timeout;
3493 /* get the next timeout interval */
3494 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3496 /* see if the timeout period expired */
3497 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
3498 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
3499 /* send keep-alive, only act on interrupt, a warning will be posted for
3501 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3503 /* get new timeout */
3504 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3507 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
3508 tv_timeout.tv_sec, tv_timeout.tv_usec);
3510 /* protect the connection with the connection lock so that we can see when
3511 * we are finished doing server communication */
3513 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
3514 &message, src->ptcp_timeout);
3518 GST_DEBUG_OBJECT (src, "we received a server message");
3520 case GST_RTSP_EINTR:
3521 /* we got interrupted this means we need to stop */
3523 case GST_RTSP_ETIMEOUT:
3524 /* no reply, send keep alive */
3525 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
3526 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3530 /* go EOS when the server closed the connection */
3536 switch (message.type) {
3537 case GST_RTSP_MESSAGE_REQUEST:
3538 /* server sends us a request message, handle it */
3540 gst_rtspsrc_handle_request (src, src->conninfo.connection,
3542 if (res == GST_RTSP_EEOF)
3545 goto handle_request_failed;
3547 case GST_RTSP_MESSAGE_RESPONSE:
3548 /* we ignore response messages */
3549 GST_DEBUG_OBJECT (src, "ignoring response message");
3551 gst_rtsp_message_dump (&message);
3553 case GST_RTSP_MESSAGE_DATA:
3554 GST_DEBUG_OBJECT (src, "got data message");
3558 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
3565 channel = message.type_data.data.channel;
3567 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
3569 goto unknown_stream;
3571 if (channel == stream->channel[0]) {
3572 outpad = stream->channelpad[0];
3574 } else if (channel == stream->channel[1]) {
3575 outpad = stream->channelpad[1];
3581 /* take a look at the body to figure out what we have */
3582 gst_rtsp_message_get_body (&message, &data, &size);
3584 goto invalid_length;
3586 /* channels are not correct on some servers, do extra check */
3587 if (data[1] >= 200 && data[1] <= 204) {
3588 /* hmm RTCP message switch to the RTCP pad of the same stream. */
3589 outpad = stream->channelpad[1];
3593 /* we have no clue what this is, just ignore then. */
3595 goto unknown_stream;
3597 /* take the message body for further processing */
3598 gst_rtsp_message_steal_body (&message, &data, &size);
3600 /* strip the trailing \0 */
3603 buf = gst_buffer_new ();
3604 gst_buffer_take_memory (buf, -1,
3605 gst_memory_new_wrapped (0, data, g_free, size, 0, size));
3607 /* don't need message anymore */
3608 gst_rtsp_message_unset (&message);
3610 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
3613 if (src->need_activate) {
3614 gst_rtspsrc_activate_streams (src);
3615 src->need_activate = FALSE;
3618 if (src->base_time == -1) {
3619 /* Take current running_time. This timestamp will be put on
3620 * the first buffer of each stream because we are a live source and so we
3621 * timestamp with the running_time. When we are dealing with TCP, we also
3622 * only timestamp the first buffer (using the DISCONT flag) because a server
3623 * typically bursts data, for which we don't want to compensate by speeding
3624 * up the media. The other timestamps will be interpollated from this one
3625 * using the RTP timestamps. */
3626 GST_OBJECT_LOCK (src);
3627 if (GST_ELEMENT_CLOCK (src)) {
3629 GstClockTime base_time;
3631 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
3632 base_time = GST_ELEMENT_CAST (src)->base_time;
3634 src->base_time = now - base_time;
3636 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
3637 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
3639 GST_OBJECT_UNLOCK (src);
3642 if (stream->discont && !is_rtcp) {
3643 /* mark first RTP buffer as discont */
3644 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
3645 stream->discont = FALSE;
3646 /* first buffer gets the timestamp, other buffers are not timestamped and
3647 * their presentation time will be interpollated from the rtp timestamps. */
3648 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
3649 GST_TIME_ARGS (src->base_time));
3651 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
3654 /* chain to the peer pad */
3655 if (GST_PAD_IS_SINK (outpad))
3656 ret = gst_pad_chain (outpad, buf);
3658 ret = gst_pad_push (outpad, buf);
3661 /* combine all stream flows for the data transport */
3662 ret = gst_rtspsrc_combine_flows (src, stream, ret);
3669 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
3670 gst_rtsp_message_unset (&message);
3675 GST_DEBUG_OBJECT (src, "we got an eof from the server");
3676 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3677 ("The server closed the connection."));
3678 src->conninfo.connected = FALSE;
3679 gst_rtsp_message_unset (&message);
3680 return GST_FLOW_EOS;
3684 gst_rtsp_message_unset (&message);
3685 GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
3686 gst_rtspsrc_connection_flush (src, FALSE);
3687 return GST_FLOW_WRONG_STATE;
3691 gchar *str = gst_rtsp_strresult (res);
3693 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3694 ("Could not receive message. (%s)", str));
3697 gst_rtsp_message_unset (&message);
3698 return GST_FLOW_ERROR;
3700 handle_request_failed:
3702 gchar *str = gst_rtsp_strresult (res);
3704 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
3705 ("Could not handle server message. (%s)", str));
3707 gst_rtsp_message_unset (&message);
3708 return GST_FLOW_ERROR;
3712 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3713 ("Short message received, ignoring."));
3714 gst_rtsp_message_unset (&message);
3719 static GstFlowReturn
3720 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
3723 GstRTSPMessage message = { 0 };
3727 GTimeVal tv_timeout;
3729 /* get the next timeout interval */
3730 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3732 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
3733 (gint) tv_timeout.tv_sec);
3735 gst_rtsp_message_unset (&message);
3737 /* we should continue reading the TCP socket because the server might
3738 * send us requests. When the session timeout expires, we need to send a
3739 * keep-alive request to keep the session open. */
3740 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
3741 &message, &tv_timeout);
3745 GST_DEBUG_OBJECT (src, "we received a server message");
3747 case GST_RTSP_EINTR:
3748 /* we got interrupted, see what we have to do */
3750 case GST_RTSP_ETIMEOUT:
3751 /* send keep-alive, ignore the result, a warning will be posted. */
3752 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
3753 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3757 /* server closed the connection. not very fatal for UDP, reconnect and
3758 * see what happens. */
3759 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3760 ("The server closed the connection."));
3762 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
3770 switch (message.type) {
3771 case GST_RTSP_MESSAGE_REQUEST:
3772 /* server sends us a request message, handle it */
3774 gst_rtspsrc_handle_request (src, src->conninfo.connection,
3776 if (res == GST_RTSP_EEOF)
3779 goto handle_request_failed;
3781 case GST_RTSP_MESSAGE_RESPONSE:
3782 /* we ignore response and data messages */
3783 GST_DEBUG_OBJECT (src, "ignoring response message");
3785 gst_rtsp_message_dump (&message);
3786 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
3787 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
3788 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
3789 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
3790 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3797 case GST_RTSP_MESSAGE_DATA:
3798 /* we ignore response and data messages */
3799 GST_DEBUG_OBJECT (src, "ignoring data message");
3802 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
3808 /* we get here when the connection got interrupted */
3811 gst_rtsp_message_unset (&message);
3812 GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
3813 gst_rtspsrc_connection_flush (src, FALSE);
3814 return GST_FLOW_WRONG_STATE;
3818 gchar *str = gst_rtsp_strresult (res);
3821 src->conninfo.connected = FALSE;
3822 if (res != GST_RTSP_EINTR) {
3823 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
3824 ("Could not connect to server. (%s)", str));
3826 ret = GST_FLOW_ERROR;
3828 ret = GST_FLOW_WRONG_STATE;
3834 gchar *str = gst_rtsp_strresult (res);
3836 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3837 ("Could not receive message. (%s)", str));
3839 return GST_FLOW_ERROR;
3841 handle_request_failed:
3843 gchar *str = gst_rtsp_strresult (res);
3846 gst_rtsp_message_unset (&message);
3847 if (res != GST_RTSP_EINTR) {
3848 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
3849 ("Could not handle server message. (%s)", str));
3851 ret = GST_FLOW_ERROR;
3853 ret = GST_FLOW_WRONG_STATE;
3859 GST_DEBUG_OBJECT (src, "we got an eof from the server");
3860 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3861 ("The server closed the connection."));
3862 src->conninfo.connected = FALSE;
3863 gst_rtsp_message_unset (&message);
3864 return GST_FLOW_EOS;
3868 static GstRTSPResult
3869 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
3871 GstRTSPResult res = GST_RTSP_OK;
3874 GST_DEBUG_OBJECT (src, "doing reconnect");
3876 GST_OBJECT_LOCK (src);
3877 /* only restart when the pads were not yet activated, else we were
3878 * streaming over UDP */
3879 restart = src->need_activate;
3880 GST_OBJECT_UNLOCK (src);
3882 /* no need to restart, we're done */
3886 /* we can try only TCP now */
3887 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
3889 /* close and cleanup our state */
3890 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
3893 /* see if we have TCP left to try. Also don't try TCP when we were configured
3895 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
3898 /* We post a warning message now to inform the user
3899 * that nothing happened. It's most likely a firewall thing. */
3900 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3901 ("Could not receive any UDP packets for %.4f seconds, maybe your "
3902 "firewall is blocking it. Retrying using a TCP connection.",
3903 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
3905 /* open new connection using tcp */
3906 if (gst_rtspsrc_open (src, async) < 0)
3909 /* start playback */
3910 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
3919 src->cur_protocols = 0;
3920 /* no transport possible, post an error and stop */
3921 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3922 ("Could not receive any UDP packets for %.4f seconds, maybe your "
3923 "firewall is blocking it. No other protocols to try.",
3924 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
3925 return GST_FLOW_ERROR;
3929 GST_DEBUG_OBJECT (src, "open failed");
3934 GST_DEBUG_OBJECT (src, "play failed");
3940 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
3944 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
3947 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
3950 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
3953 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
3961 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
3965 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
3968 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
3971 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
3974 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
3982 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
3986 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
3989 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
3992 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
3995 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
4003 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
4007 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
4010 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
4013 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
4016 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
4024 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
4026 if (ret == GST_RTSP_OK)
4027 gst_rtspsrc_loop_complete_cmd (src, cmd);
4028 else if (ret == GST_RTSP_EINTR)
4029 gst_rtspsrc_loop_cancel_cmd (src, cmd);
4031 gst_rtspsrc_loop_error_cmd (src, cmd);
4035 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd)
4039 /* start new request */
4040 gst_rtspsrc_loop_start_cmd (src, cmd);
4042 GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
4044 GST_OBJECT_LOCK (src);
4045 old = src->loop_cmd;
4046 if (old != CMD_WAIT) {
4047 src->loop_cmd = CMD_WAIT;
4048 GST_OBJECT_UNLOCK (src);
4049 /* cancel previous request */
4050 gst_rtspsrc_loop_cancel_cmd (src, old);
4051 GST_OBJECT_LOCK (src);
4053 src->loop_cmd = cmd;
4054 /* interrupt if allowed */
4056 GST_DEBUG_OBJECT (src, "start connection flush");
4057 gst_rtspsrc_connection_flush (src, TRUE);
4060 gst_task_start (src->task);
4061 GST_OBJECT_UNLOCK (src);
4065 gst_rtspsrc_loop (GstRTSPSrc * src)
4069 if (!src->conninfo.connection || !src->conninfo.connected)
4072 if (src->interleaved)
4073 ret = gst_rtspsrc_loop_interleaved (src);
4075 ret = gst_rtspsrc_loop_udp (src);
4077 if (ret != GST_FLOW_OK)
4085 GST_WARNING_OBJECT (src, "we are not connected");
4086 ret = GST_FLOW_WRONG_STATE;
4091 const gchar *reason = gst_flow_get_name (ret);
4093 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
4094 src->running = FALSE;
4095 if (ret == GST_FLOW_EOS) {
4096 /* perform EOS logic */
4097 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
4098 gst_element_post_message (GST_ELEMENT_CAST (src),
4099 gst_message_new_segment_done (GST_OBJECT_CAST (src),
4100 src->segment.format, src->segment.position));
4102 gst_rtspsrc_push_event (src, gst_event_new_eos (), FALSE);
4104 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
4105 /* for fatal errors we post an error message, post the error before the
4106 * EOS so the app knows about the error first. */
4107 GST_ELEMENT_ERROR (src, STREAM, FAILED,
4108 ("Internal data flow error."),
4109 ("streaming task paused, reason %s (%d)", reason, ret));
4110 gst_rtspsrc_push_event (src, gst_event_new_eos (), FALSE);
4116 #ifndef GST_DISABLE_GST_DEBUG
4117 static const gchar *
4118 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
4122 while (method != 0) {
4139 static const gchar *
4140 gst_rtspsrc_skip_lws (const gchar * s)
4142 while (g_ascii_isspace (*s))
4147 static const gchar *
4148 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
4150 while (s > start && g_ascii_isspace (*(s - 1)))
4155 static const gchar *
4156 gst_rtspsrc_skip_commas (const gchar * s)
4158 /* The grammar allows for multiple commas */
4159 while (g_ascii_isspace (*s) || *s == ',')
4164 static const gchar *
4165 gst_rtspsrc_skip_item (const gchar * s)
4167 gboolean quoted = FALSE;
4168 const gchar *start = s;
4170 /* A list item ends at the last non-whitespace character
4171 * before a comma which is not inside a quoted-string. Or at
4172 * the end of the string.
4178 if (*s == '\\' && *(s + 1))
4187 return gst_rtspsrc_unskip_lws (s, start);
4191 gst_rtsp_decode_quoted_string (gchar * quoted_string)
4195 src = quoted_string + 1;
4196 dst = quoted_string;
4197 while (*src && *src != '"') {
4198 if (*src == '\\' && *(src + 1))
4205 /* Extract the authentication tokens that the server provided for each method
4206 * into an array of structures and give those to the connection object.
4209 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
4210 const gchar * header, gboolean * stale)
4212 GSList *list = NULL, *iter;
4214 gchar *item, *eq, *name_end, *value;
4216 g_return_if_fail (stale != NULL);
4218 gst_rtsp_connection_clear_auth_params (conn);
4221 /* Parse a header whose content is described by RFC2616 as
4222 * "#something", where "something" does not itself contain commas,
4223 * except as part of quoted-strings, into a list of allocated strings.
4225 header = gst_rtspsrc_skip_commas (header);
4227 end = gst_rtspsrc_skip_item (header);
4228 list = g_slist_prepend (list, g_strndup (header, end - header));
4229 header = gst_rtspsrc_skip_commas (end);
4234 list = g_slist_reverse (list);
4235 for (iter = list; iter; iter = iter->next) {
4238 eq = strchr (item, '=');
4240 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
4241 if (name_end == item) {
4242 /* That's no good... */
4249 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
4251 gst_rtsp_decode_quoted_string (value);
4255 if (item && (strcmp (item, "stale") == 0) &&
4256 value && (strcmp (value, "TRUE") == 0))
4258 gst_rtsp_connection_set_auth_param (conn, item, value);
4262 g_slist_free (list);
4265 /* Parse a WWW-Authenticate Response header and determine the
4266 * available authentication methods
4268 * This code should also cope with the fact that each WWW-Authenticate
4269 * header can contain multiple challenge methods + tokens
4271 * At the moment, for Basic auth, we just do a minimal check and don't
4272 * even parse out the realm */
4274 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
4275 GstRTSPConnection * conn, gboolean * stale)
4279 g_return_if_fail (hdr != NULL);
4280 g_return_if_fail (methods != NULL);
4281 g_return_if_fail (stale != NULL);
4283 /* Skip whitespace at the start of the string */
4284 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
4286 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
4287 *methods |= GST_RTSP_AUTH_BASIC;
4288 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
4289 *methods |= GST_RTSP_AUTH_DIGEST;
4290 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
4295 * gst_rtspsrc_setup_auth:
4296 * @src: the rtsp source
4298 * Configure a username and password and auth method on the
4299 * connection object based on a response we received from the
4302 * Currently, this requires that a username and password were supplied
4303 * in the uri. In the future, they may be requested on demand by sending
4304 * a message up the bus.
4306 * Returns: TRUE if authentication information could be set up correctly.
4309 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
4313 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
4314 GstRTSPAuthMethod method;
4315 GstRTSPResult auth_result;
4317 GstRTSPConnection *conn;
4319 gboolean stale = FALSE;
4321 conn = src->conninfo.connection;
4323 /* Identify the available auth methods and see if any are supported */
4324 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
4325 &hdr, 0) == GST_RTSP_OK) {
4326 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
4329 if (avail_methods == GST_RTSP_AUTH_NONE)
4330 goto no_auth_available;
4332 /* For digest auth, if the response indicates that the session
4333 * data are stale, we just update them in the connection object and
4334 * return TRUE to retry the request */
4336 src->tried_url_auth = FALSE;
4338 url = gst_rtsp_connection_get_url (conn);
4340 /* Do we have username and password available? */
4341 if (url != NULL && !src->tried_url_auth && url->user != NULL
4342 && url->passwd != NULL) {
4345 src->tried_url_auth = TRUE;
4346 GST_DEBUG_OBJECT (src,
4347 "Attempting authentication using credentials from the URL");
4349 user = src->user_id;
4350 pass = src->user_pw;
4351 GST_DEBUG_OBJECT (src,
4352 "Attempting authentication using credentials from the properties");
4355 /* FIXME: If the url didn't contain username and password or we tried them
4356 * already, request a username and passwd from the application via some kind
4357 * of credentials request message */
4359 /* If we don't have a username and passwd at this point, bail out. */
4360 if (user == NULL || pass == NULL)
4363 /* Try to configure for each available authentication method, strongest to
4365 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
4366 /* Check if this method is available on the server */
4367 if ((method & avail_methods) == 0)
4370 /* Pass the credentials to the connection to try on the next request */
4371 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
4372 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
4373 * ignore it and end up retrying later */
4374 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
4375 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
4376 gst_rtsp_auth_method_to_string (method));
4381 if (method == GST_RTSP_AUTH_NONE)
4382 goto no_auth_available;
4388 /* Output an error indicating that we couldn't connect because there were
4389 * no supported authentication protocols */
4390 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4391 ("No supported authentication protocol was found"));
4396 /* We don't fire an error message, we just return FALSE and let the
4397 * normal NOT_AUTHORIZED error be propagated */
4402 static GstRTSPResult
4403 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
4404 GstRTSPMessage * request, GstRTSPMessage * response,
4405 GstRTSPStatusCode * code)
4408 GstRTSPStatusCode thecode;
4409 gchar *content_base = NULL;
4413 if (!src->short_header)
4414 gst_rtsp_ext_list_before_send (src->extensions, request);
4416 GST_DEBUG_OBJECT (src, "sending message");
4419 gst_rtsp_message_dump (request);
4421 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
4425 gst_rtsp_connection_reset_timeout (conn);
4428 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
4433 gst_rtsp_message_dump (response);
4435 switch (response->type) {
4436 case GST_RTSP_MESSAGE_REQUEST:
4437 res = gst_rtspsrc_handle_request (src, conn, response);
4438 if (res == GST_RTSP_EEOF)
4441 goto handle_request_failed;
4443 case GST_RTSP_MESSAGE_RESPONSE:
4444 /* ok, a response is good */
4445 GST_DEBUG_OBJECT (src, "received response message");
4447 case GST_RTSP_MESSAGE_DATA:
4448 /* get next response */
4449 GST_DEBUG_OBJECT (src, "ignoring data response message");
4452 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4457 thecode = response->type_data.response.code;
4459 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
4461 /* if the caller wanted the result code, we store it. */
4465 /* If the request didn't succeed, bail out before doing any more */
4466 if (thecode != GST_RTSP_STS_OK)
4469 /* store new content base if any */
4470 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
4473 g_free (src->content_base);
4474 src->content_base = g_strdup (content_base);
4476 gst_rtsp_ext_list_after_send (src->extensions, request, response);
4483 gchar *str = gst_rtsp_strresult (res);
4485 if (res != GST_RTSP_EINTR) {
4486 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4487 ("Could not send message. (%s)", str));
4489 GST_WARNING_OBJECT (src, "send interrupted");
4498 GST_WARNING_OBJECT (src, "server closed connection, doing reconnect");
4501 /* if reconnect succeeds, try again */
4503 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
4507 /* only try once after reconnect, then fallthrough and error out */
4510 gchar *str = gst_rtsp_strresult (res);
4512 if (res != GST_RTSP_EINTR) {
4513 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4514 ("Could not receive message. (%s)", str));
4516 GST_WARNING_OBJECT (src, "receive interrupted");
4524 handle_request_failed:
4526 /* ERROR was posted */
4527 gst_rtsp_message_unset (response);
4532 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4533 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4534 ("The server closed the connection."));
4535 gst_rtsp_message_unset (response);
4542 * @src: the rtsp source
4543 * @conn: the connection to send on
4544 * @request: must point to a valid request
4545 * @response: must point to an empty #GstRTSPMessage
4546 * @code: an optional code result
4548 * send @request and retrieve the response in @response. optionally @code can be
4549 * non-NULL in which case it will contain the status code of the response.
4551 * If This function returns #GST_RTSP_OK, @response will contain a valid response
4552 * message that should be cleaned with gst_rtsp_message_unset() after usage.
4554 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
4555 * @response message) if the response code was not 200 (OK).
4557 * If the attempt results in an authentication failure, then this will attempt
4558 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
4561 * Returns: #GST_RTSP_OK if the processing was successful.
4563 static GstRTSPResult
4564 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
4565 GstRTSPMessage * request, GstRTSPMessage * response,
4566 GstRTSPStatusCode * code)
4568 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
4569 GstRTSPResult res = GST_RTSP_ERROR;
4572 GstRTSPMethod method = GST_RTSP_INVALID;
4578 /* make sure we don't loop forever */
4582 /* save method so we can disable it when the server complains */
4583 method = request->type_data.request.method;
4586 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
4590 case GST_RTSP_STS_UNAUTHORIZED:
4591 if (gst_rtspsrc_setup_auth (src, response)) {
4592 /* Try the request/response again after configuring the auth info
4600 } while (retry == TRUE);
4602 /* If the user requested the code, let them handle errors, otherwise
4603 * post an error below */
4606 else if (int_code != GST_RTSP_STS_OK)
4607 goto error_response;
4614 GST_DEBUG_OBJECT (src, "got error %d", res);
4619 res = GST_RTSP_ERROR;
4621 switch (response->type_data.response.code) {
4622 case GST_RTSP_STS_NOT_FOUND:
4623 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
4624 response->type_data.response.reason));
4626 case GST_RTSP_STS_MOVED_PERMANENTLY:
4627 case GST_RTSP_STS_MOVE_TEMPORARILY:
4629 gchar *new_location;
4630 GstRTSPLowerTrans transports;
4632 GST_DEBUG_OBJECT (src, "got redirection");
4633 /* if we don't have a Location Header, we must error */
4634 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
4635 &new_location, 0) < 0)
4638 /* When we receive a redirect result, we go back to the INIT state after
4639 * parsing the new URI. The caller should do the needed steps to issue
4640 * a new setup when it detects this state change. */
4641 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
4643 /* save current transports */
4644 if (src->conninfo.url)
4645 transports = src->conninfo.url->transports;
4647 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
4649 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
4651 /* set old transports */
4652 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
4653 src->conninfo.url->transports = transports;
4655 src->need_redirect = TRUE;
4656 src->state = GST_RTSP_STATE_INIT;
4660 case GST_RTSP_STS_NOT_ACCEPTABLE:
4661 case GST_RTSP_STS_NOT_IMPLEMENTED:
4662 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
4663 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
4664 gst_rtsp_method_as_text (method));
4665 src->methods &= ~method;
4669 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4670 ("Got error response: %d (%s).", response->type_data.response.code,
4671 response->type_data.response.reason));
4674 /* if we return ERROR we should unset the response ourselves */
4675 if (res == GST_RTSP_ERROR)
4676 gst_rtsp_message_unset (response);
4682 static GstRTSPResult
4683 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
4684 GstRTSPMessage * response, GstRTSPSrc * src)
4686 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
4691 /* parse the response and collect all the supported methods. We need this
4692 * information so that we don't try to send an unsupported request to the
4696 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
4698 GstRTSPHeaderField field;
4704 /* reset supported methods */
4707 /* Try Allow Header first */
4708 field = GST_RTSP_HDR_ALLOW;
4711 gst_rtsp_message_get_header (response, field, &respoptions, indx);
4712 if (indx == 0 && !respoptions) {
4713 /* if no Allow header was found then try the Public header... */
4714 field = GST_RTSP_HDR_PUBLIC;
4715 gst_rtsp_message_get_header (response, field, &respoptions, indx);
4720 /* If we get here, the server gave a list of supported methods, parse
4721 * them here. The string is like:
4723 * OPTIONS, DESCRIBE, ANNOUNCE, PLAY, SETUP, ...
4725 options = g_strsplit (respoptions, ",", 0);
4727 for (i = 0; options[i]; i++) {
4731 stripped = g_strstrip (options[i]);
4732 method = gst_rtsp_find_method (stripped);
4734 /* keep bitfield of supported methods */
4735 if (method != GST_RTSP_INVALID)
4736 src->methods |= method;
4738 g_strfreev (options);
4743 if (src->methods == 0) {
4744 /* neither Allow nor Public are required, assume the server supports
4745 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
4747 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
4748 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
4750 /* always assume PLAY, FIXME, extensions should be able to override
4752 src->methods |= GST_RTSP_PLAY;
4753 /* also assume it will support Range */
4754 src->seekable = TRUE;
4756 /* we need describe and setup */
4757 if (!(src->methods & GST_RTSP_DESCRIBE))
4759 if (!(src->methods & GST_RTSP_SETUP))
4767 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4768 ("Server does not support DESCRIBE."));
4773 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4774 ("Server does not support SETUP."));
4779 /* masks to be kept in sync with the hardcoded protocol order of preference
4781 static guint protocol_masks[] = {
4782 GST_RTSP_LOWER_TRANS_UDP,
4783 GST_RTSP_LOWER_TRANS_UDP_MCAST,
4784 GST_RTSP_LOWER_TRANS_TCP,
4788 static GstRTSPResult
4789 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
4790 GstRTSPLowerTrans protocols, gchar ** transports)
4794 gboolean add_udp_str;
4799 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
4804 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
4806 /* extension listed transports, use those */
4807 if (*transports != NULL)
4810 /* it's the default */
4811 add_udp_str = FALSE;
4813 /* the default RTSP transports */
4814 result = g_string_new ("");
4815 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
4816 GST_DEBUG_OBJECT (src, "adding UDP unicast");
4818 g_string_append (result, "RTP/AVP");
4820 g_string_append (result, "/UDP");
4821 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
4822 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
4823 GST_DEBUG_OBJECT (src, "adding UDP multicast");
4825 /* we don't have to allocate any UDP ports yet, if the selected transport
4826 * turns out to be multicast we can create them and join the multicast
4827 * group indicated in the transport reply */
4828 if (result->len > 0)
4829 g_string_append (result, ",");
4830 g_string_append (result, "RTP/AVP");
4832 g_string_append (result, "/UDP");
4833 g_string_append (result, ";multicast");
4834 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
4835 GST_DEBUG_OBJECT (src, "adding TCP");
4837 if (result->len > 0)
4838 g_string_append (result, ",");
4839 g_string_append (result, "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2");
4841 *transports = g_string_free (result, FALSE);
4843 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
4854 static GstRTSPResult
4855 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
4856 gint orig_rtpport, gint orig_rtcpport)
4859 gint nr_udp, nr_int;
4861 gint rtpport = 0, rtcpport = 0;
4864 src = stream->parent;
4866 /* find number of placeholders first */
4867 if (strstr (*transports, "%%i2"))
4869 else if (strstr (*transports, "%%i1"))
4874 if (strstr (*transports, "%%u2"))
4876 else if (strstr (*transports, "%%u1"))
4881 if (nr_udp == 0 && nr_int == 0)
4885 if (!orig_rtpport || !orig_rtcpport) {
4886 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
4889 rtpport = orig_rtpport;
4890 rtcpport = orig_rtcpport;
4894 str = g_string_new ("");
4896 while ((next = strstr (p, "%%"))) {
4897 g_string_append_len (str, p, next - p);
4898 if (next[2] == 'u') {
4900 g_string_append_printf (str, "%d", rtpport);
4901 else if (next[3] == '2')
4902 g_string_append_printf (str, "%d", rtcpport);
4904 if (next[2] == 'i') {
4906 g_string_append_printf (str, "%d", src->free_channel);
4907 else if (next[3] == '2')
4908 g_string_append_printf (str, "%d", src->free_channel + 1);
4913 /* append final part */
4914 g_string_append (str, p);
4916 g_free (*transports);
4917 *transports = g_string_free (str, FALSE);
4925 return GST_RTSP_ERROR;
4930 gst_rtspsrc_stream_is_real_media (GstRTSPStream * stream)
4932 gboolean res = FALSE;
4936 const gchar *enc = NULL;
4938 s = gst_caps_get_structure (stream->caps, 0);
4939 if ((enc = gst_structure_get_string (s, "encoding-name"))) {
4940 res = (strstr (enc, "-REAL") != NULL);
4946 /* Perform the SETUP request for all the streams.
4948 * We ask the server for a specific transport, which initially includes all the
4949 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
4950 * two local UDP ports that we send to the server.
4952 * Once the server replied with a transport, we configure the other streams
4953 * with the same transport.
4955 * This function will also configure the stream for the selected transport,
4956 * which basically means creating the pipeline.
4958 static GstRTSPResult
4959 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
4962 GstRTSPResult res = GST_RTSP_ERROR;
4963 GstRTSPMessage request = { 0 };
4964 GstRTSPMessage response = { 0 };
4965 GstRTSPStream *stream = NULL;
4966 GstRTSPLowerTrans protocols;
4967 GstRTSPStatusCode code;
4968 gboolean unsupported_real = FALSE;
4969 gint rtpport, rtcpport;
4973 if (src->conninfo.connection) {
4974 url = gst_rtsp_connection_get_url (src->conninfo.connection);
4975 /* we initially allow all configured lower transports. based on the URL
4976 * transports and the replies from the server we narrow them down. */
4977 protocols = url->transports & src->cur_protocols;
4980 protocols = src->cur_protocols;
4986 /* reset some state */
4987 src->free_channel = 0;
4988 src->interleaved = FALSE;
4989 src->need_activate = FALSE;
4990 /* keep track of next port number, 0 is random */
4991 src->next_port_num = src->client_port_range.min;
4992 rtpport = rtcpport = 0;
4994 if (G_UNLIKELY (src->streams == NULL))
4997 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4998 GstRTSPConnection *conn;
5003 stream = (GstRTSPStream *) walk->data;
5005 /* see if we need to configure this stream */
5006 if (!gst_rtsp_ext_list_configure_stream (src->extensions, stream->caps)) {
5007 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
5009 stream->disabled = TRUE;
5013 /* merge/overwrite global caps */
5018 s = gst_caps_get_structure (stream->caps, 0);
5020 num = gst_structure_n_fields (src->props);
5021 for (j = 0; j < num; j++) {
5025 name = gst_structure_nth_field_name (src->props, j);
5026 val = gst_structure_get_value (src->props, name);
5027 gst_structure_set_value (s, name, val);
5029 GST_DEBUG_OBJECT (src, "copied %s", name);
5033 /* skip setup if we have no URL for it */
5034 if (stream->conninfo.location == NULL) {
5035 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
5039 if (src->conninfo.connection == NULL) {
5040 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
5041 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
5044 conn = stream->conninfo.connection;
5046 conn = src->conninfo.connection;
5048 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
5049 stream->conninfo.location);
5051 /* if we have a multicast connection, only suggest multicast from now on */
5052 if (stream->is_multicast)
5053 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
5056 /* first selectable protocol */
5057 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5059 if (!protocol_masks[mask])
5063 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
5064 protocol_masks[mask]);
5065 /* create a string with first transport in line */
5067 res = gst_rtspsrc_create_transports_string (src,
5068 protocols & protocol_masks[mask], &transports);
5069 if (res < 0 || transports == NULL)
5070 goto setup_transport_failed;
5072 if (strlen (transports) == 0) {
5073 g_free (transports);
5074 GST_DEBUG_OBJECT (src, "no transports found");
5079 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
5081 /* replace placeholders with real values, this function will optionally
5082 * allocate UDP ports and other info needed to execute the setup request */
5083 res = gst_rtspsrc_prepare_transports (stream, &transports,
5084 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
5086 g_free (transports);
5087 goto setup_transport_failed;
5090 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
5092 /* create SETUP request */
5094 gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
5095 stream->conninfo.location);
5097 g_free (transports);
5098 goto create_request_failed;
5101 /* select transport, copy is made when adding to header so we can free it. */
5102 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
5103 g_free (transports);
5105 /* if the user wants a non default RTP packet size we add the blocksize
5107 if (src->rtp_blocksize > 0) {
5108 hval = g_strdup_printf ("%d", src->rtp_blocksize);
5109 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
5114 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
5117 /* handle the code ourselves */
5118 if ((res = gst_rtspsrc_send (src, conn, &request, &response, &code) < 0))
5122 case GST_RTSP_STS_OK:
5124 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
5125 gst_rtsp_message_unset (&request);
5126 gst_rtsp_message_unset (&response);
5127 /* cleanup of leftover transport */
5128 gst_rtspsrc_stream_free_udp (stream);
5129 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
5130 * we might be in this case */
5131 if (stream->container && rtpport && rtcpport && !retry) {
5132 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
5137 /* this transport did not go down well, but we may have others to try
5138 * that we did not send yet, try those and only give up then
5139 * but not without checking for lost cause/extension so we can
5140 * post a nicer/more useful error message later */
5141 if (!unsupported_real)
5142 unsupported_real = gst_rtspsrc_stream_is_real_media (stream);
5143 /* select next available protocol, give up on this stream if none */
5145 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5147 if (!protocol_masks[mask] || unsupported_real)
5152 /* cleanup of leftover transport and move to the next stream */
5153 gst_rtspsrc_stream_free_udp (stream);
5154 goto response_error;
5157 /* parse response transport */
5159 gchar *resptrans = NULL;
5160 GstRTSPTransport transport = { 0 };
5162 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
5165 gst_rtspsrc_stream_free_udp (stream);
5169 /* parse transport, go to next stream on parse error */
5170 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
5171 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
5175 /* update allowed transports for other streams. once the transport of
5176 * one stream has been determined, we make sure that all other streams
5177 * are configured in the same way */
5178 switch (transport.lower_transport) {
5179 case GST_RTSP_LOWER_TRANS_TCP:
5180 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
5181 protocols = GST_RTSP_LOWER_TRANS_TCP;
5182 src->interleaved = TRUE;
5183 /* update free channels */
5185 MAX (transport.interleaved.min, src->free_channel);
5187 MAX (transport.interleaved.max, src->free_channel);
5188 src->free_channel++;
5190 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
5191 /* only allow multicast for other streams */
5192 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
5193 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
5195 case GST_RTSP_LOWER_TRANS_UDP:
5196 /* only allow unicast for other streams */
5197 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
5198 protocols = GST_RTSP_LOWER_TRANS_UDP;
5201 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
5202 transport.lower_transport);
5206 if (!stream->container || (!src->interleaved && !retry)) {
5207 /* now configure the stream with the selected transport */
5208 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
5209 GST_DEBUG_OBJECT (src,
5210 "could not configure stream %p transport, skipping stream",
5213 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
5214 /* retain the first allocated UDP port pair */
5215 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
5216 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
5219 /* we need to activate at least one streams when we detect activity */
5220 src->need_activate = TRUE;
5222 /* clean up our transport struct */
5223 gst_rtsp_transport_init (&transport);
5224 /* clean up used RTSP messages */
5225 gst_rtsp_message_unset (&request);
5226 gst_rtsp_message_unset (&response);
5230 /* store the transport protocol that was configured */
5231 src->cur_protocols = protocols;
5233 gst_rtsp_ext_list_stream_select (src->extensions, url);
5235 /* if there is nothing to activate, error out */
5236 if (!src->need_activate)
5237 goto nothing_to_activate;
5244 /* no transport possible, post an error and stop */
5245 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5246 ("Could not connect to server, no protocols left"));
5247 return GST_RTSP_ERROR;
5251 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5252 ("SDP contains no streams"));
5253 return GST_RTSP_ERROR;
5255 create_request_failed:
5257 gchar *str = gst_rtsp_strresult (res);
5259 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5260 ("Could not create request. (%s)", str));
5264 setup_transport_failed:
5266 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5267 ("Could not setup transport."));
5268 res = GST_RTSP_ERROR;
5273 const gchar *str = gst_rtsp_status_as_text (code);
5275 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5276 ("Error (%d): %s", code, GST_STR_NULL (str)));
5277 res = GST_RTSP_ERROR;
5282 gchar *str = gst_rtsp_strresult (res);
5284 if (res != GST_RTSP_EINTR) {
5285 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5286 ("Could not send message. (%s)", str));
5288 GST_WARNING_OBJECT (src, "send interrupted");
5295 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5296 ("Server did not select transport."));
5297 res = GST_RTSP_ERROR;
5300 nothing_to_activate:
5302 /* none of the available error codes is really right .. */
5303 if (unsupported_real) {
5304 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
5305 (_("No supported stream was found. You might need to install a "
5306 "GStreamer RTSP extension plugin for Real media streams.")),
5309 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
5310 (_("No supported stream was found. You might need to allow "
5311 "more transport protocols or may otherwise be missing "
5312 "the right GStreamer RTSP extension plugin.")), (NULL));
5314 return GST_RTSP_ERROR;
5318 gst_rtsp_message_unset (&request);
5319 gst_rtsp_message_unset (&response);
5325 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
5326 GstSegment * segment)
5329 GstRTSPTimeRange *therange;
5332 gst_rtsp_range_free (src->range);
5334 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
5335 GST_DEBUG_OBJECT (src, "parsed range %s", range);
5336 src->range = therange;
5338 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
5340 gst_segment_init (segment, GST_FORMAT_TIME);
5344 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
5345 therange->min.type, therange->min.seconds, therange->max.type,
5346 therange->max.seconds);
5348 if (therange->min.type == GST_RTSP_TIME_NOW)
5350 else if (therange->min.type == GST_RTSP_TIME_END)
5353 seconds = therange->min.seconds * GST_SECOND;
5355 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
5356 GST_TIME_ARGS (seconds));
5358 /* we need to start playback without clipping from the position reported by
5360 segment->start = seconds;
5361 segment->position = seconds;
5363 if (therange->max.type == GST_RTSP_TIME_NOW)
5365 else if (therange->max.type == GST_RTSP_TIME_END)
5368 seconds = therange->max.seconds * GST_SECOND;
5370 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
5371 GST_TIME_ARGS (seconds));
5373 /* live (WMS) server might send overflowed large max as its idea of infinity,
5374 * compensate to prevent problems later on */
5375 if (seconds != -1 && seconds < 0) {
5377 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
5380 /* live (WMS) might send min == max, which is not worth recording */
5381 if (segment->duration == -1 && seconds == segment->start)
5384 /* don't change duration with unknown value, we might have a valid value
5385 * there that we want to keep. */
5387 segment->duration = seconds;
5392 /* must be called with the RTSP state lock */
5393 static GstRTSPResult
5394 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
5400 /* prepare global stream caps properties */
5402 gst_structure_remove_all_fields (src->props);
5404 src->props = gst_structure_new_empty ("RTSPProperties");
5407 gst_sdp_message_dump (sdp);
5409 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
5411 gst_segment_init (&src->segment, GST_FORMAT_TIME);
5413 /* parse range for duration reporting. */
5418 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
5422 /* keep track of the range and configure it in the segment */
5423 if (gst_rtspsrc_parse_range (src, range, &src->segment))
5427 /* try to find a global control attribute. Note that a '*' means that we should
5428 * do aggregate control with the current url (so we don't do anything and
5429 * leave the current connection as is) */
5431 const gchar *control;
5434 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
5435 if (control == NULL)
5438 /* only take fully qualified urls */
5439 if (g_str_has_prefix (control, "rtsp://"))
5443 g_free (src->conninfo.location);
5444 src->conninfo.location = g_strdup (control);
5445 /* make a connection for this, if there was a connection already, nothing
5447 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
5448 GST_ERROR_OBJECT (src, "could not connect");
5451 /* we need to keep the control url separate from the connection url because
5452 * the rules for constructing the media control url need it */
5453 g_free (src->control);
5454 src->control = g_strdup (control);
5457 /* create streams */
5458 n_streams = gst_sdp_message_medias_len (sdp);
5459 for (i = 0; i < n_streams; i++) {
5460 gst_rtspsrc_create_stream (src, sdp, i);
5463 src->state = GST_RTSP_STATE_INIT;
5466 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
5469 /* reset our state */
5470 src->need_range = TRUE;
5473 src->state = GST_RTSP_STATE_READY;
5480 GST_ERROR_OBJECT (src, "setup failed");
5485 static GstRTSPResult
5486 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
5490 GstRTSPMessage request = { 0 };
5491 GstRTSPMessage response = { 0 };
5494 gchar *respcont = NULL;
5497 src->need_redirect = FALSE;
5499 /* can't continue without a valid url */
5500 if (G_UNLIKELY (src->conninfo.url == NULL)) {
5501 res = GST_RTSP_EINVAL;
5504 src->tried_url_auth = FALSE;
5506 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
5507 goto connect_failed;
5509 /* create OPTIONS */
5510 GST_DEBUG_OBJECT (src, "create options...");
5512 gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
5513 src->conninfo.url_str);
5515 goto create_request_failed;
5518 GST_DEBUG_OBJECT (src, "send options...");
5521 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
5524 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
5529 if (!gst_rtspsrc_parse_methods (src, &response))
5532 /* create DESCRIBE */
5533 GST_DEBUG_OBJECT (src, "create describe...");
5535 gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
5536 src->conninfo.url_str);
5538 goto create_request_failed;
5540 /* we only accept SDP for now */
5541 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
5545 GST_DEBUG_OBJECT (src, "send describe...");
5548 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
5551 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
5555 /* we only perform redirect for the describe, currently */
5556 if (src->need_redirect) {
5557 /* close connection, we don't have to send a TEARDOWN yet, ignore the
5559 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5561 gst_rtsp_message_unset (&request);
5562 gst_rtsp_message_unset (&response);
5568 /* it could be that the DESCRIBE method was not implemented */
5569 if (!src->methods & GST_RTSP_DESCRIBE)
5572 /* check if reply is SDP */
5573 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
5575 /* could not be set but since the request returned OK, we assume it
5576 * was SDP, else check it. */
5578 if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
5579 goto wrong_content_type;
5582 /* get message body and parse as SDP */
5583 gst_rtsp_message_get_body (&response, &data, &size);
5584 if (data == NULL || size == 0)
5587 GST_DEBUG_OBJECT (src, "parse SDP...");
5588 gst_sdp_message_new (sdp);
5589 gst_sdp_message_parse_buffer (data, size, *sdp);
5591 /* clean up any messages */
5592 gst_rtsp_message_unset (&request);
5593 gst_rtsp_message_unset (&response);
5600 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
5601 ("No valid RTSP URL was provided"));
5606 gchar *str = gst_rtsp_strresult (res);
5608 if (res != GST_RTSP_EINTR) {
5609 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
5610 ("Failed to connect. (%s)", str));
5612 GST_WARNING_OBJECT (src, "connect interrupted");
5617 create_request_failed:
5619 gchar *str = gst_rtsp_strresult (res);
5621 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5622 ("Could not create request. (%s)", str));
5628 /* Don't post a message - the rtsp_send method will have
5629 * taken care of it because we passed NULL for the response code */
5634 /* error was posted */
5635 res = GST_RTSP_ERROR;
5640 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5641 ("Server does not support SDP, got %s.", respcont));
5642 res = GST_RTSP_ERROR;
5647 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5648 ("Server can not provide an SDP."));
5649 res = GST_RTSP_ERROR;
5654 if (src->conninfo.connection) {
5655 GST_DEBUG_OBJECT (src, "free connection");
5656 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5658 gst_rtsp_message_unset (&request);
5659 gst_rtsp_message_unset (&response);
5664 static GstRTSPResult
5665 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
5670 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
5672 if (src->sdp == NULL) {
5673 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
5677 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
5682 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
5689 GST_WARNING_OBJECT (src, "can't get sdp");
5690 src->open_error = TRUE;
5695 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
5696 src->open_error = TRUE;
5701 static GstRTSPResult
5702 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
5704 GstRTSPMessage request = { 0 };
5705 GstRTSPMessage response = { 0 };
5706 GstRTSPResult res = GST_RTSP_OK;
5710 GST_DEBUG_OBJECT (src, "TEARDOWN...");
5712 if (src->state < GST_RTSP_STATE_READY) {
5713 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
5720 /* construct a control url */
5722 control = src->control;
5724 control = src->conninfo.url_str;
5726 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
5729 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5730 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5732 GstRTSPConnInfo *info;
5734 /* try aggregate control first but do non-aggregate control otherwise */
5736 setup_url = control;
5737 else if ((setup_url = stream->conninfo.location) == NULL)
5740 if (src->conninfo.connection) {
5741 info = &src->conninfo;
5742 } else if (stream->conninfo.connection) {
5743 info = &stream->conninfo;
5747 if (!info->connected)
5752 gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
5754 goto create_request_failed;
5757 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
5760 gst_rtspsrc_send (src, info->connection, &request, &response,
5764 /* FIXME, parse result? */
5765 gst_rtsp_message_unset (&request);
5766 gst_rtsp_message_unset (&response);
5769 /* early exit when we did aggregate control */
5775 /* close connections */
5776 GST_DEBUG_OBJECT (src, "closing connection...");
5777 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5778 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5779 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5780 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
5784 gst_rtspsrc_cleanup (src);
5786 src->state = GST_RTSP_STATE_INVALID;
5789 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
5794 create_request_failed:
5796 gchar *str = gst_rtsp_strresult (res);
5798 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5799 ("Could not create request. (%s)", str));
5805 gchar *str = gst_rtsp_strresult (res);
5807 gst_rtsp_message_unset (&request);
5808 if (res != GST_RTSP_EINTR) {
5809 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5810 ("Could not send message. (%s)", str));
5812 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
5819 GST_DEBUG_OBJECT (src,
5820 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
5825 /* RTP-Info is of the format:
5827 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
5829 * rtptime corresponds to the timestamp for the NPT time given in the header
5830 * seqbase corresponds to the next sequence number we received. This number
5831 * indicates the first seqnum after the seek and should be used to discard
5832 * packets that are from before the seek.
5835 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
5840 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
5842 infos = g_strsplit (rtpinfo, ",", 0);
5843 for (i = 0; infos[i]; i++) {
5845 GstRTSPStream *stream;
5849 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
5851 /* init values, types of seqbase and timebase are bigger than needed so we
5852 * can store -1 as uninitialized values */
5857 /* parse url, find stream for url.
5858 * parse seq and rtptime. The seq number should be configured in the rtp
5859 * depayloader or session manager to detect gaps. Same for the rtptime, it
5860 * should be used to create an initial time newsegment. */
5861 fields = g_strsplit (infos[i], ";", 0);
5862 for (j = 0; fields[j]; j++) {
5863 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
5864 /* remove leading whitespace */
5865 fields[j] = g_strchug (fields[j]);
5866 if (g_str_has_prefix (fields[j], "url=")) {
5867 /* get the url and the stream */
5869 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
5870 } else if (g_str_has_prefix (fields[j], "seq=")) {
5871 seqbase = atoi (fields[j] + 4);
5872 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
5873 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
5876 g_strfreev (fields);
5877 /* now we need to store the values for the caps of the stream */
5878 if (stream != NULL) {
5879 GST_DEBUG_OBJECT (src,
5880 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
5881 stream, seqbase, timebase);
5883 /* we have a stream, configure detected params */
5884 stream->seqbase = seqbase;
5885 stream->timebase = timebase;
5894 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
5899 interval = strtoul (rtcp, NULL, 10);
5900 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
5905 interval *= GST_MSECOND;
5907 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5908 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5910 /* already (optionally) retrieved this when configuring manager */
5911 if (stream->session) {
5912 GObject *rtpsession = stream->session;
5914 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
5916 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
5920 /* now it happens that (Xenon) server sending this may also provide bogus
5921 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
5922 * and just use RTP-Info to sync */
5924 GObjectClass *klass;
5926 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
5927 if (g_object_class_find_property (klass, "rtcp-sync")) {
5928 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
5929 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
5935 gst_rtspsrc_get_float (const gchar * dstr)
5937 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
5939 /* canonicalise floating point string so we can handle float strings
5940 * in the form "24.930" or "24,930" irrespective of the current locale */
5941 g_strlcpy (s, dstr, sizeof (s));
5942 g_strdelimit (s, ",", '.');
5943 return g_ascii_strtod (s, NULL);
5947 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
5949 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
5951 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
5952 g_strlcpy (val_str, "now", sizeof (val_str));
5954 if (segment->position == 0) {
5955 g_strlcpy (val_str, "0", sizeof (val_str));
5957 g_ascii_dtostr (val_str, sizeof (val_str),
5958 ((gdouble) segment->position) / GST_SECOND);
5961 return g_strdup_printf ("npt=%s-", val_str);
5965 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
5967 stream->timebase = -1;
5968 stream->seqbase = -1;
5972 stream->caps = gst_caps_make_writable (stream->caps);
5973 s = gst_caps_get_structure (stream->caps, 0);
5974 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
5978 static GstRTSPResult
5979 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
5981 GstRTSPResult res = GST_RTSP_OK;
5983 if (src->state < GST_RTSP_STATE_READY) {
5984 res = GST_RTSP_ERROR;
5985 if (src->open_error) {
5986 GST_DEBUG_OBJECT (src, "the stream was in error");
5990 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
5992 if ((res = gst_rtspsrc_open (src, async)) < 0) {
5993 GST_DEBUG_OBJECT (src, "failed to open stream");
6002 static GstRTSPResult
6003 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
6005 GstRTSPMessage request = { 0 };
6006 GstRTSPMessage response = { 0 };
6007 GstRTSPResult res = GST_RTSP_OK;
6013 GST_DEBUG_OBJECT (src, "PLAY...");
6015 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6018 if (!(src->methods & GST_RTSP_PLAY))
6021 if (src->state == GST_RTSP_STATE_PLAYING)
6024 if (!src->conninfo.connection || !src->conninfo.connected)
6027 /* send some dummy packets before we activate the receive in the
6029 gst_rtspsrc_send_dummy_packets (src);
6031 /* activate receive elements;
6032 * only in async case, since receive elements may not have been affected
6033 * by overall state change (e.g. not around yet),
6034 * do not mess with state in sync case (e.g. seeking) */
6036 gst_element_set_state (GST_ELEMENT_CAST (src), GST_STATE_PLAYING);
6038 /* construct a control url */
6040 control = src->control;
6042 control = src->conninfo.url_str;
6044 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6045 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6047 GstRTSPConnection *conn;
6049 /* try aggregate control first but do non-aggregate control otherwise */
6051 setup_url = control;
6052 else if ((setup_url = stream->conninfo.location) == NULL)
6055 if (src->conninfo.connection) {
6056 conn = src->conninfo.connection;
6057 } else if (stream->conninfo.connection) {
6058 conn = stream->conninfo.connection;
6064 res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
6066 goto create_request_failed;
6068 if (src->need_range) {
6069 hval = gen_range_header (src, segment);
6071 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
6075 if (segment->rate != 1.0) {
6076 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
6078 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
6080 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
6082 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
6086 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
6088 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
6091 /* seek may have silently failed as it is not supported */
6092 if (!(src->methods & GST_RTSP_PLAY)) {
6093 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
6094 /* obviously it is supported as we made it here */
6095 src->methods |= GST_RTSP_PLAY;
6096 src->seekable = FALSE;
6097 /* but there is nothing to parse in the response,
6098 * so convey we have no idea and not to expect anything particular */
6099 clear_rtp_base (src, stream);
6103 /* need to do for all streams */
6104 for (run = src->streams; run; run = g_list_next (run))
6105 clear_rtp_base (src, (GstRTSPStream *) run->data);
6107 /* NOTE the above also disables npt based eos detection */
6108 /* and below forces position to 0,
6109 * which is visible feedback we lost the plot */
6110 segment->start = segment->position = src->last_pos;
6113 gst_rtsp_message_unset (&request);
6115 /* parse RTP npt field. This is the current position in the stream (Normal
6116 * Play Time) and should be put in the NEWSEGMENT position field. */
6117 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
6119 gst_rtspsrc_parse_range (src, hval, segment);
6121 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
6122 segment->rate = 1.0;
6124 /* parse Speed header. This is the intended playback rate of the stream
6125 * and should be put in the NEWSEGMENT rate field. */
6126 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
6127 0) == GST_RTSP_OK) {
6128 segment->rate = gst_rtspsrc_get_float (hval);
6129 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
6130 &hval, 0) == GST_RTSP_OK) {
6131 segment->rate = gst_rtspsrc_get_float (hval);
6134 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
6135 * for the RTP packets. If this is not present, we assume all starts from 0...
6136 * This is info for the RTP session manager that we pass to it in caps. */
6138 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
6139 &hval, hval_idx++) == GST_RTSP_OK)
6140 gst_rtspsrc_parse_rtpinfo (src, hval);
6142 /* some servers indicate RTCP parameters in PLAY response,
6143 * rather than properly in SDP */
6144 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
6145 &hval, 0) == GST_RTSP_OK)
6146 gst_rtspsrc_handle_rtcp_interval (src, hval);
6148 gst_rtsp_message_unset (&response);
6150 /* early exit when we did aggregate control */
6154 /* set again when needed */
6155 src->need_range = FALSE;
6157 /* configure the caps of the streams after we parsed all headers. */
6158 gst_rtspsrc_configure_caps (src, segment);
6160 src->running = TRUE;
6161 src->base_time = -1;
6162 src->state = GST_RTSP_STATE_PLAYING;
6165 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
6166 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6167 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6168 stream->discont = TRUE;
6173 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
6180 GST_DEBUG_OBJECT (src, "failed to open stream");
6185 GST_DEBUG_OBJECT (src, "PLAY is not supported");
6190 GST_DEBUG_OBJECT (src, "we were already PLAYING");
6193 create_request_failed:
6195 gchar *str = gst_rtsp_strresult (res);
6197 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6198 ("Could not create request. (%s)", str));
6204 gchar *str = gst_rtsp_strresult (res);
6206 gst_rtsp_message_unset (&request);
6207 if (res != GST_RTSP_EINTR) {
6208 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6209 ("Could not send message. (%s)", str));
6211 GST_WARNING_OBJECT (src, "PLAY interrupted");
6218 static GstRTSPResult
6219 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle, gboolean async)
6221 GstRTSPResult res = GST_RTSP_OK;
6222 GstRTSPMessage request = { 0 };
6223 GstRTSPMessage response = { 0 };
6227 GST_DEBUG_OBJECT (src, "PAUSE...");
6229 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6232 if (!(src->methods & GST_RTSP_PAUSE))
6235 if (src->state == GST_RTSP_STATE_READY)
6238 if (!src->conninfo.connection || !src->conninfo.connected)
6241 /* construct a control url */
6243 control = src->control;
6245 control = src->conninfo.url_str;
6247 /* loop over the streams. We might exit the loop early when we could do an
6248 * aggregate control */
6249 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6250 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6251 GstRTSPConnection *conn;
6254 /* try aggregate control first but do non-aggregate control otherwise */
6256 setup_url = control;
6257 else if ((setup_url = stream->conninfo.location) == NULL)
6260 if (src->conninfo.connection) {
6261 conn = src->conninfo.connection;
6262 } else if (stream->conninfo.connection) {
6263 conn = stream->conninfo.connection;
6269 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
6270 ("Sending PAUSE request"));
6273 gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
6275 goto create_request_failed;
6277 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
6280 gst_rtsp_message_unset (&request);
6281 gst_rtsp_message_unset (&response);
6283 /* exit early when we did agregate control */
6289 src->state = GST_RTSP_STATE_READY;
6293 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
6300 GST_DEBUG_OBJECT (src, "failed to open stream");
6305 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
6310 GST_DEBUG_OBJECT (src, "we were already PAUSED");
6313 create_request_failed:
6315 gchar *str = gst_rtsp_strresult (res);
6317 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6318 ("Could not create request. (%s)", str));
6324 gchar *str = gst_rtsp_strresult (res);
6326 gst_rtsp_message_unset (&request);
6327 if (res != GST_RTSP_EINTR) {
6328 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6329 ("Could not send message. (%s)", str));
6331 GST_WARNING_OBJECT (src, "PAUSE interrupted");
6339 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
6341 GstRTSPSrc *rtspsrc;
6343 rtspsrc = GST_RTSPSRC (bin);
6345 switch (GST_MESSAGE_TYPE (message)) {
6346 case GST_MESSAGE_EOS:
6347 gst_message_unref (message);
6349 case GST_MESSAGE_ELEMENT:
6351 const GstStructure *s = gst_message_get_structure (message);
6353 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
6354 gboolean ignore_timeout;
6356 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
6358 GST_OBJECT_LOCK (rtspsrc);
6359 ignore_timeout = rtspsrc->ignore_timeout;
6360 rtspsrc->ignore_timeout = TRUE;
6361 GST_OBJECT_UNLOCK (rtspsrc);
6363 /* we only act on the first udp timeout message, others are irrelevant
6364 * and can be ignored. */
6365 if (!ignore_timeout)
6366 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT);
6368 gst_message_unref (message);
6371 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6374 case GST_MESSAGE_ERROR:
6377 GstRTSPStream *stream;
6380 udpsrc = GST_MESSAGE_SRC (message);
6382 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
6383 GST_ELEMENT_NAME (udpsrc));
6385 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
6389 /* we ignore the RTCP udpsrc */
6390 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
6393 /* if we get error messages from the udp sources, that's not a problem as
6394 * long as not all of them error out. We also don't really know what the
6395 * problem is, the message does not give enough detail... */
6396 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
6397 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
6398 if (ret != GST_FLOW_OK)
6402 gst_message_unref (message);
6406 /* fatal but not our message, forward */
6407 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6412 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6418 /* the thread where everything happens */
6420 gst_rtspsrc_thread (GstRTSPSrc * src)
6424 gboolean running = FALSE;
6426 GST_OBJECT_LOCK (src);
6427 cmd = src->loop_cmd;
6428 src->loop_cmd = CMD_WAIT;
6429 GST_DEBUG_OBJECT (src, "got command %d", cmd);
6431 /* we got the message command, so ensure communication is possible again */
6432 gst_rtspsrc_connection_flush (src, FALSE);
6434 /* we allow these to be interrupted */
6435 if (cmd == CMD_LOOP || cmd == CMD_CLOSE || cmd == CMD_PAUSE)
6436 src->waiting = TRUE;
6437 GST_OBJECT_UNLOCK (src);
6441 ret = gst_rtspsrc_open (src, TRUE);
6444 ret = gst_rtspsrc_play (src, &src->segment, TRUE);
6445 if (ret == GST_RTSP_OK)
6449 ret = gst_rtspsrc_pause (src, TRUE, TRUE);
6450 if (ret == GST_RTSP_OK)
6454 ret = gst_rtspsrc_close (src, TRUE, FALSE);
6457 running = gst_rtspsrc_loop (src);
6460 ret = gst_rtspsrc_reconnect (src, FALSE);
6461 if (ret == GST_RTSP_OK)
6468 GST_OBJECT_LOCK (src);
6469 /* and go back to sleep */
6470 if (src->loop_cmd == CMD_WAIT) {
6472 src->loop_cmd = CMD_LOOP;
6474 gst_task_pause (src->task);
6477 src->waiting = FALSE;
6478 GST_OBJECT_UNLOCK (src);
6482 gst_rtspsrc_start (GstRTSPSrc * src)
6484 GST_DEBUG_OBJECT (src, "starting");
6486 GST_OBJECT_LOCK (src);
6488 src->loop_cmd = CMD_WAIT;
6490 if (src->task == NULL) {
6491 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src);
6492 if (src->task == NULL)
6495 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
6497 GST_OBJECT_UNLOCK (src);
6504 GST_ERROR_OBJECT (src, "failed to create task");
6510 gst_rtspsrc_stop (GstRTSPSrc * src)
6514 GST_DEBUG_OBJECT (src, "stopping");
6516 /* also cancels pending task */
6517 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT);
6519 GST_OBJECT_LOCK (src);
6520 if ((task = src->task)) {
6522 GST_OBJECT_UNLOCK (src);
6524 gst_task_stop (task);
6526 /* make sure it is not running */
6527 GST_RTSP_STREAM_LOCK (src);
6528 GST_RTSP_STREAM_UNLOCK (src);
6530 /* now wait for the task to finish */
6531 gst_task_join (task);
6533 /* and free the task */
6534 gst_object_unref (GST_OBJECT (task));
6536 GST_OBJECT_LOCK (src);
6538 GST_OBJECT_UNLOCK (src);
6540 /* ensure synchronously all is closed and clean */
6541 gst_rtspsrc_close (src, FALSE, TRUE);
6546 static GstStateChangeReturn
6547 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
6549 GstRTSPSrc *rtspsrc;
6550 GstStateChangeReturn ret;
6552 rtspsrc = GST_RTSPSRC (element);
6554 switch (transition) {
6555 case GST_STATE_CHANGE_NULL_TO_READY:
6556 if (!gst_rtspsrc_start (rtspsrc))
6559 case GST_STATE_CHANGE_READY_TO_PAUSED:
6560 /* init some state */
6561 rtspsrc->cur_protocols = rtspsrc->protocols;
6562 /* first attempt, don't ignore timeouts */
6563 rtspsrc->ignore_timeout = FALSE;
6564 rtspsrc->open_error = FALSE;
6565 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN);
6567 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
6568 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
6569 /* unblock the tcp tasks and make the loop waiting */
6570 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT);
6572 case GST_STATE_CHANGE_PAUSED_TO_READY:
6578 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
6579 if (ret == GST_STATE_CHANGE_FAILURE)
6582 switch (transition) {
6583 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
6584 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY);
6586 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
6587 /* send pause request and keep the idle task around */
6588 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE);
6589 ret = GST_STATE_CHANGE_NO_PREROLL;
6591 case GST_STATE_CHANGE_READY_TO_PAUSED:
6592 ret = GST_STATE_CHANGE_NO_PREROLL;
6594 case GST_STATE_CHANGE_PAUSED_TO_READY:
6595 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE);
6597 case GST_STATE_CHANGE_READY_TO_NULL:
6598 gst_rtspsrc_stop (rtspsrc);
6609 GST_DEBUG_OBJECT (rtspsrc, "start failed");
6610 return GST_STATE_CHANGE_FAILURE;
6615 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
6618 GstRTSPSrc *rtspsrc;
6620 rtspsrc = GST_RTSPSRC (element);
6622 if (GST_EVENT_IS_DOWNSTREAM (event)) {
6623 res = gst_rtspsrc_push_event (rtspsrc, event, TRUE);
6625 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
6632 /*** GSTURIHANDLER INTERFACE *************************************************/
6635 gst_rtspsrc_uri_get_type (GType type)
6640 static const gchar *const *
6641 gst_rtspsrc_uri_get_protocols (GType type)
6643 static const gchar *protocols[] =
6644 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp", NULL };
6650 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
6652 GstRTSPSrc *src = GST_RTSPSRC (handler);
6654 /* FIXME: make thread-safe */
6655 return g_strdup (src->conninfo.location);
6659 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
6664 GstRTSPUrl *newurl = NULL;
6665 GstSDPMessage *sdp = NULL;
6667 src = GST_RTSPSRC (handler);
6669 /* same URI, we're fine */
6670 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
6673 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
6674 if ((res = gst_sdp_message_new (&sdp) < 0))
6677 GST_DEBUG_OBJECT (src, "parsing SDP message");
6678 if ((res = gst_sdp_message_parse_uri (uri, sdp) < 0))
6682 GST_DEBUG_OBJECT (src, "parsing URI");
6683 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
6687 /* if worked, free previous and store new url object along with the original
6689 GST_DEBUG_OBJECT (src, "configuring URI");
6690 g_free (src->conninfo.location);
6691 src->conninfo.location = g_strdup (uri);
6692 gst_rtsp_url_free (src->conninfo.url);
6693 src->conninfo.url = newurl;
6694 g_free (src->conninfo.url_str);
6696 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
6698 src->conninfo.url_str = NULL;
6701 gst_sdp_message_free (src->sdp);
6703 src->from_sdp = sdp != NULL;
6705 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
6706 GST_DEBUG_OBJECT (src, "request uri is: %s",
6707 GST_STR_NULL (src->conninfo.url_str));
6714 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
6719 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", res);
6720 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6721 "Could not create SDP");
6726 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", res,
6727 GST_STR_NULL (uri));
6728 gst_sdp_message_free (sdp);
6729 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6735 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
6736 GST_STR_NULL (uri), res);
6737 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6738 "Invalid RTSP URI");
6744 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
6746 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
6748 iface->get_type = gst_rtspsrc_uri_get_type;
6749 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
6750 iface->get_uri = gst_rtspsrc_uri_get_uri;
6751 iface->set_uri = gst_rtspsrc_uri_set_uri;