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., 51 Franklin St, Fifth Floor,
18 * Boston, MA 02110-1301, 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-1.0 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 */
92 #include <gst/net/gstnet.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);
123 SIGNAL_HANDLE_REQUEST,
125 SIGNAL_SELECT_STREAM,
130 enum _GstRtspSrcRtcpSyncMode
137 enum _GstRtspSrcBufferMode
146 #define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
148 gst_rtsp_src_buffer_mode_get_type (void)
150 static GType buffer_mode_type = 0;
151 static const GEnumValue buffer_modes[] = {
152 {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
153 {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
154 {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
155 {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
156 {BUFFER_MODE_SYNCED, "Synchronized sender and receiver clocks", "synced"},
160 if (!buffer_mode_type) {
162 g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
164 return buffer_mode_type;
167 #define DEFAULT_LOCATION NULL
168 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
169 #define DEFAULT_DEBUG FALSE
170 #define DEFAULT_RETRY 20
171 #define DEFAULT_TIMEOUT 5000000
172 #define DEFAULT_UDP_BUFFER_SIZE 0x80000
173 #define DEFAULT_TCP_TIMEOUT 20000000
174 #define DEFAULT_LATENCY_MS 2000
175 #define DEFAULT_DROP_ON_LATENCY FALSE
176 #define DEFAULT_CONNECTION_SPEED 0
177 #define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
178 #define DEFAULT_DO_RTCP TRUE
179 #define DEFAULT_DO_RTSP_KEEP_ALIVE TRUE
180 #define DEFAULT_PROXY NULL
181 #define DEFAULT_RTP_BLOCKSIZE 0
182 #define DEFAULT_USER_ID NULL
183 #define DEFAULT_USER_PW NULL
184 #define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO
185 #define DEFAULT_PORT_RANGE NULL
186 #define DEFAULT_SHORT_HEADER FALSE
187 #define DEFAULT_PROBATION 2
188 #define DEFAULT_UDP_RECONNECT TRUE
189 #define DEFAULT_MULTICAST_IFACE NULL
190 #define DEFAULT_NTP_SYNC FALSE
191 #define DEFAULT_USE_PIPELINE_CLOCK FALSE
192 #define DEFAULT_TLS_VALIDATION_FLAGS G_TLS_CERTIFICATE_VALIDATE_ALL
193 #define DEFAULT_TLS_DATABASE NULL
205 PROP_DROP_ON_LATENCY,
206 PROP_CONNECTION_SPEED,
209 PROP_DO_RTSP_KEEP_ALIVE,
218 PROP_UDP_BUFFER_SIZE,
222 PROP_MULTICAST_IFACE,
224 PROP_USE_PIPELINE_CLOCK,
226 PROP_TLS_VALIDATION_FLAGS,
231 #define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
233 gst_rtsp_nat_method_get_type (void)
235 static GType rtsp_nat_method_type = 0;
236 static const GEnumValue rtsp_nat_method[] = {
237 {GST_RTSP_NAT_NONE, "None", "none"},
238 {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
242 if (!rtsp_nat_method_type) {
243 rtsp_nat_method_type =
244 g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
246 return rtsp_nat_method_type;
249 static void gst_rtspsrc_finalize (GObject * object);
251 static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
252 const GValue * value, GParamSpec * pspec);
253 static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
254 GValue * value, GParamSpec * pspec);
256 static GstClock *gst_rtspsrc_provide_clock (GstElement * element);
258 static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
259 gpointer iface_data);
261 static void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
264 static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
265 static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
267 static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
269 static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
270 GstStateChange transition);
271 static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
272 static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
274 static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
275 GstRTSPMessage * response);
277 static gboolean gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd,
279 static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
280 GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
282 static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
283 static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
285 static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
286 static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
287 gboolean only_close);
289 static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
290 const gchar * uri, GError ** error);
291 static gchar *gst_rtspsrc_uri_get_uri (GstURIHandler * handler);
293 static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
294 static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
295 static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
296 GstRTSPStream * stream, GstEvent * event);
297 static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
298 static void gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush);
306 /* commands we send to out loop to notify it of events */
307 #define CMD_OPEN (1 << 0)
308 #define CMD_PLAY (1 << 1)
309 #define CMD_PAUSE (1 << 2)
310 #define CMD_CLOSE (1 << 3)
311 #define CMD_WAIT (1 << 4)
312 #define CMD_RECONNECT (1 << 5)
313 #define CMD_LOOP (1 << 6)
315 /* mask for all commands */
316 #define CMD_ALL ((CMD_LOOP << 1) - 1)
318 #define GST_ELEMENT_PROGRESS(el, type, code, text) \
320 gchar *__txt = _gst_element_error_printf text; \
321 gst_element_post_message (GST_ELEMENT_CAST (el), \
322 gst_message_new_progress (GST_OBJECT_CAST (el), \
323 GST_PROGRESS_TYPE_ ##type, code, __txt)); \
327 static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 };
329 #define gst_rtspsrc_parent_class parent_class
330 G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
331 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
334 default_select_stream (GstRTSPSrc * src, guint id, GstCaps * caps)
336 GST_DEBUG_OBJECT (src, "default handler");
341 select_stream_accum (GSignalInvocationHint * ihint,
342 GValue * return_accu, const GValue * handler_return, gpointer data)
346 myboolean = g_value_get_boolean (handler_return);
347 GST_DEBUG ("accum %d", myboolean);
348 g_value_set_boolean (return_accu, myboolean);
350 /* stop emission if FALSE */
355 gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
357 GObjectClass *gobject_class;
358 GstElementClass *gstelement_class;
359 GstBinClass *gstbin_class;
361 gobject_class = (GObjectClass *) klass;
362 gstelement_class = (GstElementClass *) klass;
363 gstbin_class = (GstBinClass *) klass;
365 GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
367 gobject_class->set_property = gst_rtspsrc_set_property;
368 gobject_class->get_property = gst_rtspsrc_get_property;
370 gobject_class->finalize = gst_rtspsrc_finalize;
372 g_object_class_install_property (gobject_class, PROP_LOCATION,
373 g_param_spec_string ("location", "RTSP Location",
374 "Location of the RTSP url to read",
375 DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
377 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
378 g_param_spec_flags ("protocols", "Protocols",
379 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
380 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
382 g_object_class_install_property (gobject_class, PROP_DEBUG,
383 g_param_spec_boolean ("debug", "Debug",
384 "Dump request and response messages to stdout",
385 DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
387 g_object_class_install_property (gobject_class, PROP_RETRY,
388 g_param_spec_uint ("retry", "Retry",
389 "Max number of retries when allocating RTP ports.",
390 0, G_MAXUINT16, DEFAULT_RETRY,
391 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
393 g_object_class_install_property (gobject_class, PROP_TIMEOUT,
394 g_param_spec_uint64 ("timeout", "Timeout",
395 "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
396 0, G_MAXUINT64, DEFAULT_TIMEOUT,
397 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
399 g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
400 g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
401 "Fail after timeout microseconds on TCP connections (0 = disabled)",
402 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
403 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
405 g_object_class_install_property (gobject_class, PROP_LATENCY,
406 g_param_spec_uint ("latency", "Buffer latency in ms",
407 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
408 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
410 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
411 g_param_spec_boolean ("drop-on-latency",
412 "Drop buffers when maximum latency is reached",
413 "Tells the jitterbuffer to never exceed the given latency in size",
414 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
416 g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
417 g_param_spec_uint64 ("connection-speed", "Connection Speed",
418 "Network connection speed in kbps (0 = unknown)",
419 0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
420 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
422 g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
423 g_param_spec_enum ("nat-method", "NAT Method",
424 "Method to use for traversing firewalls and NAT",
425 GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
426 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
429 * GstRTSPSrc:do-rtcp:
431 * Enable RTCP support. Some old server don't like RTCP and then this property
432 * needs to be set to FALSE.
434 g_object_class_install_property (gobject_class, PROP_DO_RTCP,
435 g_param_spec_boolean ("do-rtcp", "Do RTCP",
436 "Send RTCP packets, disable for old incompatible server.",
437 DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
440 * GstRTSPSrc:do-rtsp-keep-alive:
442 * Enable RTSP keep alive support. Some old server don't like RTSP
443 * keep alive and then this property needs to be set to FALSE.
445 g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
446 g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
447 "Send RTSP keep alive packets, disable for old incompatible server.",
448 DEFAULT_DO_RTSP_KEEP_ALIVE,
449 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
454 * Set the proxy parameters. This has to be a string of the format
455 * [http://][user:passwd@]host[:port].
457 g_object_class_install_property (gobject_class, PROP_PROXY,
458 g_param_spec_string ("proxy", "Proxy",
459 "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
460 DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
462 * GstRTSPSrc:proxy-id:
464 * Sets the proxy URI user id for authentication. If the URI set via the
465 * "proxy" property contains a user-id already, that will take precedence.
469 g_object_class_install_property (gobject_class, PROP_PROXY_ID,
470 g_param_spec_string ("proxy-id", "proxy-id",
471 "HTTP proxy URI user id for authentication", "",
472 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
474 * GstRTSPSrc:proxy-pw:
476 * Sets the proxy URI password for authentication. If the URI set via the
477 * "proxy" property contains a password already, that will take precedence.
481 g_object_class_install_property (gobject_class, PROP_PROXY_PW,
482 g_param_spec_string ("proxy-pw", "proxy-pw",
483 "HTTP proxy URI user password for authentication", "",
484 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
487 * GstRTSPSrc:rtp-blocksize:
489 * RTP package size to suggest to server.
491 g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
492 g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
493 "RTP package size to suggest to server (0 = disabled)",
494 0, 65536, DEFAULT_RTP_BLOCKSIZE,
495 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
497 g_object_class_install_property (gobject_class,
499 g_param_spec_string ("user-id", "user-id",
500 "RTSP location URI user id for authentication", DEFAULT_USER_ID,
501 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
502 g_object_class_install_property (gobject_class, PROP_USER_PW,
503 g_param_spec_string ("user-pw", "user-pw",
504 "RTSP location URI user password for authentication", DEFAULT_USER_PW,
505 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
508 * GstRTSPSrc:buffer-mode:
510 * Control the buffering and timestamping mode used by the jitterbuffer.
512 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
513 g_param_spec_enum ("buffer-mode", "Buffer Mode",
514 "Control the buffering algorithm in use",
515 GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
516 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
519 * GstRTSPSrc:port-range:
521 * Configure the client port numbers that can be used to recieve RTP and
524 g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
525 g_param_spec_string ("port-range", "Port range",
526 "Client port range that can be used to receive RTP and RTCP data, "
527 "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
528 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
531 * GstRTSPSrc:udp-buffer-size:
533 * Size of the kernel UDP receive buffer in bytes.
535 g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
536 g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
537 "Size of the kernel UDP receive buffer in bytes, 0=default",
538 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
539 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
542 * GstRTSPSrc:short-header:
544 * Only send the basic RTSP headers for broken encoders.
546 g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
547 g_param_spec_boolean ("short-header", "Short Header",
548 "Only send the basic RTSP headers for broken encoders",
549 DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
551 g_object_class_install_property (gobject_class, PROP_PROBATION,
552 g_param_spec_uint ("probation", "Number of probations",
553 "Consecutive packet sequence numbers to accept the source",
554 0, G_MAXUINT, DEFAULT_PROBATION,
555 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
557 g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
558 g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
559 "Reconnect to the server if RTSP connection is closed when doing UDP",
560 DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
562 g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
563 g_param_spec_string ("multicast-iface", "Multicast Interface",
564 "The network interface on which to join the multicast group",
565 DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
567 g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
568 g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
569 "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
570 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
572 g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
573 g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
574 "Use the pipeline running-time to set the NTP time in the RTCP SR messages",
575 DEFAULT_USE_PIPELINE_CLOCK,
576 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
578 g_object_class_install_property (gobject_class, PROP_SDES,
579 g_param_spec_boxed ("sdes", "SDES",
580 "The SDES items of this session",
581 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
584 * GstRTSPSrc::tls-validation-flags:
586 * TLS certificate validation flags used to validate server
591 g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
592 g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
593 "TLS certificate validation flags used to validate the server certificate",
594 G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS,
595 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
598 * GstRTSPSrc::tls-database:
600 * TLS database with anchor certificate authorities used to validate
601 * the server certificate.
605 g_object_class_install_property (gobject_class, PROP_TLS_DATABASE,
606 g_param_spec_object ("tls-database", "TLS database",
607 "TLS database with anchor certificate authorities used to validate the server certificate",
608 G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
611 * GstRTSPSrc::handle-request:
612 * @rtspsrc: a #GstRTSPSrc
613 * @request: a #GstRTSPMessage
614 * @response: a #GstRTSPMessage
616 * Handle a server request in @request and prepare @response.
618 * This signal is called from the streaming thread, you should therefore not
619 * do any state changes on @rtspsrc because this might deadlock. If you want
620 * to modify the state as a result of this signal, post a
621 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
626 gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST] =
627 g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
628 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2,
629 G_TYPE_POINTER, G_TYPE_POINTER);
632 * GstRTSPSrc::on-sdp:
633 * @rtspsrc: a #GstRTSPSrc
634 * @sdp: a #GstSDPMessage
636 * Emited when the client has retrieved the SDP and before it configures the
637 * streams in the SDP. @sdp can be inspected and modified.
639 * This signal is called from the streaming thread, you should therefore not
640 * do any state changes on @rtspsrc because this might deadlock. If you want
641 * to modify the state as a result of this signal, post a
642 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
647 gst_rtspsrc_signals[SIGNAL_ON_SDP] =
648 g_signal_new ("on-sdp", G_TYPE_FROM_CLASS (klass), 0,
649 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
650 GST_TYPE_SDP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
653 * GstRTSPSrc::select-stream:
654 * @rtspsrc: a #GstRTSPSrc
655 * @num: the stream number
656 * @caps: the stream caps
658 * Emited before the client decides to configure the stream @num with
661 * Returns: %TRUE when the stream should be selected, %FALSE when the stream
666 gst_rtspsrc_signals[SIGNAL_SELECT_STREAM] =
667 g_signal_new_class_handler ("select-stream", G_TYPE_FROM_CLASS (klass),
668 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP,
669 (GCallback) default_select_stream, select_stream_accum, NULL,
670 g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2, G_TYPE_UINT,
673 * GstRTSPSrc::new-manager:
674 * @rtspsrc: a #GstRTSPSrc
675 * @manager: a #GstElement
677 * Emited after a new manager (like rtpbin) was created and the default
678 * properties were configured.
682 gst_rtspsrc_signals[SIGNAL_NEW_MANAGER] =
683 g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass),
684 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
685 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
687 gstelement_class->send_event = gst_rtspsrc_send_event;
688 gstelement_class->provide_clock = gst_rtspsrc_provide_clock;
689 gstelement_class->change_state = gst_rtspsrc_change_state;
691 gst_element_class_add_pad_template (gstelement_class,
692 gst_static_pad_template_get (&rtptemplate));
694 gst_element_class_set_static_metadata (gstelement_class,
695 "RTSP packet receiver", "Source/Network",
696 "Receive data over the network via RTSP (RFC 2326)",
697 "Wim Taymans <wim@fluendo.com>, "
698 "Thijs Vermeir <thijs.vermeir@barco.com>, "
699 "Lutz Mueller <lutz@topfrose.de>");
701 gstbin_class->handle_message = gst_rtspsrc_handle_message;
703 gst_rtsp_ext_list_init ();
707 gst_rtspsrc_init (GstRTSPSrc * src)
709 src->conninfo.location = g_strdup (DEFAULT_LOCATION);
710 src->protocols = DEFAULT_PROTOCOLS;
711 src->debug = DEFAULT_DEBUG;
712 src->retry = DEFAULT_RETRY;
713 src->udp_timeout = DEFAULT_TIMEOUT;
714 gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
715 src->latency = DEFAULT_LATENCY_MS;
716 src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
717 src->connection_speed = DEFAULT_CONNECTION_SPEED;
718 src->nat_method = DEFAULT_NAT_METHOD;
719 src->do_rtcp = DEFAULT_DO_RTCP;
720 src->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
721 gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
722 src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
723 src->user_id = g_strdup (DEFAULT_USER_ID);
724 src->user_pw = g_strdup (DEFAULT_USER_PW);
725 src->buffer_mode = DEFAULT_BUFFER_MODE;
726 src->client_port_range.min = 0;
727 src->client_port_range.max = 0;
728 src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
729 src->short_header = DEFAULT_SHORT_HEADER;
730 src->probation = DEFAULT_PROBATION;
731 src->udp_reconnect = DEFAULT_UDP_RECONNECT;
732 src->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
733 src->ntp_sync = DEFAULT_NTP_SYNC;
734 src->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
736 src->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
737 src->tls_database = DEFAULT_TLS_DATABASE;
739 /* get a list of all extensions */
740 src->extensions = gst_rtsp_ext_list_get ();
742 /* connect to send signal */
743 gst_rtsp_ext_list_connect (src->extensions, "send",
744 (GCallback) gst_rtspsrc_send_cb, src);
746 /* protects the streaming thread in interleaved mode or the polling
747 * thread in UDP mode. */
748 g_rec_mutex_init (&src->stream_rec_lock);
750 /* protects our state changes from multiple invocations */
751 g_rec_mutex_init (&src->state_rec_lock);
753 src->state = GST_RTSP_STATE_INVALID;
755 GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
759 gst_rtspsrc_finalize (GObject * object)
763 rtspsrc = GST_RTSPSRC (object);
765 gst_rtsp_ext_list_free (rtspsrc->extensions);
766 g_free (rtspsrc->conninfo.location);
767 gst_rtsp_url_free (rtspsrc->conninfo.url);
768 g_free (rtspsrc->conninfo.url_str);
769 g_free (rtspsrc->user_id);
770 g_free (rtspsrc->user_pw);
771 g_free (rtspsrc->multi_iface);
774 gst_sdp_message_free (rtspsrc->sdp);
777 if (rtspsrc->provided_clock)
778 gst_object_unref (rtspsrc->provided_clock);
781 gst_structure_free (rtspsrc->sdes);
783 if (rtspsrc->tls_database)
784 g_object_unref (rtspsrc->tls_database);
787 g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
788 g_rec_mutex_clear (&rtspsrc->state_rec_lock);
790 G_OBJECT_CLASS (parent_class)->finalize (object);
794 gst_rtspsrc_provide_clock (GstElement * element)
796 GstRTSPSrc *src = GST_RTSPSRC (element);
799 if ((clock = src->provided_clock) != NULL)
800 gst_object_ref (clock);
805 /* a proxy string of the format [user:passwd@]host[:port] */
807 gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
811 g_free (rtsp->proxy_user);
812 rtsp->proxy_user = NULL;
813 g_free (rtsp->proxy_passwd);
814 rtsp->proxy_passwd = NULL;
815 g_free (rtsp->proxy_host);
816 rtsp->proxy_host = NULL;
817 rtsp->proxy_port = 0;
824 /* we allow http:// in front but ignore it */
825 if (g_str_has_prefix (p, "http://"))
828 at = strchr (p, '@');
830 /* look for user:passwd */
831 col = strchr (proxy, ':');
832 if (col == NULL || col > at)
835 rtsp->proxy_user = g_strndup (p, col - p);
837 rtsp->proxy_passwd = g_strndup (col, at - col);
842 if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
843 rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
844 if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
845 rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
846 if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
847 GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
848 GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
851 col = strchr (p, ':');
854 /* everything before the colon is the hostname */
855 rtsp->proxy_host = g_strndup (p, col - p);
857 rtsp->proxy_port = strtoul (p, (char **) &p, 10);
859 rtsp->proxy_host = g_strdup (p);
860 rtsp->proxy_port = 8080;
866 gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
868 rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
869 rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
872 rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
874 rtspsrc->ptcp_timeout = NULL;
878 gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
883 rtspsrc = GST_RTSPSRC (object);
887 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
888 g_value_get_string (value), NULL);
891 rtspsrc->protocols = g_value_get_flags (value);
894 rtspsrc->debug = g_value_get_boolean (value);
897 rtspsrc->retry = g_value_get_uint (value);
900 rtspsrc->udp_timeout = g_value_get_uint64 (value);
902 case PROP_TCP_TIMEOUT:
903 gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
906 rtspsrc->latency = g_value_get_uint (value);
908 case PROP_DROP_ON_LATENCY:
909 rtspsrc->drop_on_latency = g_value_get_boolean (value);
911 case PROP_CONNECTION_SPEED:
912 rtspsrc->connection_speed = g_value_get_uint64 (value);
914 case PROP_NAT_METHOD:
915 rtspsrc->nat_method = g_value_get_enum (value);
918 rtspsrc->do_rtcp = g_value_get_boolean (value);
920 case PROP_DO_RTSP_KEEP_ALIVE:
921 rtspsrc->do_rtsp_keep_alive = g_value_get_boolean (value);
924 gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
927 if (rtspsrc->prop_proxy_id)
928 g_free (rtspsrc->prop_proxy_id);
929 rtspsrc->prop_proxy_id = g_value_dup_string (value);
932 if (rtspsrc->prop_proxy_pw)
933 g_free (rtspsrc->prop_proxy_pw);
934 rtspsrc->prop_proxy_pw = g_value_dup_string (value);
936 case PROP_RTP_BLOCKSIZE:
937 rtspsrc->rtp_blocksize = g_value_get_uint (value);
940 if (rtspsrc->user_id)
941 g_free (rtspsrc->user_id);
942 rtspsrc->user_id = g_value_dup_string (value);
945 if (rtspsrc->user_pw)
946 g_free (rtspsrc->user_pw);
947 rtspsrc->user_pw = g_value_dup_string (value);
949 case PROP_BUFFER_MODE:
950 rtspsrc->buffer_mode = g_value_get_enum (value);
952 case PROP_PORT_RANGE:
956 str = g_value_get_string (value);
958 sscanf (str, "%u-%u",
959 &rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
961 rtspsrc->client_port_range.min = 0;
962 rtspsrc->client_port_range.max = 0;
966 case PROP_UDP_BUFFER_SIZE:
967 rtspsrc->udp_buffer_size = g_value_get_int (value);
969 case PROP_SHORT_HEADER:
970 rtspsrc->short_header = g_value_get_boolean (value);
973 rtspsrc->probation = g_value_get_uint (value);
975 case PROP_UDP_RECONNECT:
976 rtspsrc->udp_reconnect = g_value_get_boolean (value);
978 case PROP_MULTICAST_IFACE:
979 g_free (rtspsrc->multi_iface);
981 if (g_value_get_string (value) == NULL)
982 rtspsrc->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
984 rtspsrc->multi_iface = g_value_dup_string (value);
987 rtspsrc->ntp_sync = g_value_get_boolean (value);
989 case PROP_USE_PIPELINE_CLOCK:
990 rtspsrc->use_pipeline_clock = g_value_get_boolean (value);
993 rtspsrc->sdes = g_value_dup_boxed (value);
995 case PROP_TLS_VALIDATION_FLAGS:
996 rtspsrc->tls_validation_flags = g_value_get_flags (value);
998 case PROP_TLS_DATABASE:
999 g_clear_object (&rtspsrc->tls_database);
1000 rtspsrc->tls_database = g_value_dup_object (value);
1003 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1009 gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
1012 GstRTSPSrc *rtspsrc;
1014 rtspsrc = GST_RTSPSRC (object);
1018 g_value_set_string (value, rtspsrc->conninfo.location);
1020 case PROP_PROTOCOLS:
1021 g_value_set_flags (value, rtspsrc->protocols);
1024 g_value_set_boolean (value, rtspsrc->debug);
1027 g_value_set_uint (value, rtspsrc->retry);
1030 g_value_set_uint64 (value, rtspsrc->udp_timeout);
1032 case PROP_TCP_TIMEOUT:
1036 timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
1037 rtspsrc->tcp_timeout.tv_usec;
1038 g_value_set_uint64 (value, timeout);
1042 g_value_set_uint (value, rtspsrc->latency);
1044 case PROP_DROP_ON_LATENCY:
1045 g_value_set_boolean (value, rtspsrc->drop_on_latency);
1047 case PROP_CONNECTION_SPEED:
1048 g_value_set_uint64 (value, rtspsrc->connection_speed);
1050 case PROP_NAT_METHOD:
1051 g_value_set_enum (value, rtspsrc->nat_method);
1054 g_value_set_boolean (value, rtspsrc->do_rtcp);
1056 case PROP_DO_RTSP_KEEP_ALIVE:
1057 g_value_set_boolean (value, rtspsrc->do_rtsp_keep_alive);
1063 if (rtspsrc->proxy_host) {
1065 g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
1069 g_value_take_string (value, str);
1073 g_value_set_string (value, rtspsrc->prop_proxy_id);
1076 g_value_set_string (value, rtspsrc->prop_proxy_pw);
1078 case PROP_RTP_BLOCKSIZE:
1079 g_value_set_uint (value, rtspsrc->rtp_blocksize);
1082 g_value_set_string (value, rtspsrc->user_id);
1085 g_value_set_string (value, rtspsrc->user_pw);
1087 case PROP_BUFFER_MODE:
1088 g_value_set_enum (value, rtspsrc->buffer_mode);
1090 case PROP_PORT_RANGE:
1094 if (rtspsrc->client_port_range.min != 0) {
1095 str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
1096 rtspsrc->client_port_range.max);
1100 g_value_take_string (value, str);
1103 case PROP_UDP_BUFFER_SIZE:
1104 g_value_set_int (value, rtspsrc->udp_buffer_size);
1106 case PROP_SHORT_HEADER:
1107 g_value_set_boolean (value, rtspsrc->short_header);
1109 case PROP_PROBATION:
1110 g_value_set_uint (value, rtspsrc->probation);
1112 case PROP_UDP_RECONNECT:
1113 g_value_set_boolean (value, rtspsrc->udp_reconnect);
1115 case PROP_MULTICAST_IFACE:
1116 g_value_set_string (value, rtspsrc->multi_iface);
1119 g_value_set_boolean (value, rtspsrc->ntp_sync);
1121 case PROP_USE_PIPELINE_CLOCK:
1122 g_value_set_boolean (value, rtspsrc->use_pipeline_clock);
1125 g_value_set_boxed (value, rtspsrc->sdes);
1127 case PROP_TLS_VALIDATION_FLAGS:
1128 g_value_set_flags (value, rtspsrc->tls_validation_flags);
1130 case PROP_TLS_DATABASE:
1131 g_value_set_object (value, rtspsrc->tls_database);
1134 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1140 find_stream_by_id (GstRTSPStream * stream, gint * id)
1142 if (stream->id == *id)
1149 find_stream_by_channel (GstRTSPStream * stream, gint * channel)
1151 if (stream->channel[0] == *channel || stream->channel[1] == *channel)
1158 find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
1160 GstElement *src = (GstElement *) a;
1162 if (stream->udpsrc[0] == src)
1164 if (stream->udpsrc[1] == src)
1171 find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
1173 if (stream->conninfo.location) {
1174 /* check qualified setup_url */
1175 if (!strcmp (stream->conninfo.location, (gchar *) a))
1178 if (stream->control_url) {
1179 /* check original control_url */
1180 if (!strcmp (stream->control_url, (gchar *) a))
1183 /* check if qualified setup_url ends with string */
1184 if (g_str_has_suffix (stream->control_url, (gchar *) a))
1191 static GstRTSPStream *
1192 find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
1196 /* find and get stream */
1197 if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
1198 return (GstRTSPStream *) lstream->data;
1203 static const GstSDPBandwidth *
1204 gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1205 const GstSDPMedia * media, const gchar * type)
1209 /* first look in the media specific section */
1210 len = gst_sdp_media_bandwidths_len (media);
1211 for (i = 0; i < len; i++) {
1212 const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
1214 if (strcmp (bw->bwtype, type) == 0)
1217 /* then look in the message specific section */
1218 len = gst_sdp_message_bandwidths_len (sdp);
1219 for (i = 0; i < len; i++) {
1220 const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
1222 if (strcmp (bw->bwtype, type) == 0)
1229 gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1230 const GstSDPMedia * media, GstRTSPStream * stream)
1232 const GstSDPBandwidth *bw;
1234 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
1235 stream->as_bandwidth = bw->bandwidth;
1237 stream->as_bandwidth = -1;
1239 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
1240 stream->rr_bandwidth = bw->bandwidth;
1242 stream->rr_bandwidth = -1;
1244 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
1245 stream->rs_bandwidth = bw->bandwidth;
1247 stream->rs_bandwidth = -1;
1251 gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
1252 const GstSDPConnection * conn)
1254 if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
1257 if (conn->addrtype == NULL)
1260 /* check for IPV6 */
1261 if (strcmp (conn->addrtype, "IP4") == 0)
1262 stream->is_ipv6 = FALSE;
1263 else if (strcmp (conn->addrtype, "IP6") == 0)
1264 stream->is_ipv6 = TRUE;
1269 g_free (stream->destination);
1270 stream->destination = g_strdup (conn->address);
1272 /* check for multicast */
1273 stream->is_multicast =
1274 gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
1276 stream->ttl = conn->ttl;
1279 /* Go over the connections for a stream.
1280 * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
1282 * - If we are dealing with a localhost address, we disable multicast
1285 gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
1286 const GstSDPMedia * media, GstRTSPStream * stream)
1288 const GstSDPConnection *conn;
1291 /* first look in the media specific section */
1292 len = gst_sdp_media_connections_len (media);
1293 for (i = 0; i < len; i++) {
1294 conn = gst_sdp_media_get_connection (media, i);
1296 gst_rtspsrc_do_stream_connection (src, stream, conn);
1298 /* then look in the message specific section */
1299 if ((conn = gst_sdp_message_get_connection (sdp))) {
1300 gst_rtspsrc_do_stream_connection (src, stream, conn);
1305 /* m=<media> <UDP port> RTP/AVP <payload>
1308 gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
1309 const GstSDPMedia * media, GstRTSPStream * stream)
1315 proto = gst_sdp_media_get_proto (media);
1319 if (g_str_equal (proto, "RTP/AVP"))
1320 stream->profile = GST_RTSP_PROFILE_AVP;
1321 else if (g_str_equal (proto, "RTP/SAVP"))
1322 stream->profile = GST_RTSP_PROFILE_SAVP;
1323 else if (g_str_equal (proto, "RTP/AVPF"))
1324 stream->profile = GST_RTSP_PROFILE_AVPF;
1325 else if (g_str_equal (proto, "RTP/SAVPF"))
1326 stream->profile = GST_RTSP_PROFILE_SAVPF;
1330 len = gst_sdp_media_formats_len (media);
1331 for (i = 0; i < len; i++) {
1338 pt = atoi (gst_sdp_media_get_format (media, i));
1340 GST_DEBUG_OBJECT (src, " looking at %d pt: %d", i, pt);
1343 caps = gst_rtspsrc_media_to_caps (pt, media);
1345 GST_WARNING_OBJECT (src, " skipping pt %d without caps", pt);
1349 /* do some tweaks */
1350 s = gst_caps_get_structure (caps, 0);
1351 if ((enc = gst_structure_get_string (s, "encoding-name"))) {
1352 stream->is_real = (strstr (enc, "-REAL") != NULL);
1353 if (strcmp (enc, "X-ASF-PF") == 0)
1354 stream->container = TRUE;
1356 GST_DEBUG ("mapping sdp session level attributes to caps");
1357 gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, caps);
1358 GST_DEBUG ("mapping sdp media level attributes to caps");
1359 gst_rtspsrc_sdp_attributes_to_caps (media->attributes, caps);
1361 /* the first pt will be the default */
1362 if (stream->ptmap->len == 0)
1363 stream->default_pt = pt;
1367 g_array_append_val (stream->ptmap, item);
1370 if (stream->pt >= 96) {
1371 /* If we have a dynamic payload type, see if we have a stream with the
1372 * same payload number. If there is one, they are part of the same
1373 * container and we only need to add one pad. */
1374 if (find_stream (src, &stream->pt, (gpointer) find_stream_by_pt)) {
1375 stream->container = TRUE;
1376 GST_DEBUG ("found another stream with pt %d, marking as container",
1386 GST_ERROR_OBJECT (src, "can't find proto in media");
1391 GST_ERROR_OBJECT (src, "unknown proto in media %s", proto);
1396 static const gchar *
1397 get_aggregate_control (GstRTSPSrc * src)
1402 base = src->control;
1403 else if (src->content_base)
1404 base = src->content_base;
1405 else if (src->conninfo.url_str)
1406 base = src->conninfo.url_str;
1413 static GstRTSPStream *
1414 gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
1416 GstRTSPStream *stream;
1417 const gchar *control_url;
1418 const GstSDPMedia *media;
1420 /* get media, should not return NULL */
1421 media = gst_sdp_message_get_media (sdp, idx);
1425 stream = g_new0 (GstRTSPStream, 1);
1426 stream->parent = src;
1427 /* we mark the pad as not linked, we will mark it as OK when we add the pad to
1429 stream->last_ret = GST_FLOW_NOT_LINKED;
1430 stream->added = FALSE;
1431 stream->disabled = FALSE;
1433 stream->eos = FALSE;
1434 stream->discont = TRUE;
1435 stream->seqbase = -1;
1436 stream->timebase = -1;
1437 stream->profile = GST_RTSP_PROFILE_AVP;
1438 stream->ptmap = g_array_new (FALSE, FALSE, sizeof (PtMapItem));
1440 /* collect bandwidth information for this steam. FIXME, configure in the RTP
1441 * session manager to scale RTCP. */
1442 gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
1444 /* collect connection info */
1445 gst_rtspsrc_collect_connections (src, sdp, media, stream);
1447 /* make the payload type map */
1448 gst_rtspsrc_collect_payloads (src, sdp, media, stream);
1450 /* collect port number */
1451 stream->port = gst_sdp_media_get_port (media);
1453 /* get control url to construct the setup url. The setup url is used to
1454 * configure the transport of the stream and is used to identity the stream in
1455 * the RTP-Info header field returned from PLAY. */
1456 control_url = gst_sdp_media_get_attribute_val (media, "control");
1457 if (control_url == NULL)
1458 control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
1460 GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
1461 GST_DEBUG_OBJECT (src, " port: %d", stream->port);
1462 GST_DEBUG_OBJECT (src, " container: %d", stream->container);
1463 GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
1465 if (control_url != NULL) {
1466 stream->control_url = g_strdup (control_url);
1467 /* Build a fully qualified url using the content_base if any or by prefixing
1468 * the original request.
1469 * If the control_url starts with a '/' or a non rtsp: protocol we will most
1470 * likely build a URL that the server will fail to understand, this is ok,
1471 * we will fail then. */
1472 if (g_str_has_prefix (control_url, "rtsp://"))
1473 stream->conninfo.location = g_strdup (control_url);
1478 if (g_strcmp0 (control_url, "*") == 0)
1481 base = get_aggregate_control (src);
1483 /* check if the base ends or control starts with / */
1484 has_slash = g_str_has_prefix (control_url, "/");
1485 has_slash = has_slash || g_str_has_suffix (base, "/");
1487 /* concatenate the two strings, insert / when not present */
1488 stream->conninfo.location =
1489 g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
1492 GST_DEBUG_OBJECT (src, " setup: %s",
1493 GST_STR_NULL (stream->conninfo.location));
1495 /* we keep track of all streams */
1496 src->streams = g_list_append (src->streams, stream);
1504 gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
1508 GST_DEBUG_OBJECT (src, "free stream %p", stream);
1510 g_array_free (stream->ptmap, TRUE);
1512 g_free (stream->destination);
1513 g_free (stream->control_url);
1514 g_free (stream->conninfo.location);
1516 for (i = 0; i < 2; i++) {
1517 if (stream->udpsrc[i]) {
1518 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
1519 gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
1520 gst_object_unref (stream->udpsrc[i]);
1521 stream->udpsrc[i] = NULL;
1523 if (stream->channelpad[i]) {
1524 gst_object_unref (stream->channelpad[i]);
1525 stream->channelpad[i] = NULL;
1527 if (stream->udpsink[i]) {
1528 gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
1529 gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
1530 gst_object_unref (stream->udpsink[i]);
1531 stream->udpsink[i] = NULL;
1534 if (stream->fakesrc) {
1535 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
1536 gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
1537 gst_object_unref (stream->fakesrc);
1538 stream->fakesrc = NULL;
1540 if (stream->srcpad) {
1541 gst_pad_set_active (stream->srcpad, FALSE);
1542 if (stream->added) {
1543 gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
1544 stream->added = FALSE;
1546 stream->srcpad = NULL;
1548 if (stream->rtcppad) {
1549 gst_object_unref (stream->rtcppad);
1550 stream->rtcppad = NULL;
1552 if (stream->session) {
1553 g_object_unref (stream->session);
1554 stream->session = NULL;
1560 gst_rtspsrc_cleanup (GstRTSPSrc * src)
1564 GST_DEBUG_OBJECT (src, "cleanup");
1566 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1567 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1569 gst_rtspsrc_stream_free (src, stream);
1571 g_list_free (src->streams);
1572 src->streams = NULL;
1574 if (src->manager_sig_id) {
1575 g_signal_handler_disconnect (src->manager, src->manager_sig_id);
1576 src->manager_sig_id = 0;
1578 gst_element_set_state (src->manager, GST_STATE_NULL);
1579 gst_bin_remove (GST_BIN_CAST (src), src->manager);
1580 src->manager = NULL;
1583 gst_structure_free (src->props);
1586 g_free (src->content_base);
1587 src->content_base = NULL;
1589 g_free (src->control);
1590 src->control = NULL;
1593 gst_rtsp_range_free (src->range);
1596 /* don't clear the SDP when it was used in the url */
1597 if (src->sdp && !src->from_sdp) {
1598 gst_sdp_message_free (src->sdp);
1601 if (src->start_segment) {
1602 gst_event_unref (src->start_segment);
1603 src->start_segment = NULL;
1605 if (src->provided_clock) {
1606 gst_object_unref (src->provided_clock);
1607 src->provided_clock = NULL;
1611 #define PARSE_INT(p, del, res) \
1614 p = strstr (p, del); \
1624 #define PARSE_STRING(p, del, res) \
1627 p = strstr (p, del); \
1639 #define SKIP_SPACES(p) \
1640 while (*p && g_ascii_isspace (*p)) \
1645 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1648 gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
1649 gint * rate, gchar ** params)
1653 p = (gchar *) rtpmap;
1655 PARSE_INT (p, " ", *payload);
1663 PARSE_STRING (p, "/", *name);
1664 if (*name == NULL) {
1665 GST_DEBUG ("no rate, name %s", p);
1666 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
1667 * streams seem to omit the rate. */
1674 p = strstr (p, "/");
1692 * Mapping SDP attributes to caps
1694 * prepend 'a-' to IANA registered sdp attributes names
1695 * (ie: not prefixed with 'x-') in order to avoid
1696 * collision with gstreamer standard caps properties names
1699 gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
1701 if (attributes->len > 0) {
1705 s = gst_caps_get_structure (caps, 0);
1707 for (i = 0; i < attributes->len; i++) {
1708 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
1709 gchar *tofree, *key;
1713 /* skip some of the attribute we already handle */
1714 if (!strcmp (key, "fmtp"))
1716 if (!strcmp (key, "rtpmap"))
1718 if (!strcmp (key, "control"))
1720 if (!strcmp (key, "range"))
1723 /* string must be valid UTF8 */
1724 if (!g_utf8_validate (attr->value, -1, NULL))
1727 if (!g_str_has_prefix (key, "x-"))
1728 tofree = key = g_strdup_printf ("a-%s", key);
1732 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
1733 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
1739 static const gchar *
1740 rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
1749 if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
1752 if (sscanf (attr, "%d ", &val) != 1)
1762 * Mapping of caps to and from SDP fields:
1764 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1765 * a=fmtp:<payload> <param>[=<value>];...
1768 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
1771 const gchar *rtpmap;
1775 gchar *params = NULL;
1781 /* get and parse rtpmap */
1782 rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
1784 /* dynamic payloads need rtpmap or we fail */
1785 if (rtpmap == NULL && pt >= 96)
1788 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
1790 g_warning ("error parsing rtpmap, ignoring");
1793 /* check if we have a rate, if not, we need to look up the rate from the
1794 * default rates based on the payload types. */
1796 const GstRTPPayloadInfo *info;
1798 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
1799 /* dynamic types, use media and encoding_name */
1800 tmp = g_ascii_strdown (media->media, -1);
1801 info = gst_rtp_payload_info_for_name (tmp, name);
1804 /* static types, use payload type */
1805 info = gst_rtp_payload_info_for_pt (pt);
1809 if ((rate = info->clock_rate) == 0)
1812 /* we fail if we cannot find one */
1817 tmp = g_ascii_strdown (media->media, -1);
1818 caps = gst_caps_new_simple ("application/x-unknown",
1819 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
1821 s = gst_caps_get_structure (caps, 0);
1823 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
1825 /* encoding name must be upper case */
1827 tmp = g_ascii_strup (name, -1);
1828 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
1832 /* params must be lower case */
1833 if (params != NULL) {
1834 tmp = g_ascii_strdown (params, -1);
1835 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
1839 /* parse optional fmtp: field */
1840 if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
1846 /* p is now of the format <payload> <param>[=<value>];... */
1847 PARSE_INT (p, " ", payload);
1848 if (payload != -1 && payload == pt) {
1852 /* <param>[=<value>] are separated with ';' */
1853 pairs = g_strsplit (p, ";", 0);
1854 for (i = 0; pairs[i]; i++) {
1856 const gchar *val, *key;
1858 /* the key may not have a '=', the value can have other '='s */
1859 valpos = strstr (pairs[i], "=");
1861 /* we have a '=' and thus a value, remove the '=' with \0 */
1863 /* value is everything between '=' and ';'. We split the pairs at ;
1864 * boundaries so we can take the remainder of the value. Some servers
1865 * put spaces around the value which we strip off here. Alternatively
1866 * we could strip those spaces in the depayloaders should these spaces
1867 * actually carry any meaning in the future. */
1868 val = g_strstrip (valpos + 1);
1870 /* simple <param>;.. is translated into <param>=1;... */
1873 /* strip the key of spaces, convert key to lowercase but not the value. */
1874 key = g_strstrip (pairs[i]);
1875 if (strlen (key) > 1) {
1876 tmp = g_ascii_strdown (key, -1);
1877 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
1889 g_warning ("rtpmap type not given for dynamic payload %d", pt);
1894 g_warning ("rate unknown for payload type %d", pt);
1900 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
1901 gint * rtpport, gint * rtcpport)
1904 GstStateChangeReturn ret;
1905 GstElement *udpsrc0, *udpsrc1;
1906 gint tmp_rtp, tmp_rtcp;
1910 src = stream->parent;
1916 /* Start at next port */
1917 tmp_rtp = src->next_port_num;
1919 if (stream->is_ipv6)
1920 host = "udp://[::0]";
1922 host = "udp://0.0.0.0";
1924 /* try to allocate 2 UDP ports, the RTP port should be an even
1925 * number and the RTCP port should be the next (uneven) port */
1928 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
1929 tmp_rtp >= src->client_port_range.max)
1932 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
1933 if (udpsrc0 == NULL)
1934 goto no_udp_protocol;
1935 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
1937 if (src->udp_buffer_size != 0)
1938 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
1941 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
1942 if (ret == GST_STATE_CHANGE_FAILURE) {
1944 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
1947 if (++count > src->retry)
1950 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1951 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1952 gst_object_unref (udpsrc0);
1955 GST_DEBUG_OBJECT (src, "retry %d", count);
1958 goto no_udp_protocol;
1961 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
1962 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
1964 /* check if port is even */
1965 if ((tmp_rtp & 0x01) != 0) {
1966 /* port not even, close and allocate another */
1967 if (++count > src->retry)
1970 GST_DEBUG_OBJECT (src, "RTP port not even");
1972 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1973 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1974 gst_object_unref (udpsrc0);
1977 GST_DEBUG_OBJECT (src, "retry %d", count);
1982 /* allocate port+1 for RTCP now */
1983 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
1984 if (udpsrc1 == NULL)
1985 goto no_udp_rtcp_protocol;
1988 tmp_rtcp = tmp_rtp + 1;
1989 if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
1992 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
1994 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
1995 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
1996 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
1997 if (ret == GST_STATE_CHANGE_FAILURE) {
1998 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
2000 if (++count > src->retry)
2003 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2004 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2005 gst_object_unref (udpsrc0);
2008 GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
2009 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2010 gst_object_unref (udpsrc1);
2014 GST_DEBUG_OBJECT (src, "retry %d", count);
2018 /* all fine, do port check */
2019 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
2020 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
2022 /* this should not happen... */
2023 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
2026 /* we keep these elements, we configure all in configure_transport when the
2027 * server told us to really use the UDP ports. */
2028 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
2029 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
2030 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
2031 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
2033 /* keep track of next available port number when we have a range
2035 if (src->next_port_num != 0)
2036 src->next_port_num = tmp_rtcp + 1;
2043 GST_DEBUG_OBJECT (src, "could not get UDP source");
2048 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
2052 no_udp_rtcp_protocol:
2054 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
2059 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
2060 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
2066 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2067 gst_object_unref (udpsrc0);
2070 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2071 gst_object_unref (udpsrc1);
2078 gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
2083 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
2085 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2086 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2089 for (i = 0; i < 2; i++) {
2090 if (stream->udpsrc[i])
2091 gst_element_set_state (stream->udpsrc[i], state);
2097 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
2104 event = gst_event_new_flush_start ();
2105 GST_DEBUG_OBJECT (src, "start flush");
2107 state = GST_STATE_PAUSED;
2109 event = gst_event_new_flush_stop (FALSE);
2110 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
2113 state = GST_STATE_PLAYING;
2115 state = GST_STATE_PAUSED;
2117 gst_rtspsrc_push_event (src, event);
2118 gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
2119 gst_rtspsrc_set_state (src, state);
2122 static GstRTSPResult
2123 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
2124 GstRTSPMessage * message, GTimeVal * timeout)
2129 ret = gst_rtsp_connection_send (conn, message, timeout);
2131 ret = GST_RTSP_ERROR;
2136 static GstRTSPResult
2137 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
2138 GstRTSPMessage * message, GTimeVal * timeout)
2143 ret = gst_rtsp_connection_receive (conn, message, timeout);
2145 ret = GST_RTSP_ERROR;
2151 gst_rtspsrc_get_position (GstRTSPSrc * src)
2156 query = gst_query_new_position (GST_FORMAT_TIME);
2157 /* should be known somewhere down the stream (e.g. jitterbuffer) */
2158 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2159 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2163 if (stream->srcpad) {
2164 if (gst_pad_query (stream->srcpad, query)) {
2165 gst_query_parse_position (query, &fmt, &pos);
2166 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
2167 GST_TIME_ARGS (pos));
2168 src->last_pos = pos;
2178 gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
2180 src->state = GST_RTSP_STATE_SEEKING;
2181 /* PLAY will add the range header now. */
2182 src->need_range = TRUE;
2188 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
2193 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
2195 gboolean flush, skip;
2198 GstSegment seeksegment = { 0, };
2202 GST_DEBUG_OBJECT (src, "doing seek with event");
2204 gst_event_parse_seek (event, &rate, &format, &flags,
2205 &cur_type, &cur, &stop_type, &stop);
2207 /* no negative rates yet */
2211 /* we need TIME format */
2212 if (format != src->segment.format)
2215 GST_DEBUG_OBJECT (src, "doing seek without event");
2217 cur_type = GST_SEEK_TYPE_SET;
2218 stop_type = GST_SEEK_TYPE_SET;
2221 /* get flush flag */
2222 flush = flags & GST_SEEK_FLAG_FLUSH;
2223 skip = flags & GST_SEEK_FLAG_SKIP;
2225 /* now we need to make sure the streaming thread is stopped. We do this by
2226 * either sending a FLUSH_START event downstream which will cause the
2227 * streaming thread to stop with a WRONG_STATE.
2228 * For a non-flushing seek we simply pause the task, which will happen as soon
2229 * as it completes one iteration (and thus might block when the sink is
2230 * blocking in preroll). */
2232 GST_DEBUG_OBJECT (src, "starting flush");
2233 gst_rtspsrc_flush (src, TRUE, FALSE);
2236 gst_task_pause (src->task);
2240 /* we should now be able to grab the streaming thread because we stopped it
2241 * with the above flush/pause code */
2242 GST_RTSP_STREAM_LOCK (src);
2244 GST_DEBUG_OBJECT (src, "stopped streaming");
2246 /* stop flushing the rtsp connection so we can send PAUSE/PLAY below */
2247 gst_rtspsrc_connection_flush (src, FALSE);
2249 /* copy segment, we need this because we still need the old
2250 * segment when we close the current segment. */
2251 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
2253 /* configure the seek parameters in the seeksegment. We will then have the
2254 * right values in the segment to perform the seek */
2256 GST_DEBUG_OBJECT (src, "configuring seek");
2257 gst_segment_do_seek (&seeksegment, rate, format, flags,
2258 cur_type, cur, stop_type, stop, &update);
2261 /* figure out the last position we need to play. If it's configured (stop !=
2262 * -1), use that, else we play until the total duration of the file */
2263 if ((stop = seeksegment.stop) == -1)
2264 stop = seeksegment.duration;
2266 playing = (src->state == GST_RTSP_STATE_PLAYING);
2268 /* if we were playing, pause first */
2270 /* obtain current position in case seek fails */
2271 gst_rtspsrc_get_position (src);
2272 gst_rtspsrc_pause (src, FALSE);
2276 gst_rtspsrc_do_seek (src, &seeksegment);
2278 /* and continue playing */
2280 gst_rtspsrc_play (src, &seeksegment, FALSE);
2282 /* prepare for streaming again */
2284 /* if we started flush, we stop now */
2285 GST_DEBUG_OBJECT (src, "stopping flush");
2286 gst_rtspsrc_flush (src, FALSE, playing);
2289 /* now we did the seek and can activate the new segment values */
2290 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
2292 /* if we're doing a segment seek, post a SEGMENT_START message */
2293 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2294 gst_element_post_message (GST_ELEMENT_CAST (src),
2295 gst_message_new_segment_start (GST_OBJECT_CAST (src),
2296 src->segment.format, src->segment.position));
2299 /* now create the newsegment */
2300 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
2301 " to %" G_GINT64_FORMAT, src->segment.position, stop);
2304 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
2305 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2306 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2307 stream->discont = TRUE;
2310 GST_RTSP_STREAM_UNLOCK (src);
2317 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
2322 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
2328 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
2332 gboolean res = TRUE;
2335 src = GST_RTSPSRC_CAST (parent);
2337 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
2338 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
2340 switch (GST_EVENT_TYPE (event)) {
2341 case GST_EVENT_SEEK:
2342 res = gst_rtspsrc_perform_seek (src, event);
2346 case GST_EVENT_NAVIGATION:
2347 case GST_EVENT_LATENCY:
2355 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
2356 res = gst_pad_send_event (target, event);
2357 gst_object_unref (target);
2359 gst_event_unref (event);
2362 gst_event_unref (event);
2368 /* this is the final event function we receive on the internal source pad when
2369 * we deal with TCP connections */
2371 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
2376 GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
2378 switch (GST_EVENT_TYPE (event)) {
2379 case GST_EVENT_SEEK:
2381 case GST_EVENT_NAVIGATION:
2382 case GST_EVENT_LATENCY:
2384 gst_event_unref (event);
2391 /* this is the final query function we receive on the internal source pad when
2392 * we deal with TCP connections */
2394 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
2398 gboolean res = TRUE;
2400 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
2402 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2403 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2405 switch (GST_QUERY_TYPE (query)) {
2406 case GST_QUERY_POSITION:
2411 case GST_QUERY_DURATION:
2415 gst_query_parse_duration (query, &format, NULL);
2418 case GST_FORMAT_TIME:
2419 gst_query_set_duration (query, format, src->segment.duration);
2427 case GST_QUERY_LATENCY:
2429 /* we are live with a min latency of 0 and unlimited max latency, this
2430 * result will be updated by the session manager if there is any. */
2431 gst_query_set_latency (query, TRUE, 0, -1);
2441 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2443 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2447 gboolean res = FALSE;
2449 src = GST_RTSPSRC_CAST (parent);
2451 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2452 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2454 switch (GST_QUERY_TYPE (query)) {
2455 case GST_QUERY_DURATION:
2459 gst_query_parse_duration (query, &format, NULL);
2462 case GST_FORMAT_TIME:
2463 gst_query_set_duration (query, format, src->segment.duration);
2471 case GST_QUERY_SEEKING:
2475 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2476 if (format == GST_FORMAT_TIME) {
2478 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2480 /* seeking without duration is unlikely */
2481 seekable = seekable && src->seekable && src->segment.duration &&
2482 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2484 /* FIXME ?? should we have 0 and segment.duration here; see demuxers */
2485 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
2486 src->segment.start, src->segment.stop);
2495 uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
2497 gst_query_set_uri (query, uri);
2505 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2507 /* forward the query to the proxy target pad */
2509 res = gst_pad_query (target, query);
2510 gst_object_unref (target);
2519 /* callback for RTCP messages to be sent to the server when operating in TCP
2521 static GstFlowReturn
2522 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2525 GstRTSPStream *stream;
2526 GstFlowReturn res = GST_FLOW_OK;
2531 GstRTSPMessage message = { 0 };
2532 GstRTSPConnection *conn;
2534 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2535 src = stream->parent;
2537 gst_buffer_map (buffer, &map, GST_MAP_READ);
2541 gst_rtsp_message_init_data (&message, stream->channel[1]);
2543 /* lend the body data to the message */
2544 gst_rtsp_message_take_body (&message, data, size);
2546 if (stream->conninfo.connection)
2547 conn = stream->conninfo.connection;
2549 conn = src->conninfo.connection;
2551 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2552 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2553 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2555 /* and steal it away again because we will free it when unreffing the
2557 gst_rtsp_message_steal_body (&message, &data, &size);
2558 gst_rtsp_message_unset (&message);
2560 gst_buffer_unmap (buffer, &map);
2561 gst_buffer_unref (buffer);
2566 static GstPadProbeReturn
2567 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2569 GstRTSPSrc *src = user_data;
2571 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2572 GST_DEBUG_PAD_NAME (pad));
2574 /* activate the streams */
2575 GST_OBJECT_LOCK (src);
2576 if (!src->need_activate)
2579 src->need_activate = FALSE;
2580 GST_OBJECT_UNLOCK (src);
2582 gst_rtspsrc_activate_streams (src);
2584 return GST_PAD_PROBE_OK;
2588 GST_OBJECT_UNLOCK (src);
2589 return GST_PAD_PROBE_OK;
2593 /* this callback is called when the session manager generated a new src pad with
2594 * payloaded RTP packets. We simply ghost the pad here. */
2596 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2599 GstPadTemplate *template;
2602 GstRTSPStream *stream;
2605 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2607 GST_RTSP_STATE_LOCK (src);
2609 name = gst_object_get_name (GST_OBJECT_CAST (pad));
2610 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
2611 goto unknown_stream;
2613 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);
2615 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
2617 goto unknown_stream;
2620 stream->ssrc = ssrc;
2622 /* we'll add it later see below */
2623 stream->added = TRUE;
2625 /* check if we added all streams */
2627 for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
2628 GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;
2630 GST_DEBUG_OBJECT (src, "stream %p, container %d, disabled %d, added %d",
2631 ostream, ostream->container, ostream->disabled, ostream->added);
2633 /* a container stream only needs one pad added. Also disabled streams don't
2635 if (!ostream->container && !ostream->disabled && !ostream->added) {
2640 GST_RTSP_STATE_UNLOCK (src);
2642 /* create a new pad we will use to stream to */
2643 template = gst_static_pad_template_get (&rtptemplate);
2644 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
2645 gst_object_unref (template);
2648 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
2649 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
2650 gst_pad_set_active (stream->srcpad, TRUE);
2651 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
2654 GST_DEBUG_OBJECT (src, "We added all streams");
2655 /* when we get here, all stream are added and we can fire the no-more-pads
2657 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
2665 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
2666 GST_RTSP_STATE_UNLOCK (src);
2673 stream_get_caps_for_pt (GstRTSPStream * stream, guint pt)
2677 len = stream->ptmap->len;
2678 for (i = 0; i < len; i++) {
2679 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
2687 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
2689 GstRTSPStream *stream;
2692 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
2694 GST_RTSP_STATE_LOCK (src);
2695 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2697 goto unknown_stream;
2699 if ((caps = stream_get_caps_for_pt (stream, pt)))
2700 gst_caps_ref (caps);
2701 GST_RTSP_STATE_UNLOCK (src);
2707 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
2708 GST_RTSP_STATE_UNLOCK (src);
2714 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
2716 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
2722 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
2728 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
2734 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2736 GstRTSPSrc *src = stream->parent;
2739 g_object_get (source, "ssrc", &ssrc, NULL);
2741 GST_DEBUG_OBJECT (src, "source %08x, stream %08x, session %u received BYE",
2742 ssrc, stream->ssrc, stream->id);
2744 if (ssrc == stream->ssrc)
2745 gst_rtspsrc_do_stream_eos (src, stream);
2749 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2751 GstRTSPSrc *src = stream->parent;
2754 g_object_get (source, "ssrc", &ssrc, NULL);
2756 GST_WARNING_OBJECT (src, "source %08x, stream %08x in session %u timed out",
2757 ssrc, stream->ssrc, stream->id);
2759 if (ssrc == stream->ssrc)
2760 gst_rtspsrc_do_stream_eos (src, stream);
2764 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
2766 GstRTSPStream *stream;
2768 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
2770 /* get stream for session */
2771 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2773 gst_rtspsrc_do_stream_eos (src, stream);
2778 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
2780 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
2785 set_manager_buffer_mode (GstRTSPSrc * src)
2787 GObjectClass *klass;
2789 if (src->manager == NULL)
2792 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
2794 if (!g_object_class_find_property (klass, "buffer-mode"))
2797 if (src->buffer_mode != BUFFER_MODE_AUTO) {
2798 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
2803 GST_DEBUG_OBJECT (src,
2804 "auto buffering mode, have clock %" GST_PTR_FORMAT, src->provided_clock);
2806 if (src->provided_clock) {
2807 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
2809 if (clock == src->provided_clock) {
2810 GST_DEBUG_OBJECT (src, "selected synced");
2811 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SYNCED, NULL);
2814 gst_object_unref (clock);
2819 /* Otherwise fall-through and use another buffer mode */
2821 gst_object_unref (clock);
2824 GST_DEBUG_OBJECT (src, "auto buffering mode");
2825 if (src->use_buffering) {
2826 GST_DEBUG_OBJECT (src, "selected buffer");
2827 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER, NULL);
2829 GST_DEBUG_OBJECT (src, "selected slave");
2830 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
2834 /* try to get and configure a manager */
2836 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
2837 GstRTSPTransport * transport)
2839 const gchar *manager;
2841 GstStateChangeReturn ret;
2843 /* find a manager */
2844 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
2848 GST_DEBUG_OBJECT (src, "using manager %s", manager);
2850 /* configure the manager */
2851 if (src->manager == NULL) {
2852 GObjectClass *klass;
2854 if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
2856 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
2860 goto use_no_manager;
2862 if (!(src->manager = gst_element_factory_make (manager, "manager")))
2863 goto manager_failed;
2866 /* we manage this element */
2867 gst_element_set_locked_state (src->manager, TRUE);
2868 gst_bin_add (GST_BIN_CAST (src), src->manager);
2870 ret = gst_element_set_state (src->manager, GST_STATE_PAUSED);
2871 if (ret == GST_STATE_CHANGE_FAILURE)
2872 goto start_manager_failure;
2874 g_object_set (src->manager, "latency", src->latency, NULL);
2876 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
2878 if (g_object_class_find_property (klass, "ntp-sync")) {
2879 g_object_set (src->manager, "ntp-sync", src->ntp_sync, NULL);
2882 if (g_object_class_find_property (klass, "use-pipeline-clock")) {
2883 g_object_set (src->manager, "use-pipeline-clock",
2884 src->use_pipeline_clock, NULL);
2887 if (src->sdes && g_object_class_find_property (klass, "sdes")) {
2888 g_object_set (src->manager, "sdes", src->sdes, NULL);
2891 if (g_object_class_find_property (klass, "drop-on-latency")) {
2892 g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
2896 /* buffer mode pauses are handled by adding offsets to buffer times,
2897 * but some depayloaders may have a hard time syncing output times
2898 * with such input times, e.g. container ones, most notably ASF */
2899 /* TODO alternatives are having an event that indicates these shifts,
2900 * or having rtsp extensions provide suggestion on buffer mode */
2901 /* valid duration implies not likely live pipeline,
2902 * so slaving in jitterbuffer does not make much sense
2903 * (and might mess things up due to bursts) */
2904 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
2905 src->segment.duration && !stream->container) {
2906 src->use_buffering = TRUE;
2908 src->use_buffering = FALSE;
2911 set_manager_buffer_mode (src);
2913 /* connect to signals */
2914 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
2916 src->manager_sig_id =
2917 g_signal_connect (src->manager, "pad-added",
2918 (GCallback) new_manager_pad, src);
2919 src->manager_ptmap_id =
2920 g_signal_connect (src->manager, "request-pt-map",
2921 (GCallback) request_pt_map, src);
2923 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
2926 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_NEW_MANAGER], 0,
2930 /* we stream directly to the manager, get some pads. Each RTSP stream goes
2931 * into a separate RTP session. */
2932 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
2933 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
2935 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
2936 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
2939 /* now configure the bandwidth in the manager */
2940 if (g_signal_lookup ("get-internal-session",
2941 G_OBJECT_TYPE (src->manager)) != 0) {
2942 GObject *rtpsession;
2944 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
2947 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
2949 stream->session = rtpsession;
2951 if (stream->as_bandwidth != -1) {
2952 GST_INFO_OBJECT (src, "setting AS: %f",
2953 (gdouble) (stream->as_bandwidth * 1000));
2954 g_object_set (rtpsession, "bandwidth",
2955 (gdouble) (stream->as_bandwidth * 1000), NULL);
2957 if (stream->rr_bandwidth != -1) {
2958 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
2959 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
2962 if (stream->rs_bandwidth != -1) {
2963 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
2964 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
2968 g_object_set (rtpsession, "probation", src->probation, NULL);
2970 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2972 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
2974 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
2976 g_signal_connect (rtpsession, "on-ssrc-active",
2977 (GCallback) on_ssrc_active, stream);
2988 GST_DEBUG_OBJECT (src, "cannot get a session manager");
2993 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
2996 start_manager_failure:
2998 GST_DEBUG_OBJECT (src, "could not start session manager");
3003 /* free the UDP sources allocated when negotiating a transport.
3004 * This function is called when the server negotiated to a transport where the
3005 * UDP sources are not needed anymore, such as TCP or multicast. */
3007 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
3011 for (i = 0; i < 2; i++) {
3012 if (stream->udpsrc[i]) {
3013 GST_DEBUG ("free UDP source %d for stream %p", i, stream);
3014 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
3015 gst_object_unref (stream->udpsrc[i]);
3016 stream->udpsrc[i] = NULL;
3021 /* for TCP, create pads to send and receive data to and from the manager and to
3022 * intercept various events and queries
3025 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
3026 GstRTSPTransport * transport, GstPad ** outpad)
3029 GstPadTemplate *template;
3030 GstPad *pad0, *pad1;
3032 /* configure for interleaved delivery, nothing needs to be done
3033 * here, the loop function will call the chain functions of the
3034 * session manager. */
3035 stream->channel[0] = transport->interleaved.min;
3036 stream->channel[1] = transport->interleaved.max;
3037 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
3038 stream->channel[0], stream->channel[1]);
3040 /* we can remove the allocated UDP ports now */
3041 gst_rtspsrc_stream_free_udp (stream);
3043 /* no session manager, send data to srcpad directly */
3044 if (!stream->channelpad[0]) {
3045 GST_DEBUG_OBJECT (src, "no manager, creating pad");
3047 /* create a new pad we will use to stream to */
3048 name = g_strdup_printf ("stream_%u", stream->id);
3049 template = gst_static_pad_template_get (&rtptemplate);
3050 stream->channelpad[0] = gst_pad_new_from_template (template, name);
3051 gst_object_unref (template);
3054 /* set caps and activate */
3055 gst_pad_use_fixed_caps (stream->channelpad[0]);
3056 gst_pad_set_active (stream->channelpad[0], TRUE);
3058 *outpad = gst_object_ref (stream->channelpad[0]);
3060 GST_DEBUG_OBJECT (src, "using manager source pad");
3062 template = gst_static_pad_template_get (&anysrctemplate);
3064 /* allocate pads for sending the channel data into the manager */
3065 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
3066 gst_pad_link_full (pad0, stream->channelpad[0], GST_PAD_LINK_CHECK_NOTHING);
3067 gst_object_unref (stream->channelpad[0]);
3068 stream->channelpad[0] = pad0;
3069 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
3070 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
3071 gst_pad_set_element_private (pad0, src);
3072 gst_pad_set_active (pad0, TRUE);
3074 if (stream->channelpad[1]) {
3075 /* if we have a sinkpad for the other channel, create a pad and link to the
3077 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
3078 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
3079 gst_pad_link_full (pad1, stream->channelpad[1],
3080 GST_PAD_LINK_CHECK_NOTHING);
3081 gst_object_unref (stream->channelpad[1]);
3082 stream->channelpad[1] = pad1;
3083 gst_pad_set_active (pad1, TRUE);
3085 gst_object_unref (template);
3087 /* setup RTCP transport back to the server if we have to. */
3088 if (src->manager && src->do_rtcp) {
3091 template = gst_static_pad_template_get (&anysinktemplate);
3093 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
3094 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
3095 gst_pad_set_element_private (stream->rtcppad, stream);
3096 gst_pad_set_active (stream->rtcppad, TRUE);
3098 /* get session RTCP pad */
3099 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3100 pad = gst_element_get_request_pad (src->manager, name);
3105 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3106 gst_object_unref (pad);
3109 gst_object_unref (template);
3115 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
3116 GstRTSPTransport * transport, const gchar ** destination, gint * min,
3117 gint * max, guint * ttl)
3119 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3121 if (!(*destination = transport->destination))
3122 *destination = stream->destination;
3125 /* transport first */
3126 *min = transport->port.min;
3127 *max = transport->port.max;
3128 if (*min == -1 && *max == -1) {
3129 /* then try from SDP */
3130 if (stream->port != 0) {
3131 *min = stream->port;
3132 *max = stream->port + 1;
3138 if (!(*ttl = transport->ttl))
3143 /* first take the source, then the endpoint to figure out where to send
3145 if (!(*destination = transport->source)) {
3146 if (src->conninfo.connection)
3147 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
3148 else if (stream->conninfo.connection)
3150 gst_rtsp_connection_get_ip (stream->conninfo.connection);
3154 /* for unicast we only expect the ports here */
3155 *min = transport->server_port.min;
3156 *max = transport->server_port.max;
3161 /* For multicast create UDP sources and join the multicast group. */
3163 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
3164 GstRTSPTransport * transport, GstPad ** outpad)
3167 const gchar *destination;
3170 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
3172 /* we can remove the allocated UDP ports now */
3173 gst_rtspsrc_stream_free_udp (stream);
3175 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
3178 /* we need a destination now */
3179 if (destination == NULL)
3180 goto no_destination;
3182 /* we really need ports now or we won't be able to receive anything at all */
3183 if (min == -1 && max == -1)
3186 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
3187 destination, min, max);
3189 /* creating UDP source for RTP */
3191 uri = g_strdup_printf ("udp://%s:%d", destination, min);
3193 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3195 if (stream->udpsrc[0] == NULL)
3198 /* take ownership */
3199 gst_object_ref_sink (stream->udpsrc[0]);
3201 if (src->udp_buffer_size != 0)
3202 g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
3203 src->udp_buffer_size, NULL);
3205 if (src->multi_iface != NULL)
3206 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3207 src->multi_iface, NULL);
3210 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3211 gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED);
3214 /* creating another UDP source for RTCP */
3218 uri = g_strdup_printf ("udp://%s:%d", destination, max);
3220 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3222 if (stream->udpsrc[1] == NULL)
3225 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3226 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3227 gst_caps_unref (caps);
3229 /* take ownership */
3230 gst_object_ref_sink (stream->udpsrc[1]);
3232 if (src->multi_iface != NULL)
3233 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3234 src->multi_iface, NULL);
3236 gst_element_set_state (stream->udpsrc[1], GST_STATE_PAUSED);
3243 GST_DEBUG_OBJECT (src, "no UDP source element found");
3248 GST_DEBUG_OBJECT (src, "no destination found");
3253 GST_DEBUG_OBJECT (src, "no ports found");
3258 /* configure the remainder of the UDP ports */
3260 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
3261 GstRTSPTransport * transport, GstPad ** outpad)
3263 /* we manage the UDP elements now. For unicast, the UDP sources where
3264 * allocated in the stream when we suggested a transport. */
3265 if (stream->udpsrc[0]) {
3266 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3267 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
3269 GST_DEBUG_OBJECT (src, "setting up UDP source");
3271 /* configure a timeout on the UDP port. When the timeout message is
3272 * posted, we assume UDP transport is not possible. We reconnect using TCP
3274 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
3275 src->udp_timeout * 1000, NULL);
3277 /* get output pad of the UDP source. */
3278 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
3280 /* save it so we can unblock */
3281 stream->blockedpad = *outpad;
3283 /* configure pad block on the pad. As soon as there is dataflow on the
3284 * UDP source, we know that UDP is not blocked by a firewall and we can
3285 * configure all the streams to let the application autoplug decoders. */
3287 gst_pad_add_probe (stream->blockedpad,
3288 GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, pad_blocked, src, NULL);
3290 if (stream->channelpad[0]) {
3291 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
3292 /* configure for UDP delivery, we need to connect the UDP pads to
3293 * the session plugin. */
3294 gst_pad_link_full (*outpad, stream->channelpad[0],
3295 GST_PAD_LINK_CHECK_NOTHING);
3296 gst_object_unref (*outpad);
3298 /* we connected to pad-added signal to get pads from the manager */
3300 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
3305 if (stream->udpsrc[1]) {
3308 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
3309 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
3311 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3312 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3313 gst_caps_unref (caps);
3315 if (stream->channelpad[1]) {
3318 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
3320 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
3321 gst_pad_link_full (pad, stream->channelpad[1],
3322 GST_PAD_LINK_CHECK_NOTHING);
3323 gst_object_unref (pad);
3325 /* leave unlinked */
3331 /* configure the UDP sink back to the server for status reports */
3333 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
3334 GstRTSPStream * stream, GstRTSPTransport * transport)
3337 gint rtp_port, rtcp_port;
3338 gboolean do_rtp, do_rtcp;
3339 const gchar *destination;
3344 /* get transport info */
3345 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
3346 &rtp_port, &rtcp_port, &ttl);
3348 /* see what we need to do */
3349 do_rtp = (rtp_port != -1);
3350 /* it's possible that the server does not want us to send RTCP in which case
3352 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
3354 /* we need a destination when we have RTP or RTCP ports */
3355 if (destination == NULL && (do_rtp || do_rtcp))
3356 goto no_destination;
3358 /* try to construct the fakesrc to the RTP port of the server to open up any
3361 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
3364 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
3365 stream->udpsink[0] =
3366 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3368 if (stream->udpsink[0] == NULL)
3369 goto no_sink_element;
3371 /* don't join multicast group, we will have the source socket do that */
3372 /* no sync or async state changes needed */
3373 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
3374 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3376 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3378 if (stream->udpsrc[0]) {
3379 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
3380 * so that NAT firewalls will open a hole for us */
3381 g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
3382 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
3383 /* configure socket and make sure udpsink does not close it when shutting
3384 * down, it belongs to udpsrc after all. */
3385 g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
3386 "close-socket", FALSE, NULL);
3387 g_object_unref (socket);
3390 /* the source for the dummy packets to open up NAT */
3391 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
3392 if (stream->fakesrc == NULL)
3393 goto no_fakesrc_element;
3395 /* random data in 5 buffers, a size of 200 bytes should be fine */
3396 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
3397 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
3399 /* we don't want to consider this a sink */
3400 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
3402 /* keep everything locked */
3403 gst_element_set_locked_state (stream->udpsink[0], TRUE);
3404 gst_element_set_locked_state (stream->fakesrc, TRUE);
3406 gst_object_ref (stream->udpsink[0]);
3407 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
3408 gst_object_ref (stream->fakesrc);
3409 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
3411 gst_element_link_pads_full (stream->fakesrc, "src", stream->udpsink[0],
3412 "sink", GST_PAD_LINK_CHECK_NOTHING);
3415 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
3418 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
3419 stream->udpsink[1] =
3420 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3422 if (stream->udpsink[1] == NULL)
3423 goto no_sink_element;
3425 /* don't join multicast group, we will have the source socket do that */
3426 /* no sync or async state changes needed */
3427 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
3428 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3430 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3432 if (stream->udpsrc[1]) {
3433 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
3434 * because some servers check the port number of where it sends RTCP to identify
3435 * the RTCP packets it receives */
3436 g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
3437 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
3438 /* configure socket and make sure udpsink does not close it when shutting
3439 * down, it belongs to udpsrc after all. */
3440 g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
3441 "close-socket", FALSE, NULL);
3442 g_object_unref (socket);
3445 /* we don't want to consider this a sink */
3446 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
3448 /* we keep this playing always */
3449 gst_element_set_locked_state (stream->udpsink[1], TRUE);
3450 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
3452 gst_object_ref (stream->udpsink[1]);
3453 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
3455 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
3457 /* get session RTCP pad */
3458 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3459 pad = gst_element_get_request_pad (src->manager, name);
3464 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3465 gst_object_unref (pad);
3474 GST_DEBUG_OBJECT (src, "no destination address specified");
3479 GST_DEBUG_OBJECT (src, "no UDP sink element found");
3484 GST_DEBUG_OBJECT (src, "no fakesrc element found");
3489 /* sets up all elements needed for streaming over the specified transport.
3490 * Does not yet expose the element pads, this will be done when there is actuall
3491 * dataflow detected, which might never happen when UDP is blocked in a
3492 * firewall, for example.
3495 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
3496 GstRTSPTransport * transport)
3499 GstPad *outpad = NULL;
3500 GstPadTemplate *template;
3502 const gchar *media_type;
3505 src = stream->parent;
3507 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
3509 /* get the proper media type for this stream now */
3510 if (gst_rtsp_transport_get_media_type (transport, &media_type) < 0)
3511 goto unknown_transport;
3513 goto unknown_transport;
3515 /* configure the final media type */
3516 GST_DEBUG_OBJECT (src, "setting media type to %s", media_type);
3518 len = stream->ptmap->len;
3519 for (i = 0; i < len; i++) {
3521 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
3523 if (item->caps == NULL)
3526 s = gst_caps_get_structure (item->caps, 0);
3527 gst_structure_set_name (s, media_type);
3530 /* try to get and configure a manager, channelpad[0-1] will be configured with
3531 * the pads for the manager, or NULL when no manager is needed. */
3532 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
3535 switch (transport->lower_transport) {
3536 case GST_RTSP_LOWER_TRANS_TCP:
3537 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
3538 goto transport_failed;
3540 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3541 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
3542 goto transport_failed;
3543 /* fallthrough, the rest is the same for UDP and MCAST */
3544 case GST_RTSP_LOWER_TRANS_UDP:
3545 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
3546 goto transport_failed;
3547 /* configure udpsinks back to the server for RTCP messages and for the
3548 * dummy RTP messages to open NAT. */
3549 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
3550 goto transport_failed;
3553 goto unknown_transport;
3557 GST_DEBUG_OBJECT (src, "creating ghostpad");
3559 gst_pad_use_fixed_caps (outpad);
3561 /* create ghostpad, don't add just yet, this will be done when we activate
3563 name = g_strdup_printf ("stream_%u", stream->id);
3564 template = gst_static_pad_template_get (&rtptemplate);
3565 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
3566 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
3567 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
3568 gst_object_unref (template);
3571 gst_object_unref (outpad);
3573 /* mark pad as ok */
3574 stream->last_ret = GST_FLOW_OK;
3581 GST_DEBUG_OBJECT (src, "failed to configure transport");
3586 GST_DEBUG_OBJECT (src, "unknown transport");
3591 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3596 /* send a couple of dummy random packets on the receiver RTP port to the server,
3597 * this should make a firewall think we initiated the data transfer and
3598 * hopefully allow packets to go from the sender port to our RTP receiver port */
3600 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
3604 if (src->nat_method != GST_RTSP_NAT_DUMMY)
3607 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3608 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3610 if (stream->fakesrc && stream->udpsink[0]) {
3611 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
3612 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
3613 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
3614 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
3615 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
3621 /* Adds the source pads of all configured streams to the element.
3622 * This code is performed when we detected dataflow.
3624 * We detect dataflow from either the _loop function or with pad probes on the
3628 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
3632 GST_DEBUG_OBJECT (src, "activating streams");
3634 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3635 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3637 if (stream->udpsrc[0]) {
3638 /* remove timeout, we are streaming now and timeouts will be handled by
3639 * the session manager and jitter buffer */
3640 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
3642 if (stream->srcpad) {
3643 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
3644 gst_pad_set_active (stream->srcpad, TRUE);
3646 /* if we don't have a session manager, set the caps now. If we have a
3647 * session, we will get a notification of the pad and the caps. */
3648 if (!src->manager) {
3651 caps = stream_get_caps_for_pt (stream, stream->default_pt);
3652 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
3653 gst_pad_set_caps (stream->srcpad, caps);
3656 if (!stream->added) {
3657 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
3658 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
3659 stream->added = TRUE;
3664 /* unblock all pads */
3665 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3666 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3668 if (stream->blockid) {
3669 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
3670 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
3671 stream->blockid = 0;
3679 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
3680 gboolean reset_manager)
3683 guint64 start, stop;
3684 gdouble play_speed, play_scale;
3686 GST_DEBUG_OBJECT (src, "configuring stream caps");
3688 start = segment->position;
3689 stop = segment->duration;
3690 play_speed = segment->rate;
3691 play_scale = segment->applied_rate;
3693 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3694 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3697 len = stream->ptmap->len;
3698 for (j = 0; j < len; j++) {
3700 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, j);
3702 if (item->caps == NULL)
3705 caps = gst_caps_make_writable (item->caps);
3707 if (stream->timebase != -1)
3708 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
3709 (guint) stream->timebase, NULL);
3710 if (stream->seqbase != -1)
3711 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
3712 (guint) stream->seqbase, NULL);
3713 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
3715 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
3716 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
3717 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
3720 GST_DEBUG_OBJECT (src, "stream %p, pt %d, caps %" GST_PTR_FORMAT, stream,
3724 if (reset_manager && src->manager) {
3725 GST_DEBUG_OBJECT (src, "clear session");
3726 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
3730 static GstFlowReturn
3731 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
3736 /* store the value */
3737 stream->last_ret = ret;
3739 /* if it's success we can return the value right away */
3740 if (ret == GST_FLOW_OK)
3743 /* any other error that is not-linked can be returned right
3745 if (ret != GST_FLOW_NOT_LINKED)
3748 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
3749 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3750 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3752 ret = ostream->last_ret;
3753 /* some other return value (must be SUCCESS but we can return
3754 * other values as well) */
3755 if (ret != GST_FLOW_NOT_LINKED)
3758 /* if we get here, all other pads were unlinked and we return
3759 * NOT_LINKED then */
3765 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
3768 gboolean res = TRUE;
3770 /* only streams that have a connection to the outside world */
3771 if (stream->container || stream->disabled)
3774 if (stream->udpsrc[0]) {
3775 gst_event_ref (event);
3776 res = gst_element_send_event (stream->udpsrc[0], event);
3777 } else if (stream->channelpad[0]) {
3778 gst_event_ref (event);
3779 if (GST_PAD_IS_SRC (stream->channelpad[0]))
3780 res = gst_pad_push_event (stream->channelpad[0], event);
3782 res = gst_pad_send_event (stream->channelpad[0], event);
3785 if (stream->udpsrc[1]) {
3786 gst_event_ref (event);
3787 res &= gst_element_send_event (stream->udpsrc[1], event);
3788 } else if (stream->channelpad[1]) {
3789 gst_event_ref (event);
3790 if (GST_PAD_IS_SRC (stream->channelpad[1]))
3791 res &= gst_pad_push_event (stream->channelpad[1], event);
3793 res &= gst_pad_send_event (stream->channelpad[1], event);
3797 gst_event_unref (event);
3803 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
3806 gboolean res = TRUE;
3808 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3809 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3811 gst_event_ref (event);
3812 res &= gst_rtspsrc_stream_push_event (src, ostream, event);
3814 gst_event_unref (event);
3819 static GstRTSPResult
3820 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3825 if (info->connection == NULL) {
3826 if (info->url == NULL) {
3827 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
3828 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
3832 /* create connection */
3833 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
3834 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
3835 goto could_not_create;
3838 g_free (info->url_str);
3839 info->url_str = gst_rtsp_url_get_request_uri (info->url);
3841 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
3843 if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
3844 if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
3845 src->tls_validation_flags))
3846 GST_WARNING_OBJECT (src, "Unable to set TLS validation flags");
3848 if (src->tls_database)
3849 gst_rtsp_connection_set_tls_database (info->connection,
3853 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
3854 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
3856 if (src->proxy_host) {
3857 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
3859 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
3864 if (!info->connected) {
3867 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
3868 ("Connecting to %s", info->location));
3869 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
3871 gst_rtsp_connection_connect (info->connection,
3872 src->ptcp_timeout)) < 0)
3873 goto could_not_connect;
3875 info->connected = TRUE;
3882 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
3887 gchar *str = gst_rtsp_strresult (res);
3888 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
3894 gchar *str = gst_rtsp_strresult (res);
3895 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
3901 static GstRTSPResult
3902 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
3905 GST_RTSP_STATE_LOCK (src);
3906 if (info->connected) {
3907 GST_DEBUG_OBJECT (src, "closing connection...");
3908 gst_rtsp_connection_close (info->connection);
3909 info->connected = FALSE;
3911 if (free && info->connection) {
3912 /* free connection */
3913 GST_DEBUG_OBJECT (src, "freeing connection...");
3914 gst_rtsp_connection_free (info->connection);
3915 info->connection = NULL;
3917 GST_RTSP_STATE_UNLOCK (src);
3921 static GstRTSPResult
3922 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3927 GST_DEBUG_OBJECT (src, "reconnecting connection...");
3928 gst_rtsp_conninfo_close (src, info, FALSE);
3929 res = gst_rtsp_conninfo_connect (src, info, async);
3935 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
3939 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
3940 GST_RTSP_STATE_LOCK (src);
3941 if (src->conninfo.connection && src->conninfo.flushing != flush) {
3942 GST_DEBUG_OBJECT (src, "connection flush");
3943 gst_rtsp_connection_flush (src->conninfo.connection, flush);
3944 src->conninfo.flushing = flush;
3946 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3947 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3948 if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
3949 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
3950 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
3951 stream->conninfo.flushing = flush;
3954 GST_RTSP_STATE_UNLOCK (src);
3957 /* FIXME, handle server request, reply with OK, for now */
3958 static GstRTSPResult
3959 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
3960 GstRTSPMessage * request)
3962 GstRTSPMessage response = { 0 };
3965 GST_DEBUG_OBJECT (src, "got server request message");
3968 gst_rtsp_message_dump (request);
3970 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
3972 if (res == GST_RTSP_ENOTIMPL) {
3973 /* default implementation, send OK */
3974 GST_DEBUG_OBJECT (src, "prepare OK reply");
3976 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
3981 /* let app parse and reply */
3982 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST],
3983 0, request, &response);
3986 gst_rtsp_message_dump (&response);
3988 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
3992 gst_rtsp_message_unset (&response);
3993 } else if (res == GST_RTSP_EEOF)
4001 gst_rtsp_message_unset (&response);
4006 /* send server keep-alive */
4007 static GstRTSPResult
4008 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
4010 GstRTSPMessage request = { 0 };
4012 GstRTSPMethod method;
4013 const gchar *control;
4015 if (src->do_rtsp_keep_alive == FALSE) {
4016 GST_DEBUG_OBJECT (src, "do-rtsp-keep-alive is FALSE, not sending.");
4017 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4021 GST_DEBUG_OBJECT (src, "creating server keep-alive");
4023 /* find a method to use for keep-alive */
4024 if (src->methods & GST_RTSP_GET_PARAMETER)
4025 method = GST_RTSP_GET_PARAMETER;
4027 method = GST_RTSP_OPTIONS;
4029 control = get_aggregate_control (src);
4030 if (control == NULL)
4033 res = gst_rtsp_message_init_request (&request, method, control);
4038 gst_rtsp_message_dump (&request);
4041 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
4046 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4047 gst_rtsp_message_unset (&request);
4054 GST_WARNING_OBJECT (src, "no control url to send keepalive");
4059 gchar *str = gst_rtsp_strresult (res);
4061 gst_rtsp_message_unset (&request);
4062 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
4063 ("Could not send keep-alive. (%s)", str));
4069 static GstFlowReturn
4070 gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
4072 GstFlowReturn ret = GST_FLOW_OK;
4074 GstRTSPStream *stream;
4075 GstPad *outpad = NULL;
4082 channel = message->type_data.data.channel;
4084 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
4086 goto unknown_stream;
4088 if (channel == stream->channel[0]) {
4089 outpad = stream->channelpad[0];
4091 } else if (channel == stream->channel[1]) {
4092 outpad = stream->channelpad[1];
4098 /* take a look at the body to figure out what we have */
4099 gst_rtsp_message_get_body (message, &data, &size);
4101 goto invalid_length;
4103 /* channels are not correct on some servers, do extra check */
4104 if (data[1] >= 200 && data[1] <= 204) {
4105 /* hmm RTCP message switch to the RTCP pad of the same stream. */
4106 outpad = stream->channelpad[1];
4110 /* we have no clue what this is, just ignore then. */
4112 goto unknown_stream;
4114 /* take the message body for further processing */
4115 gst_rtsp_message_steal_body (message, &data, &size);
4117 /* strip the trailing \0 */
4120 buf = gst_buffer_new ();
4121 gst_buffer_append_memory (buf,
4122 gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
4124 /* don't need message anymore */
4125 gst_rtsp_message_unset (message);
4127 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
4130 if (src->need_activate) {
4136 guint group_id = gst_util_group_id_next ();
4138 /* generate an SHA256 sum of the URI */
4139 cs = g_checksum_new (G_CHECKSUM_SHA256);
4140 uri = src->conninfo.location;
4141 g_checksum_update (cs, (const guchar *) uri, strlen (uri));
4143 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4144 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4147 g_strdup_printf ("%s/%d", g_checksum_get_string (cs), ostream->id);
4148 event = gst_event_new_stream_start (stream_id);
4149 gst_event_set_group_id (event, group_id);
4152 gst_rtspsrc_stream_push_event (src, ostream, event);
4154 g_checksum_free (cs);
4156 gst_rtspsrc_activate_streams (src);
4157 src->need_activate = FALSE;
4159 if ((event = src->start_segment) != NULL) {
4160 src->start_segment = NULL;
4161 gst_rtspsrc_push_event (src, event);
4164 if (src->base_time == -1) {
4165 /* Take current running_time. This timestamp will be put on
4166 * the first buffer of each stream because we are a live source and so we
4167 * timestamp with the running_time. When we are dealing with TCP, we also
4168 * only timestamp the first buffer (using the DISCONT flag) because a server
4169 * typically bursts data, for which we don't want to compensate by speeding
4170 * up the media. The other timestamps will be interpollated from this one
4171 * using the RTP timestamps. */
4172 GST_OBJECT_LOCK (src);
4173 if (GST_ELEMENT_CLOCK (src)) {
4175 GstClockTime base_time;
4177 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
4178 base_time = GST_ELEMENT_CAST (src)->base_time;
4180 src->base_time = now - base_time;
4182 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
4183 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
4185 GST_OBJECT_UNLOCK (src);
4188 if (stream->discont && !is_rtcp) {
4189 /* mark first RTP buffer as discont */
4190 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
4191 stream->discont = FALSE;
4192 /* first buffer gets the timestamp, other buffers are not timestamped and
4193 * their presentation time will be interpollated from the rtp timestamps. */
4194 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
4195 GST_TIME_ARGS (src->base_time));
4197 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
4200 /* chain to the peer pad */
4201 if (GST_PAD_IS_SINK (outpad))
4202 ret = gst_pad_chain (outpad, buf);
4204 ret = gst_pad_push (outpad, buf);
4207 /* combine all stream flows for the data transport */
4208 ret = gst_rtspsrc_combine_flows (src, stream, ret);
4215 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
4216 gst_rtsp_message_unset (message);
4221 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4222 ("Short message received, ignoring."));
4223 gst_rtsp_message_unset (message);
4228 static GstFlowReturn
4229 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
4231 GstRTSPMessage message = { 0 };
4233 GstFlowReturn ret = GST_FLOW_OK;
4234 GTimeVal tv_timeout;
4237 /* get the next timeout interval */
4238 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4240 /* see if the timeout period expired */
4241 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
4242 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
4243 /* send keep-alive, only act on interrupt, a warning will be posted for
4245 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4247 /* get new timeout */
4248 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4251 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
4252 tv_timeout.tv_sec, tv_timeout.tv_usec);
4254 /* protect the connection with the connection lock so that we can see when
4255 * we are finished doing server communication */
4257 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4258 &message, src->ptcp_timeout);
4262 GST_DEBUG_OBJECT (src, "we received a server message");
4264 case GST_RTSP_EINTR:
4265 /* we got interrupted this means we need to stop */
4267 case GST_RTSP_ETIMEOUT:
4268 /* no reply, send keep alive */
4269 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4270 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4274 /* go EOS when the server closed the connection */
4280 switch (message.type) {
4281 case GST_RTSP_MESSAGE_REQUEST:
4282 /* server sends us a request message, handle it */
4284 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4286 if (res == GST_RTSP_EEOF)
4289 goto handle_request_failed;
4291 case GST_RTSP_MESSAGE_RESPONSE:
4292 /* we ignore response messages */
4293 GST_DEBUG_OBJECT (src, "ignoring response message");
4295 gst_rtsp_message_dump (&message);
4297 case GST_RTSP_MESSAGE_DATA:
4298 GST_DEBUG_OBJECT (src, "got data message");
4299 ret = gst_rtspsrc_handle_data (src, &message);
4300 if (ret != GST_FLOW_OK)
4301 goto handle_data_failed;
4304 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4309 g_assert_not_reached ();
4314 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4315 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4316 ("The server closed the connection."));
4317 src->conninfo.connected = FALSE;
4318 gst_rtsp_message_unset (&message);
4319 return GST_FLOW_EOS;
4323 gst_rtsp_message_unset (&message);
4324 GST_DEBUG_OBJECT (src, "got interrupted");
4325 return GST_FLOW_FLUSHING;
4329 gchar *str = gst_rtsp_strresult (res);
4331 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4332 ("Could not receive message. (%s)", str));
4335 gst_rtsp_message_unset (&message);
4336 return GST_FLOW_ERROR;
4338 handle_request_failed:
4340 gchar *str = gst_rtsp_strresult (res);
4342 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4343 ("Could not handle server message. (%s)", str));
4345 gst_rtsp_message_unset (&message);
4346 return GST_FLOW_ERROR;
4350 GST_DEBUG_OBJECT (src, "could no handle data message");
4355 static GstFlowReturn
4356 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
4359 GstRTSPMessage message = { 0 };
4363 GTimeVal tv_timeout;
4365 /* get the next timeout interval */
4366 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4368 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
4369 (gint) tv_timeout.tv_sec);
4371 gst_rtsp_message_unset (&message);
4373 /* we should continue reading the TCP socket because the server might
4374 * send us requests. When the session timeout expires, we need to send a
4375 * keep-alive request to keep the session open. */
4376 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4377 &message, &tv_timeout);
4381 GST_DEBUG_OBJECT (src, "we received a server message");
4383 case GST_RTSP_EINTR:
4384 /* we got interrupted, see what we have to do */
4386 case GST_RTSP_ETIMEOUT:
4387 /* send keep-alive, ignore the result, a warning will be posted. */
4388 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4389 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4393 /* server closed the connection. not very fatal for UDP, reconnect and
4394 * see what happens. */
4395 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4396 ("The server closed the connection."));
4397 if (src->udp_reconnect) {
4399 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
4406 GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
4408 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4409 ("Unhandled return value %d.", res));
4413 switch (message.type) {
4414 case GST_RTSP_MESSAGE_REQUEST:
4415 /* server sends us a request message, handle it */
4417 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4419 if (res == GST_RTSP_EEOF)
4422 goto handle_request_failed;
4424 case GST_RTSP_MESSAGE_RESPONSE:
4425 /* we ignore response and data messages */
4426 GST_DEBUG_OBJECT (src, "ignoring response message");
4428 gst_rtsp_message_dump (&message);
4429 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
4430 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
4431 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
4432 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
4433 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4440 case GST_RTSP_MESSAGE_DATA:
4441 /* we ignore response and data messages */
4442 GST_DEBUG_OBJECT (src, "ignoring data message");
4445 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4450 g_assert_not_reached ();
4452 /* we get here when the connection got interrupted */
4455 gst_rtsp_message_unset (&message);
4456 GST_DEBUG_OBJECT (src, "got interrupted");
4457 return GST_FLOW_FLUSHING;
4461 gchar *str = gst_rtsp_strresult (res);
4464 src->conninfo.connected = FALSE;
4465 if (res != GST_RTSP_EINTR) {
4466 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
4467 ("Could not connect to server. (%s)", str));
4469 ret = GST_FLOW_ERROR;
4471 ret = GST_FLOW_FLUSHING;
4477 gchar *str = gst_rtsp_strresult (res);
4479 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4480 ("Could not receive message. (%s)", str));
4482 return GST_FLOW_ERROR;
4484 handle_request_failed:
4486 gchar *str = gst_rtsp_strresult (res);
4489 gst_rtsp_message_unset (&message);
4490 if (res != GST_RTSP_EINTR) {
4491 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4492 ("Could not handle server message. (%s)", str));
4494 ret = GST_FLOW_ERROR;
4496 ret = GST_FLOW_FLUSHING;
4502 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4503 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4504 ("The server closed the connection."));
4505 src->conninfo.connected = FALSE;
4506 gst_rtsp_message_unset (&message);
4507 return GST_FLOW_EOS;
4511 static GstRTSPResult
4512 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
4514 GstRTSPResult res = GST_RTSP_OK;
4517 GST_DEBUG_OBJECT (src, "doing reconnect");
4519 GST_OBJECT_LOCK (src);
4520 /* only restart when the pads were not yet activated, else we were
4521 * streaming over UDP */
4522 restart = src->need_activate;
4523 GST_OBJECT_UNLOCK (src);
4525 /* no need to restart, we're done */
4529 /* we can try only TCP now */
4530 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
4532 /* close and cleanup our state */
4533 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
4536 /* see if we have TCP left to try. Also don't try TCP when we were configured
4538 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
4541 /* We post a warning message now to inform the user
4542 * that nothing happened. It's most likely a firewall thing. */
4543 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4544 ("Could not receive any UDP packets for %.4f seconds, maybe your "
4545 "firewall is blocking it. Retrying using a TCP connection.",
4546 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
4548 /* open new connection using tcp */
4549 if (gst_rtspsrc_open (src, async) < 0)
4552 /* start playback */
4553 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
4562 src->cur_protocols = 0;
4563 /* no transport possible, post an error and stop */
4564 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4565 ("Could not receive any UDP packets for %.4f seconds, maybe your "
4566 "firewall is blocking it. No other protocols to try.",
4567 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
4568 return GST_RTSP_ERROR;
4572 GST_DEBUG_OBJECT (src, "open failed");
4577 GST_DEBUG_OBJECT (src, "play failed");
4583 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
4587 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
4590 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
4593 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
4596 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
4604 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
4608 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
4611 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
4614 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
4617 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
4625 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
4629 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
4632 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
4635 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
4638 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
4646 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
4650 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
4653 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
4656 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
4659 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
4667 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
4669 if (ret == GST_RTSP_OK)
4670 gst_rtspsrc_loop_complete_cmd (src, cmd);
4671 else if (ret == GST_RTSP_EINTR)
4672 gst_rtspsrc_loop_cancel_cmd (src, cmd);
4674 gst_rtspsrc_loop_error_cmd (src, cmd);
4678 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
4681 gboolean flushed = FALSE;
4683 /* start new request */
4684 gst_rtspsrc_loop_start_cmd (src, cmd);
4686 GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
4688 GST_OBJECT_LOCK (src);
4689 old = src->pending_cmd;
4690 if (old == CMD_RECONNECT) {
4691 GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
4692 cmd = CMD_RECONNECT;
4694 if (old != CMD_WAIT) {
4695 src->pending_cmd = CMD_WAIT;
4696 GST_OBJECT_UNLOCK (src);
4697 /* cancel previous request */
4698 GST_DEBUG_OBJECT (src, "cancel previous request %d", old);
4699 gst_rtspsrc_loop_cancel_cmd (src, old);
4700 GST_OBJECT_LOCK (src);
4702 src->pending_cmd = cmd;
4703 /* interrupt if allowed */
4704 if (src->busy_cmd & mask) {
4705 GST_DEBUG_OBJECT (src, "connection flush busy %d", src->busy_cmd);
4706 gst_rtspsrc_connection_flush (src, TRUE);
4709 GST_DEBUG_OBJECT (src, "not interrupting busy cmd %d", src->busy_cmd);
4712 gst_task_start (src->task);
4713 GST_OBJECT_UNLOCK (src);
4719 gst_rtspsrc_loop (GstRTSPSrc * src)
4723 if (!src->conninfo.connection || !src->conninfo.connected)
4726 if (src->interleaved)
4727 ret = gst_rtspsrc_loop_interleaved (src);
4729 ret = gst_rtspsrc_loop_udp (src);
4731 if (ret != GST_FLOW_OK)
4739 GST_WARNING_OBJECT (src, "we are not connected");
4740 ret = GST_FLOW_FLUSHING;
4745 const gchar *reason = gst_flow_get_name (ret);
4747 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
4748 src->running = FALSE;
4749 if (ret == GST_FLOW_EOS) {
4750 /* perform EOS logic */
4751 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
4752 gst_element_post_message (GST_ELEMENT_CAST (src),
4753 gst_message_new_segment_done (GST_OBJECT_CAST (src),
4754 src->segment.format, src->segment.position));
4755 gst_rtspsrc_push_event (src,
4756 gst_event_new_segment_done (src->segment.format,
4757 src->segment.position));
4759 gst_rtspsrc_push_event (src, gst_event_new_eos ());
4761 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
4762 /* for fatal errors we post an error message, post the error before the
4763 * EOS so the app knows about the error first. */
4764 GST_ELEMENT_ERROR (src, STREAM, FAILED,
4765 ("Internal data flow error."),
4766 ("streaming task paused, reason %s (%d)", reason, ret));
4767 gst_rtspsrc_push_event (src, gst_event_new_eos ());
4769 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_LOOP);
4774 #ifndef GST_DISABLE_GST_DEBUG
4775 static const gchar *
4776 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
4780 while (method != 0) {
4797 static const gchar *
4798 gst_rtspsrc_skip_lws (const gchar * s)
4800 while (g_ascii_isspace (*s))
4805 static const gchar *
4806 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
4808 while (s > start && g_ascii_isspace (*(s - 1)))
4813 static const gchar *
4814 gst_rtspsrc_skip_commas (const gchar * s)
4816 /* The grammar allows for multiple commas */
4817 while (g_ascii_isspace (*s) || *s == ',')
4822 static const gchar *
4823 gst_rtspsrc_skip_item (const gchar * s)
4825 gboolean quoted = FALSE;
4826 const gchar *start = s;
4828 /* A list item ends at the last non-whitespace character
4829 * before a comma which is not inside a quoted-string. Or at
4830 * the end of the string.
4836 if (*s == '\\' && *(s + 1))
4845 return gst_rtspsrc_unskip_lws (s, start);
4849 gst_rtsp_decode_quoted_string (gchar * quoted_string)
4853 src = quoted_string + 1;
4854 dst = quoted_string;
4855 while (*src && *src != '"') {
4856 if (*src == '\\' && *(src + 1))
4863 /* Extract the authentication tokens that the server provided for each method
4864 * into an array of structures and give those to the connection object.
4867 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
4868 const gchar * header, gboolean * stale)
4870 GSList *list = NULL, *iter;
4872 gchar *item, *eq, *name_end, *value;
4874 g_return_if_fail (stale != NULL);
4876 gst_rtsp_connection_clear_auth_params (conn);
4879 /* Parse a header whose content is described by RFC2616 as
4880 * "#something", where "something" does not itself contain commas,
4881 * except as part of quoted-strings, into a list of allocated strings.
4883 header = gst_rtspsrc_skip_commas (header);
4885 end = gst_rtspsrc_skip_item (header);
4886 list = g_slist_prepend (list, g_strndup (header, end - header));
4887 header = gst_rtspsrc_skip_commas (end);
4892 list = g_slist_reverse (list);
4893 for (iter = list; iter; iter = iter->next) {
4896 eq = strchr (item, '=');
4898 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
4899 if (name_end == item) {
4900 /* That's no good... */
4907 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
4909 gst_rtsp_decode_quoted_string (value);
4913 if (item && (strcmp (item, "stale") == 0) &&
4914 value && (strcmp (value, "TRUE") == 0))
4916 gst_rtsp_connection_set_auth_param (conn, item, value);
4920 g_slist_free (list);
4923 /* Parse a WWW-Authenticate Response header and determine the
4924 * available authentication methods
4926 * This code should also cope with the fact that each WWW-Authenticate
4927 * header can contain multiple challenge methods + tokens
4929 * At the moment, for Basic auth, we just do a minimal check and don't
4930 * even parse out the realm */
4932 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
4933 GstRTSPConnection * conn, gboolean * stale)
4937 g_return_if_fail (hdr != NULL);
4938 g_return_if_fail (methods != NULL);
4939 g_return_if_fail (stale != NULL);
4941 /* Skip whitespace at the start of the string */
4942 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
4944 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
4945 *methods |= GST_RTSP_AUTH_BASIC;
4946 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
4947 *methods |= GST_RTSP_AUTH_DIGEST;
4948 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
4953 * gst_rtspsrc_setup_auth:
4954 * @src: the rtsp source
4956 * Configure a username and password and auth method on the
4957 * connection object based on a response we received from the
4960 * Currently, this requires that a username and password were supplied
4961 * in the uri. In the future, they may be requested on demand by sending
4962 * a message up the bus.
4964 * Returns: TRUE if authentication information could be set up correctly.
4967 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
4971 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
4972 GstRTSPAuthMethod method;
4973 GstRTSPResult auth_result;
4975 GstRTSPConnection *conn;
4977 gboolean stale = FALSE;
4979 conn = src->conninfo.connection;
4981 /* Identify the available auth methods and see if any are supported */
4982 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
4983 &hdr, 0) == GST_RTSP_OK) {
4984 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
4987 if (avail_methods == GST_RTSP_AUTH_NONE)
4988 goto no_auth_available;
4990 /* For digest auth, if the response indicates that the session
4991 * data are stale, we just update them in the connection object and
4992 * return TRUE to retry the request */
4994 src->tried_url_auth = FALSE;
4996 url = gst_rtsp_connection_get_url (conn);
4998 /* Do we have username and password available? */
4999 if (url != NULL && !src->tried_url_auth && url->user != NULL
5000 && url->passwd != NULL) {
5003 src->tried_url_auth = TRUE;
5004 GST_DEBUG_OBJECT (src,
5005 "Attempting authentication using credentials from the URL");
5007 user = src->user_id;
5008 pass = src->user_pw;
5009 GST_DEBUG_OBJECT (src,
5010 "Attempting authentication using credentials from the properties");
5013 /* FIXME: If the url didn't contain username and password or we tried them
5014 * already, request a username and passwd from the application via some kind
5015 * of credentials request message */
5017 /* If we don't have a username and passwd at this point, bail out. */
5018 if (user == NULL || pass == NULL)
5021 /* Try to configure for each available authentication method, strongest to
5023 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
5024 /* Check if this method is available on the server */
5025 if ((method & avail_methods) == 0)
5028 /* Pass the credentials to the connection to try on the next request */
5029 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
5030 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
5031 * ignore it and end up retrying later */
5032 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
5033 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
5034 gst_rtsp_auth_method_to_string (method));
5039 if (method == GST_RTSP_AUTH_NONE)
5040 goto no_auth_available;
5046 /* Output an error indicating that we couldn't connect because there were
5047 * no supported authentication protocols */
5048 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5049 ("No supported authentication protocol was found"));
5054 /* We don't fire an error message, we just return FALSE and let the
5055 * normal NOT_AUTHORIZED error be propagated */
5060 static GstRTSPResult
5061 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5062 GstRTSPMessage * request, GstRTSPMessage * response,
5063 GstRTSPStatusCode * code)
5066 GstRTSPStatusCode thecode;
5067 gchar *content_base = NULL;
5071 if (!src->short_header)
5072 gst_rtsp_ext_list_before_send (src->extensions, request);
5074 GST_DEBUG_OBJECT (src, "sending message");
5077 gst_rtsp_message_dump (request);
5079 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
5083 gst_rtsp_connection_reset_timeout (conn);
5086 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
5091 gst_rtsp_message_dump (response);
5093 switch (response->type) {
5094 case GST_RTSP_MESSAGE_REQUEST:
5095 res = gst_rtspsrc_handle_request (src, conn, response);
5096 if (res == GST_RTSP_EEOF)
5099 goto handle_request_failed;
5101 case GST_RTSP_MESSAGE_RESPONSE:
5102 /* ok, a response is good */
5103 GST_DEBUG_OBJECT (src, "received response message");
5105 case GST_RTSP_MESSAGE_DATA:
5106 /* get next response */
5107 GST_DEBUG_OBJECT (src, "handle data response message");
5108 gst_rtspsrc_handle_data (src, response);
5111 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
5116 thecode = response->type_data.response.code;
5118 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
5120 /* if the caller wanted the result code, we store it. */
5124 /* If the request didn't succeed, bail out before doing any more */
5125 if (thecode != GST_RTSP_STS_OK)
5128 /* store new content base if any */
5129 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
5132 g_free (src->content_base);
5133 src->content_base = g_strdup (content_base);
5135 gst_rtsp_ext_list_after_send (src->extensions, request, response);
5142 gchar *str = gst_rtsp_strresult (res);
5144 if (res != GST_RTSP_EINTR) {
5145 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5146 ("Could not send message. (%s)", str));
5148 GST_WARNING_OBJECT (src, "send interrupted");
5157 GST_WARNING_OBJECT (src, "server closed connection");
5158 if ((try == 0) && !src->interleaved && src->udp_reconnect) {
5160 /* if reconnect succeeds, try again */
5162 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
5166 /* only try once after reconnect, then fallthrough and error out */
5169 gchar *str = gst_rtsp_strresult (res);
5171 if (res != GST_RTSP_EINTR) {
5172 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5173 ("Could not receive message. (%s)", str));
5175 GST_WARNING_OBJECT (src, "receive interrupted");
5183 handle_request_failed:
5185 /* ERROR was posted */
5186 gst_rtsp_message_unset (response);
5191 GST_DEBUG_OBJECT (src, "we got an eof from the server");
5192 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5193 ("The server closed the connection."));
5194 gst_rtsp_message_unset (response);
5201 * @src: the rtsp source
5202 * @conn: the connection to send on
5203 * @request: must point to a valid request
5204 * @response: must point to an empty #GstRTSPMessage
5205 * @code: an optional code result
5207 * send @request and retrieve the response in @response. optionally @code can be
5208 * non-NULL in which case it will contain the status code of the response.
5210 * If This function returns #GST_RTSP_OK, @response will contain a valid response
5211 * message that should be cleaned with gst_rtsp_message_unset() after usage.
5213 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
5214 * @response message) if the response code was not 200 (OK).
5216 * If the attempt results in an authentication failure, then this will attempt
5217 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
5220 * Returns: #GST_RTSP_OK if the processing was successful.
5222 static GstRTSPResult
5223 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5224 GstRTSPMessage * request, GstRTSPMessage * response,
5225 GstRTSPStatusCode * code)
5227 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
5228 GstRTSPResult res = GST_RTSP_ERROR;
5231 GstRTSPMethod method = GST_RTSP_INVALID;
5237 /* make sure we don't loop forever */
5241 /* save method so we can disable it when the server complains */
5242 method = request->type_data.request.method;
5245 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
5249 case GST_RTSP_STS_UNAUTHORIZED:
5250 if (gst_rtspsrc_setup_auth (src, response)) {
5251 /* Try the request/response again after configuring the auth info
5259 } while (retry == TRUE);
5261 /* If the user requested the code, let them handle errors, otherwise
5262 * post an error below */
5265 else if (int_code != GST_RTSP_STS_OK)
5266 goto error_response;
5273 GST_DEBUG_OBJECT (src, "got error %d", res);
5278 res = GST_RTSP_ERROR;
5280 switch (response->type_data.response.code) {
5281 case GST_RTSP_STS_NOT_FOUND:
5282 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
5283 response->type_data.response.reason));
5285 case GST_RTSP_STS_MOVED_PERMANENTLY:
5286 case GST_RTSP_STS_MOVE_TEMPORARILY:
5288 gchar *new_location;
5289 GstRTSPLowerTrans transports;
5291 GST_DEBUG_OBJECT (src, "got redirection");
5292 /* if we don't have a Location Header, we must error */
5293 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
5294 &new_location, 0) < 0)
5297 /* When we receive a redirect result, we go back to the INIT state after
5298 * parsing the new URI. The caller should do the needed steps to issue
5299 * a new setup when it detects this state change. */
5300 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
5302 /* save current transports */
5303 if (src->conninfo.url)
5304 transports = src->conninfo.url->transports;
5306 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
5308 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
5310 /* set old transports */
5311 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
5312 src->conninfo.url->transports = transports;
5314 src->need_redirect = TRUE;
5315 src->state = GST_RTSP_STATE_INIT;
5319 case GST_RTSP_STS_NOT_ACCEPTABLE:
5320 case GST_RTSP_STS_NOT_IMPLEMENTED:
5321 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
5322 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
5323 gst_rtsp_method_as_text (method));
5324 src->methods &= ~method;
5328 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5329 ("Got error response: %d (%s).", response->type_data.response.code,
5330 response->type_data.response.reason));
5333 /* if we return ERROR we should unset the response ourselves */
5334 if (res == GST_RTSP_ERROR)
5335 gst_rtsp_message_unset (response);
5341 static GstRTSPResult
5342 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
5343 GstRTSPMessage * response, GstRTSPSrc * src)
5345 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
5350 /* parse the response and collect all the supported methods. We need this
5351 * information so that we don't try to send an unsupported request to the
5355 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
5357 GstRTSPHeaderField field;
5361 /* reset supported methods */
5364 /* Try Allow Header first */
5365 field = GST_RTSP_HDR_ALLOW;
5368 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5369 if (indx == 0 && !respoptions) {
5370 /* if no Allow header was found then try the Public header... */
5371 field = GST_RTSP_HDR_PUBLIC;
5372 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5377 src->methods |= gst_rtsp_options_from_text (respoptions);
5382 if (src->methods == 0) {
5383 /* neither Allow nor Public are required, assume the server supports
5384 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
5386 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
5387 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
5389 /* always assume PLAY, FIXME, extensions should be able to override
5391 src->methods |= GST_RTSP_PLAY;
5392 /* also assume it will support Range */
5393 src->seekable = TRUE;
5395 /* we need describe and setup */
5396 if (!(src->methods & GST_RTSP_DESCRIBE))
5398 if (!(src->methods & GST_RTSP_SETUP))
5406 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5407 ("Server does not support DESCRIBE."));
5412 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5413 ("Server does not support SETUP."));
5418 /* masks to be kept in sync with the hardcoded protocol order of preference
5420 static guint protocol_masks[] = {
5421 GST_RTSP_LOWER_TRANS_UDP,
5422 GST_RTSP_LOWER_TRANS_UDP_MCAST,
5423 GST_RTSP_LOWER_TRANS_TCP,
5427 static GstRTSPResult
5428 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
5429 GstRTSPLowerTrans protocols, GstRTSPProfile profile, gchar ** transports)
5433 gboolean add_udp_str;
5438 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
5443 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
5445 /* extension listed transports, use those */
5446 if (*transports != NULL)
5449 /* it's the default */
5450 add_udp_str = FALSE;
5452 /* the default RTSP transports */
5453 result = g_string_new ("RTP");
5456 case GST_RTSP_PROFILE_AVP:
5457 g_string_append (result, "/AVP");
5459 case GST_RTSP_PROFILE_SAVP:
5460 g_string_append (result, "/SAVP");
5462 case GST_RTSP_PROFILE_AVPF:
5463 g_string_append (result, "/AVPF");
5465 case GST_RTSP_PROFILE_SAVPF:
5466 g_string_append (result, "/SAVPF");
5472 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
5473 GST_DEBUG_OBJECT (src, "adding UDP unicast");
5475 g_string_append (result, "/UDP");
5476 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
5477 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
5478 GST_DEBUG_OBJECT (src, "adding UDP multicast");
5479 /* we don't have to allocate any UDP ports yet, if the selected transport
5480 * turns out to be multicast we can create them and join the multicast
5481 * group indicated in the transport reply */
5483 g_string_append (result, "/UDP");
5484 g_string_append (result, ";multicast");
5485 if (src->next_port_num != 0) {
5486 if (src->client_port_range.max > 0 &&
5487 src->next_port_num >= src->client_port_range.max)
5490 g_string_append_printf (result, ";client_port=%d-%d",
5491 src->next_port_num, src->next_port_num + 1);
5493 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
5494 GST_DEBUG_OBJECT (src, "adding TCP");
5496 g_string_append (result, "/TCP;unicast;interleaved=%%i1-%%i2");
5498 *transports = g_string_free (result, FALSE);
5500 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
5507 GST_ERROR ("extension gave error %d", res);
5512 GST_ERROR ("no more ports available");
5513 return GST_RTSP_ERROR;
5517 static GstRTSPResult
5518 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
5519 gint orig_rtpport, gint orig_rtcpport)
5522 gint nr_udp, nr_int;
5524 gint rtpport = 0, rtcpport = 0;
5527 src = stream->parent;
5529 /* find number of placeholders first */
5530 if (strstr (*transports, "%%i2"))
5532 else if (strstr (*transports, "%%i1"))
5537 if (strstr (*transports, "%%u2"))
5539 else if (strstr (*transports, "%%u1"))
5544 if (nr_udp == 0 && nr_int == 0)
5548 if (!orig_rtpport || !orig_rtcpport) {
5549 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
5552 rtpport = orig_rtpport;
5553 rtcpport = orig_rtcpport;
5557 str = g_string_new ("");
5559 while ((next = strstr (p, "%%"))) {
5560 g_string_append_len (str, p, next - p);
5561 if (next[2] == 'u') {
5563 g_string_append_printf (str, "%d", rtpport);
5564 else if (next[3] == '2')
5565 g_string_append_printf (str, "%d", rtcpport);
5567 if (next[2] == 'i') {
5569 g_string_append_printf (str, "%d", src->free_channel);
5570 else if (next[3] == '2')
5571 g_string_append_printf (str, "%d", src->free_channel + 1);
5576 /* append final part */
5577 g_string_append (str, p);
5579 g_free (*transports);
5580 *transports = g_string_free (str, FALSE);
5588 GST_ERROR ("failed to allocate udp ports");
5589 return GST_RTSP_ERROR;
5593 /* Perform the SETUP request for all the streams.
5595 * We ask the server for a specific transport, which initially includes all the
5596 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
5597 * two local UDP ports that we send to the server.
5599 * Once the server replied with a transport, we configure the other streams
5600 * with the same transport.
5602 * This function will also configure the stream for the selected transport,
5603 * which basically means creating the pipeline.
5605 static GstRTSPResult
5606 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
5609 GstRTSPResult res = GST_RTSP_ERROR;
5610 GstRTSPMessage request = { 0 };
5611 GstRTSPMessage response = { 0 };
5612 GstRTSPStream *stream = NULL;
5613 GstRTSPLowerTrans protocols;
5614 GstRTSPStatusCode code;
5615 gboolean unsupported_real = FALSE;
5616 gint rtpport, rtcpport;
5620 if (src->conninfo.connection) {
5621 url = gst_rtsp_connection_get_url (src->conninfo.connection);
5622 /* we initially allow all configured lower transports. based on the URL
5623 * transports and the replies from the server we narrow them down. */
5624 protocols = url->transports & src->cur_protocols;
5627 protocols = src->cur_protocols;
5633 /* reset some state */
5634 src->free_channel = 0;
5635 src->interleaved = FALSE;
5636 src->need_activate = FALSE;
5637 /* keep track of next port number, 0 is random */
5638 src->next_port_num = src->client_port_range.min;
5639 rtpport = rtcpport = 0;
5641 if (G_UNLIKELY (src->streams == NULL))
5644 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5645 GstRTSPConnection *conn;
5652 stream = (GstRTSPStream *) walk->data;
5653 caps = stream_get_caps_for_pt (stream, stream->default_pt);
5655 GST_DEBUG_OBJECT (src, "skipping stream %p, no caps", stream);
5659 /* see if we need to configure this stream */
5660 if (!gst_rtsp_ext_list_configure_stream (src->extensions, caps)) {
5661 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
5663 stream->disabled = TRUE;
5667 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_SELECT_STREAM], 0,
5668 stream->id, caps, &selected);
5670 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by signal", stream);
5671 stream->disabled = TRUE;
5674 stream->disabled = FALSE;
5676 /* merge/overwrite global caps */
5681 s = gst_caps_get_structure (caps, 0);
5683 num = gst_structure_n_fields (src->props);
5684 for (j = 0; j < num; j++) {
5688 name = gst_structure_nth_field_name (src->props, j);
5689 val = gst_structure_get_value (src->props, name);
5690 gst_structure_set_value (s, name, val);
5692 GST_DEBUG_OBJECT (src, "copied %s", name);
5696 /* skip setup if we have no URL for it */
5697 if (stream->conninfo.location == NULL) {
5698 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
5702 if (src->conninfo.connection == NULL) {
5703 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
5704 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
5707 conn = stream->conninfo.connection;
5709 conn = src->conninfo.connection;
5711 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
5712 stream->conninfo.location);
5714 /* if we have a multicast connection, only suggest multicast from now on */
5715 if (stream->is_multicast)
5716 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
5719 /* first selectable protocol */
5720 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5722 if (!protocol_masks[mask])
5726 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
5727 protocol_masks[mask]);
5728 /* create a string with first transport in line */
5730 res = gst_rtspsrc_create_transports_string (src,
5731 protocols & protocol_masks[mask], stream->profile, &transports);
5732 if (res < 0 || transports == NULL)
5733 goto setup_transport_failed;
5735 if (strlen (transports) == 0) {
5736 g_free (transports);
5737 GST_DEBUG_OBJECT (src, "no transports found");
5742 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
5744 /* replace placeholders with real values, this function will optionally
5745 * allocate UDP ports and other info needed to execute the setup request */
5746 res = gst_rtspsrc_prepare_transports (stream, &transports,
5747 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
5749 g_free (transports);
5750 goto setup_transport_failed;
5753 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
5755 /* create SETUP request */
5757 gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
5758 stream->conninfo.location);
5760 g_free (transports);
5761 goto create_request_failed;
5764 /* select transport */
5765 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
5767 /* if the user wants a non default RTP packet size we add the blocksize
5769 if (src->rtp_blocksize > 0) {
5770 hval = g_strdup_printf ("%d", src->rtp_blocksize);
5771 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
5775 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
5778 /* handle the code ourselves */
5779 if ((res = gst_rtspsrc_send (src, conn, &request, &response, &code) < 0))
5783 case GST_RTSP_STS_OK:
5785 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
5786 gst_rtsp_message_unset (&request);
5787 gst_rtsp_message_unset (&response);
5788 /* cleanup of leftover transport */
5789 gst_rtspsrc_stream_free_udp (stream);
5790 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
5791 * we might be in this case */
5792 if (stream->container && rtpport && rtcpport && !retry) {
5793 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
5798 /* this transport did not go down well, but we may have others to try
5799 * that we did not send yet, try those and only give up then
5800 * but not without checking for lost cause/extension so we can
5801 * post a nicer/more useful error message later */
5802 if (!unsupported_real)
5803 unsupported_real = stream->is_real;
5804 /* select next available protocol, give up on this stream if none */
5806 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5808 if (!protocol_masks[mask] || unsupported_real)
5813 /* cleanup of leftover transport and move to the next stream */
5814 gst_rtspsrc_stream_free_udp (stream);
5815 goto response_error;
5818 /* parse response transport */
5820 gchar *resptrans = NULL;
5821 GstRTSPTransport transport = { 0 };
5823 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
5826 gst_rtspsrc_stream_free_udp (stream);
5830 /* parse transport, go to next stream on parse error */
5831 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
5832 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
5836 /* update allowed transports for other streams. once the transport of
5837 * one stream has been determined, we make sure that all other streams
5838 * are configured in the same way */
5839 switch (transport.lower_transport) {
5840 case GST_RTSP_LOWER_TRANS_TCP:
5841 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
5842 protocols = GST_RTSP_LOWER_TRANS_TCP;
5843 src->interleaved = TRUE;
5844 /* update free channels */
5846 MAX (transport.interleaved.min, src->free_channel);
5848 MAX (transport.interleaved.max, src->free_channel);
5849 src->free_channel++;
5851 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
5852 /* only allow multicast for other streams */
5853 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
5854 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
5855 /* if the server selected our ports, increment our counters so that
5856 * we select a new port later */
5857 if (src->next_port_num == transport.port.min &&
5858 src->next_port_num + 1 == transport.port.max) {
5859 src->next_port_num += 2;
5862 case GST_RTSP_LOWER_TRANS_UDP:
5863 /* only allow unicast for other streams */
5864 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
5865 protocols = GST_RTSP_LOWER_TRANS_UDP;
5868 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
5869 transport.lower_transport);
5873 if (!stream->container || (!src->interleaved && !retry)) {
5874 /* now configure the stream with the selected transport */
5875 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
5876 GST_DEBUG_OBJECT (src,
5877 "could not configure stream %p transport, skipping stream",
5880 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
5881 /* retain the first allocated UDP port pair */
5882 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
5883 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
5886 /* we need to activate at least one streams when we detect activity */
5887 src->need_activate = TRUE;
5889 /* clean up our transport struct */
5890 gst_rtsp_transport_init (&transport);
5891 /* clean up used RTSP messages */
5892 gst_rtsp_message_unset (&request);
5893 gst_rtsp_message_unset (&response);
5897 /* store the transport protocol that was configured */
5898 src->cur_protocols = protocols;
5900 gst_rtsp_ext_list_stream_select (src->extensions, url);
5902 /* if there is nothing to activate, error out */
5903 if (!src->need_activate)
5904 goto nothing_to_activate;
5911 /* no transport possible, post an error and stop */
5912 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5913 ("Could not connect to server, no protocols left"));
5914 return GST_RTSP_ERROR;
5918 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5919 ("SDP contains no streams"));
5920 return GST_RTSP_ERROR;
5922 create_request_failed:
5924 gchar *str = gst_rtsp_strresult (res);
5926 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5927 ("Could not create request. (%s)", str));
5931 setup_transport_failed:
5933 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5934 ("Could not setup transport."));
5935 res = GST_RTSP_ERROR;
5940 const gchar *str = gst_rtsp_status_as_text (code);
5942 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5943 ("Error (%d): %s", code, GST_STR_NULL (str)));
5944 res = GST_RTSP_ERROR;
5949 gchar *str = gst_rtsp_strresult (res);
5951 if (res != GST_RTSP_EINTR) {
5952 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5953 ("Could not send message. (%s)", str));
5955 GST_WARNING_OBJECT (src, "send interrupted");
5962 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5963 ("Server did not select transport."));
5964 res = GST_RTSP_ERROR;
5967 nothing_to_activate:
5969 /* none of the available error codes is really right .. */
5970 if (unsupported_real) {
5971 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
5972 (_("No supported stream was found. You might need to install a "
5973 "GStreamer RTSP extension plugin for Real media streams.")),
5976 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
5977 (_("No supported stream was found. You might need to allow "
5978 "more transport protocols or may otherwise be missing "
5979 "the right GStreamer RTSP extension plugin.")), (NULL));
5981 return GST_RTSP_ERROR;
5985 gst_rtsp_message_unset (&request);
5986 gst_rtsp_message_unset (&response);
5992 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
5993 GstSegment * segment)
5996 GstRTSPTimeRange *therange;
5999 gst_rtsp_range_free (src->range);
6001 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
6002 GST_DEBUG_OBJECT (src, "parsed range %s", range);
6003 src->range = therange;
6005 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
6007 gst_segment_init (segment, GST_FORMAT_TIME);
6011 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
6012 therange->min.type, therange->min.seconds, therange->max.type,
6013 therange->max.seconds);
6015 if (therange->min.type == GST_RTSP_TIME_NOW)
6017 else if (therange->min.type == GST_RTSP_TIME_END)
6020 seconds = therange->min.seconds * GST_SECOND;
6022 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
6023 GST_TIME_ARGS (seconds));
6025 /* we need to start playback without clipping from the position reported by
6027 segment->start = seconds;
6028 segment->position = seconds;
6030 if (therange->max.type == GST_RTSP_TIME_NOW)
6032 else if (therange->max.type == GST_RTSP_TIME_END)
6035 seconds = therange->max.seconds * GST_SECOND;
6037 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
6038 GST_TIME_ARGS (seconds));
6040 /* live (WMS) server might send overflowed large max as its idea of infinity,
6041 * compensate to prevent problems later on */
6042 if (seconds != -1 && seconds < 0) {
6044 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
6047 /* live (WMS) might send min == max, which is not worth recording */
6048 if (segment->duration == -1 && seconds == segment->start)
6051 /* don't change duration with unknown value, we might have a valid value
6052 * there that we want to keep. */
6054 segment->duration = seconds;
6059 /* Parse clock profived by the server with following syntax:
6061 * "GstNetTimeProvider <wrapped-clock> <server-IP:port> <clock-time>"
6064 gst_rtspsrc_parse_gst_clock (GstRTSPSrc * src, const gchar * gstclock)
6066 gboolean res = FALSE;
6068 if (g_str_has_prefix (gstclock, "GstNetTimeProvider ")) {
6069 gchar **fields = NULL, **parts = NULL;
6070 gchar *remote_ip, *str;
6072 GstClockTime base_time;
6075 fields = g_strsplit (gstclock, " ", 0);
6077 /* wrapped clock, not very interesting for now */
6078 if (fields[1] == NULL)
6081 /* remote IP address and port */
6082 if ((str = fields[2]) == NULL)
6085 parts = g_strsplit (str, ":", 0);
6087 if ((remote_ip = parts[0]) == NULL)
6090 if ((str = parts[1]) == NULL)
6098 if ((str = fields[3]) == NULL)
6101 base_time = g_ascii_strtoull (str, NULL, 10);
6104 gst_net_client_clock_new ((gchar *) "GstRTSPClock", remote_ip, port,
6107 if (src->provided_clock)
6108 gst_object_unref (src->provided_clock);
6109 src->provided_clock = netclock;
6111 gst_element_post_message (GST_ELEMENT_CAST (src),
6112 gst_message_new_clock_provide (GST_OBJECT_CAST (src),
6113 src->provided_clock, TRUE));
6117 g_strfreev (fields);
6123 /* must be called with the RTSP state lock */
6124 static GstRTSPResult
6125 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
6131 /* prepare global stream caps properties */
6133 gst_structure_remove_all_fields (src->props);
6135 src->props = gst_structure_new_empty ("RTSPProperties");
6138 gst_sdp_message_dump (sdp);
6140 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
6142 /* let the app inspect and change the SDP */
6143 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_ON_SDP], 0, sdp);
6145 gst_segment_init (&src->segment, GST_FORMAT_TIME);
6147 /* parse range for duration reporting. */
6152 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
6156 /* keep track of the range and configure it in the segment */
6157 if (gst_rtspsrc_parse_range (src, range, &src->segment))
6161 /* parse clock information. This is GStreamer specific, a server can tell the
6162 * client what clock it is using and wrap that in a network clock. The
6163 * advantage of that is that we can slave to it. */
6165 const gchar *gstclock;
6168 gstclock = gst_sdp_message_get_attribute_val_n (sdp, "x-gst-clock", i);
6169 if (gstclock == NULL)
6172 /* parse the clock and expose it in the provide_clock method */
6173 if (gst_rtspsrc_parse_gst_clock (src, gstclock))
6177 /* try to find a global control attribute. Note that a '*' means that we should
6178 * do aggregate control with the current url (so we don't do anything and
6179 * leave the current connection as is) */
6181 const gchar *control;
6184 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
6185 if (control == NULL)
6188 /* only take fully qualified urls */
6189 if (g_str_has_prefix (control, "rtsp://"))
6193 g_free (src->conninfo.location);
6194 src->conninfo.location = g_strdup (control);
6195 /* make a connection for this, if there was a connection already, nothing
6197 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
6198 GST_ERROR_OBJECT (src, "could not connect");
6201 /* we need to keep the control url separate from the connection url because
6202 * the rules for constructing the media control url need it */
6203 g_free (src->control);
6204 src->control = g_strdup (control);
6207 /* create streams */
6208 n_streams = gst_sdp_message_medias_len (sdp);
6209 for (i = 0; i < n_streams; i++) {
6210 gst_rtspsrc_create_stream (src, sdp, i);
6213 src->state = GST_RTSP_STATE_INIT;
6216 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
6219 /* reset our state */
6220 src->need_range = TRUE;
6223 src->state = GST_RTSP_STATE_READY;
6230 GST_ERROR_OBJECT (src, "setup failed");
6231 gst_rtspsrc_cleanup (src);
6236 static GstRTSPResult
6237 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
6241 GstRTSPMessage request = { 0 };
6242 GstRTSPMessage response = { 0 };
6245 gchar *respcont = NULL;
6248 src->need_redirect = FALSE;
6250 /* can't continue without a valid url */
6251 if (G_UNLIKELY (src->conninfo.url == NULL)) {
6252 res = GST_RTSP_EINVAL;
6255 src->tried_url_auth = FALSE;
6257 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
6258 goto connect_failed;
6260 /* create OPTIONS */
6261 GST_DEBUG_OBJECT (src, "create options...");
6263 gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
6264 src->conninfo.url_str);
6266 goto create_request_failed;
6269 GST_DEBUG_OBJECT (src, "send options...");
6272 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
6275 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6280 if (!gst_rtspsrc_parse_methods (src, &response))
6283 /* create DESCRIBE */
6284 GST_DEBUG_OBJECT (src, "create describe...");
6286 gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
6287 src->conninfo.url_str);
6289 goto create_request_failed;
6291 /* we only accept SDP for now */
6292 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
6296 GST_DEBUG_OBJECT (src, "send describe...");
6299 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
6302 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6306 /* we only perform redirect for the describe, currently */
6307 if (src->need_redirect) {
6308 /* close connection, we don't have to send a TEARDOWN yet, ignore the
6310 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6312 gst_rtsp_message_unset (&request);
6313 gst_rtsp_message_unset (&response);
6319 /* it could be that the DESCRIBE method was not implemented */
6320 if (!src->methods & GST_RTSP_DESCRIBE)
6323 /* check if reply is SDP */
6324 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
6326 /* could not be set but since the request returned OK, we assume it
6327 * was SDP, else check it. */
6329 if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
6330 goto wrong_content_type;
6333 /* get message body and parse as SDP */
6334 gst_rtsp_message_get_body (&response, &data, &size);
6335 if (data == NULL || size == 0)
6338 GST_DEBUG_OBJECT (src, "parse SDP...");
6339 gst_sdp_message_new (sdp);
6340 gst_sdp_message_parse_buffer (data, size, *sdp);
6342 /* clean up any messages */
6343 gst_rtsp_message_unset (&request);
6344 gst_rtsp_message_unset (&response);
6351 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
6352 ("No valid RTSP URL was provided"));
6357 gchar *str = gst_rtsp_strresult (res);
6359 if (res != GST_RTSP_EINTR) {
6360 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
6361 ("Failed to connect. (%s)", str));
6363 GST_WARNING_OBJECT (src, "connect interrupted");
6368 create_request_failed:
6370 gchar *str = gst_rtsp_strresult (res);
6372 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6373 ("Could not create request. (%s)", str));
6379 /* Don't post a message - the rtsp_send method will have
6380 * taken care of it because we passed NULL for the response code */
6385 /* error was posted */
6386 res = GST_RTSP_ERROR;
6391 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6392 ("Server does not support SDP, got %s.", respcont));
6393 res = GST_RTSP_ERROR;
6398 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6399 ("Server can not provide an SDP."));
6400 res = GST_RTSP_ERROR;
6405 if (src->conninfo.connection) {
6406 GST_DEBUG_OBJECT (src, "free connection");
6407 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6409 gst_rtsp_message_unset (&request);
6410 gst_rtsp_message_unset (&response);
6415 static GstRTSPResult
6416 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
6421 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
6423 if (src->sdp == NULL) {
6424 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
6428 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
6433 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
6440 GST_WARNING_OBJECT (src, "can't get sdp");
6441 src->open_error = TRUE;
6446 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
6447 src->open_error = TRUE;
6452 static GstRTSPResult
6453 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
6455 GstRTSPMessage request = { 0 };
6456 GstRTSPMessage response = { 0 };
6457 GstRTSPResult res = GST_RTSP_OK;
6459 const gchar *control;
6461 GST_DEBUG_OBJECT (src, "TEARDOWN...");
6463 gst_rtspsrc_set_state (src, GST_STATE_READY);
6465 if (src->state < GST_RTSP_STATE_READY) {
6466 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
6473 /* construct a control url */
6474 control = get_aggregate_control (src);
6476 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
6479 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6480 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6481 const gchar *setup_url;
6482 GstRTSPConnInfo *info;
6484 /* try aggregate control first but do non-aggregate control otherwise */
6486 setup_url = control;
6487 else if ((setup_url = stream->conninfo.location) == NULL)
6490 if (src->conninfo.connection) {
6491 info = &src->conninfo;
6492 } else if (stream->conninfo.connection) {
6493 info = &stream->conninfo;
6497 if (!info->connected)
6502 gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
6504 goto create_request_failed;
6507 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
6510 gst_rtspsrc_send (src, info->connection, &request, &response,
6514 /* FIXME, parse result? */
6515 gst_rtsp_message_unset (&request);
6516 gst_rtsp_message_unset (&response);
6519 /* early exit when we did aggregate control */
6525 /* close connections */
6526 GST_DEBUG_OBJECT (src, "closing connection...");
6527 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6528 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6529 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6530 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
6534 gst_rtspsrc_cleanup (src);
6536 src->state = GST_RTSP_STATE_INVALID;
6539 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
6544 create_request_failed:
6546 gchar *str = gst_rtsp_strresult (res);
6548 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6549 ("Could not create request. (%s)", str));
6555 gchar *str = gst_rtsp_strresult (res);
6557 gst_rtsp_message_unset (&request);
6558 if (res != GST_RTSP_EINTR) {
6559 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6560 ("Could not send message. (%s)", str));
6562 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
6569 GST_DEBUG_OBJECT (src,
6570 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
6575 /* RTP-Info is of the format:
6577 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
6579 * rtptime corresponds to the timestamp for the NPT time given in the header
6580 * seqbase corresponds to the next sequence number we received. This number
6581 * indicates the first seqnum after the seek and should be used to discard
6582 * packets that are from before the seek.
6585 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
6590 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
6592 infos = g_strsplit (rtpinfo, ",", 0);
6593 for (i = 0; infos[i]; i++) {
6595 GstRTSPStream *stream;
6599 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
6601 /* init values, types of seqbase and timebase are bigger than needed so we
6602 * can store -1 as uninitialized values */
6607 /* parse url, find stream for url.
6608 * parse seq and rtptime. The seq number should be configured in the rtp
6609 * depayloader or session manager to detect gaps. Same for the rtptime, it
6610 * should be used to create an initial time newsegment. */
6611 fields = g_strsplit (infos[i], ";", 0);
6612 for (j = 0; fields[j]; j++) {
6613 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
6614 /* remove leading whitespace */
6615 fields[j] = g_strchug (fields[j]);
6616 if (g_str_has_prefix (fields[j], "url=")) {
6617 /* get the url and the stream */
6619 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
6620 } else if (g_str_has_prefix (fields[j], "seq=")) {
6621 seqbase = atoi (fields[j] + 4);
6622 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
6623 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
6626 g_strfreev (fields);
6627 /* now we need to store the values for the caps of the stream */
6628 if (stream != NULL) {
6629 GST_DEBUG_OBJECT (src,
6630 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
6631 stream, seqbase, timebase);
6633 /* we have a stream, configure detected params */
6634 stream->seqbase = seqbase;
6635 stream->timebase = timebase;
6644 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
6649 interval = strtoul (rtcp, NULL, 10);
6650 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
6655 interval *= GST_MSECOND;
6657 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6658 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6660 /* already (optionally) retrieved this when configuring manager */
6661 if (stream->session) {
6662 GObject *rtpsession = stream->session;
6664 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
6666 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
6670 /* now it happens that (Xenon) server sending this may also provide bogus
6671 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
6672 * and just use RTP-Info to sync */
6674 GObjectClass *klass;
6676 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
6677 if (g_object_class_find_property (klass, "rtcp-sync")) {
6678 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
6679 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
6685 gst_rtspsrc_get_float (const gchar * dstr)
6687 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
6689 /* canonicalise floating point string so we can handle float strings
6690 * in the form "24.930" or "24,930" irrespective of the current locale */
6691 g_strlcpy (s, dstr, sizeof (s));
6692 g_strdelimit (s, ",", '.');
6693 return g_ascii_strtod (s, NULL);
6697 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
6699 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
6701 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
6702 g_strlcpy (val_str, "now", sizeof (val_str));
6704 if (segment->position == 0) {
6705 g_strlcpy (val_str, "0", sizeof (val_str));
6707 g_ascii_dtostr (val_str, sizeof (val_str),
6708 ((gdouble) segment->position) / GST_SECOND);
6711 return g_strdup_printf ("npt=%s-", val_str);
6715 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
6719 stream->timebase = -1;
6720 stream->seqbase = -1;
6722 len = stream->ptmap->len;
6723 for (i = 0; i < len; i++) {
6724 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
6727 if (item->caps == NULL)
6730 item->caps = gst_caps_make_writable (item->caps);
6731 s = gst_caps_get_structure (item->caps, 0);
6732 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
6736 static GstRTSPResult
6737 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
6739 GstRTSPResult res = GST_RTSP_OK;
6741 if (src->state < GST_RTSP_STATE_READY) {
6742 res = GST_RTSP_ERROR;
6743 if (src->open_error) {
6744 GST_DEBUG_OBJECT (src, "the stream was in error");
6748 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
6750 if ((res = gst_rtspsrc_open (src, async)) < 0) {
6751 GST_DEBUG_OBJECT (src, "failed to open stream");
6760 static GstRTSPResult
6761 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
6763 GstRTSPMessage request = { 0 };
6764 GstRTSPMessage response = { 0 };
6765 GstRTSPResult res = GST_RTSP_OK;
6769 const gchar *control;
6771 GST_DEBUG_OBJECT (src, "PLAY...");
6773 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6776 if (!(src->methods & GST_RTSP_PLAY))
6779 if (src->state == GST_RTSP_STATE_PLAYING)
6782 if (!src->conninfo.connection || !src->conninfo.connected)
6785 /* send some dummy packets before we activate the receive in the
6787 gst_rtspsrc_send_dummy_packets (src);
6789 /* require new SR packets */
6791 g_signal_emit_by_name (src->manager, "reset-sync", NULL);
6793 gst_rtspsrc_set_state (src, GST_STATE_PLAYING);
6795 /* construct a control url */
6796 control = get_aggregate_control (src);
6798 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6799 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6800 const gchar *setup_url;
6801 GstRTSPConnection *conn;
6803 /* try aggregate control first but do non-aggregate control otherwise */
6805 setup_url = control;
6806 else if ((setup_url = stream->conninfo.location) == NULL)
6809 if (src->conninfo.connection) {
6810 conn = src->conninfo.connection;
6811 } else if (stream->conninfo.connection) {
6812 conn = stream->conninfo.connection;
6818 res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
6820 goto create_request_failed;
6822 if (src->need_range) {
6823 hval = gen_range_header (src, segment);
6825 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
6827 /* store the newsegment event so it can be sent from the streaming thread. */
6828 if (src->start_segment)
6829 gst_event_unref (src->start_segment);
6830 src->start_segment = gst_event_new_segment (&src->segment);
6833 if (segment->rate != 1.0) {
6834 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
6836 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
6838 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
6840 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
6844 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
6846 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
6849 /* seek may have silently failed as it is not supported */
6850 if (!(src->methods & GST_RTSP_PLAY)) {
6851 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
6852 /* obviously it is supported as we made it here */
6853 src->methods |= GST_RTSP_PLAY;
6854 src->seekable = FALSE;
6855 /* but there is nothing to parse in the response,
6856 * so convey we have no idea and not to expect anything particular */
6857 clear_rtp_base (src, stream);
6861 /* need to do for all streams */
6862 for (run = src->streams; run; run = g_list_next (run))
6863 clear_rtp_base (src, (GstRTSPStream *) run->data);
6865 /* NOTE the above also disables npt based eos detection */
6866 /* and below forces position to 0,
6867 * which is visible feedback we lost the plot */
6868 segment->start = segment->position = src->last_pos;
6871 gst_rtsp_message_unset (&request);
6873 /* parse RTP npt field. This is the current position in the stream (Normal
6874 * Play Time) and should be put in the NEWSEGMENT position field. */
6875 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
6877 gst_rtspsrc_parse_range (src, hval, segment);
6879 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
6880 segment->rate = 1.0;
6882 /* parse Speed header. This is the intended playback rate of the stream
6883 * and should be put in the NEWSEGMENT rate field. */
6884 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
6885 0) == GST_RTSP_OK) {
6886 segment->rate = gst_rtspsrc_get_float (hval);
6887 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
6888 &hval, 0) == GST_RTSP_OK) {
6889 segment->rate = gst_rtspsrc_get_float (hval);
6892 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
6893 * for the RTP packets. If this is not present, we assume all starts from 0...
6894 * This is info for the RTP session manager that we pass to it in caps. */
6896 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
6897 &hval, hval_idx++) == GST_RTSP_OK)
6898 gst_rtspsrc_parse_rtpinfo (src, hval);
6900 /* some servers indicate RTCP parameters in PLAY response,
6901 * rather than properly in SDP */
6902 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
6903 &hval, 0) == GST_RTSP_OK)
6904 gst_rtspsrc_handle_rtcp_interval (src, hval);
6906 gst_rtsp_message_unset (&response);
6908 /* early exit when we did aggregate control */
6912 /* configure the caps of the streams after we parsed all headers. Only reset
6913 * the manager object when we set a new Range header (we did a seek) */
6914 gst_rtspsrc_configure_caps (src, segment, src->need_range);
6916 /* set again when needed */
6917 src->need_range = FALSE;
6919 src->running = TRUE;
6920 src->base_time = -1;
6921 src->state = GST_RTSP_STATE_PLAYING;
6924 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
6925 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6926 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6927 stream->discont = TRUE;
6932 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
6939 GST_DEBUG_OBJECT (src, "failed to open stream");
6944 GST_DEBUG_OBJECT (src, "PLAY is not supported");
6949 GST_DEBUG_OBJECT (src, "we were already PLAYING");
6952 create_request_failed:
6954 gchar *str = gst_rtsp_strresult (res);
6956 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6957 ("Could not create request. (%s)", str));
6963 gchar *str = gst_rtsp_strresult (res);
6965 gst_rtsp_message_unset (&request);
6966 if (res != GST_RTSP_EINTR) {
6967 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6968 ("Could not send message. (%s)", str));
6970 GST_WARNING_OBJECT (src, "PLAY interrupted");
6977 static GstRTSPResult
6978 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
6980 GstRTSPResult res = GST_RTSP_OK;
6981 GstRTSPMessage request = { 0 };
6982 GstRTSPMessage response = { 0 };
6984 const gchar *control;
6986 GST_DEBUG_OBJECT (src, "PAUSE...");
6988 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6991 if (!(src->methods & GST_RTSP_PAUSE))
6994 if (src->state == GST_RTSP_STATE_READY)
6997 if (!src->conninfo.connection || !src->conninfo.connected)
7000 /* construct a control url */
7001 control = get_aggregate_control (src);
7003 /* loop over the streams. We might exit the loop early when we could do an
7004 * aggregate control */
7005 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7006 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7007 GstRTSPConnection *conn;
7008 const gchar *setup_url;
7010 /* try aggregate control first but do non-aggregate control otherwise */
7012 setup_url = control;
7013 else if ((setup_url = stream->conninfo.location) == NULL)
7016 if (src->conninfo.connection) {
7017 conn = src->conninfo.connection;
7018 } else if (stream->conninfo.connection) {
7019 conn = stream->conninfo.connection;
7025 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
7026 ("Sending PAUSE request"));
7029 gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
7031 goto create_request_failed;
7033 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7036 gst_rtsp_message_unset (&request);
7037 gst_rtsp_message_unset (&response);
7039 /* exit early when we did agregate control */
7044 /* change element states now */
7045 gst_rtspsrc_set_state (src, GST_STATE_PAUSED);
7048 src->state = GST_RTSP_STATE_READY;
7052 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
7059 GST_DEBUG_OBJECT (src, "failed to open stream");
7064 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
7069 GST_DEBUG_OBJECT (src, "we were already PAUSED");
7072 create_request_failed:
7074 gchar *str = gst_rtsp_strresult (res);
7076 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7077 ("Could not create request. (%s)", str));
7083 gchar *str = gst_rtsp_strresult (res);
7085 gst_rtsp_message_unset (&request);
7086 if (res != GST_RTSP_EINTR) {
7087 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7088 ("Could not send message. (%s)", str));
7090 GST_WARNING_OBJECT (src, "PAUSE interrupted");
7098 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
7100 GstRTSPSrc *rtspsrc;
7102 rtspsrc = GST_RTSPSRC (bin);
7104 switch (GST_MESSAGE_TYPE (message)) {
7105 case GST_MESSAGE_EOS:
7106 gst_message_unref (message);
7108 case GST_MESSAGE_ELEMENT:
7110 const GstStructure *s = gst_message_get_structure (message);
7112 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
7113 gboolean ignore_timeout;
7115 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
7117 GST_OBJECT_LOCK (rtspsrc);
7118 ignore_timeout = rtspsrc->ignore_timeout;
7119 rtspsrc->ignore_timeout = TRUE;
7120 GST_OBJECT_UNLOCK (rtspsrc);
7122 /* we only act on the first udp timeout message, others are irrelevant
7123 * and can be ignored. */
7124 if (!ignore_timeout)
7125 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
7127 gst_message_unref (message);
7130 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7133 case GST_MESSAGE_ERROR:
7136 GstRTSPStream *stream;
7139 udpsrc = GST_MESSAGE_SRC (message);
7141 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
7142 GST_ELEMENT_NAME (udpsrc));
7144 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
7148 /* we ignore the RTCP udpsrc */
7149 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
7152 /* if we get error messages from the udp sources, that's not a problem as
7153 * long as not all of them error out. We also don't really know what the
7154 * problem is, the message does not give enough detail... */
7155 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
7156 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
7157 if (ret != GST_FLOW_OK)
7161 gst_message_unref (message);
7165 /* fatal but not our message, forward */
7166 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7171 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7177 /* the thread where everything happens */
7179 gst_rtspsrc_thread (GstRTSPSrc * src)
7183 GST_OBJECT_LOCK (src);
7184 cmd = src->pending_cmd;
7185 if (cmd == CMD_RECONNECT || cmd == CMD_PLAY || cmd == CMD_PAUSE
7186 || cmd == CMD_LOOP || cmd == CMD_OPEN)
7187 src->pending_cmd = CMD_LOOP;
7189 src->pending_cmd = CMD_WAIT;
7190 GST_DEBUG_OBJECT (src, "got command %d", cmd);
7192 /* we got the message command, so ensure communication is possible again */
7193 gst_rtspsrc_connection_flush (src, FALSE);
7195 src->busy_cmd = cmd;
7196 GST_OBJECT_UNLOCK (src);
7200 gst_rtspsrc_open (src, TRUE);
7203 gst_rtspsrc_play (src, &src->segment, TRUE);
7206 gst_rtspsrc_pause (src, TRUE);
7209 gst_rtspsrc_close (src, TRUE, FALSE);
7212 gst_rtspsrc_loop (src);
7215 gst_rtspsrc_reconnect (src, FALSE);
7221 GST_OBJECT_LOCK (src);
7222 /* and go back to sleep */
7223 if (src->pending_cmd == CMD_WAIT) {
7225 gst_task_pause (src->task);
7228 src->busy_cmd = CMD_WAIT;
7229 GST_OBJECT_UNLOCK (src);
7233 gst_rtspsrc_start (GstRTSPSrc * src)
7235 GST_DEBUG_OBJECT (src, "starting");
7237 GST_OBJECT_LOCK (src);
7239 src->pending_cmd = CMD_WAIT;
7241 if (src->task == NULL) {
7242 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src, NULL);
7243 if (src->task == NULL)
7246 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
7248 GST_OBJECT_UNLOCK (src);
7255 GST_OBJECT_UNLOCK (src);
7256 GST_ERROR_OBJECT (src, "failed to create task");
7262 gst_rtspsrc_stop (GstRTSPSrc * src)
7266 GST_DEBUG_OBJECT (src, "stopping");
7268 /* also cancels pending task */
7269 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_ALL);
7271 GST_OBJECT_LOCK (src);
7272 if ((task = src->task)) {
7274 GST_OBJECT_UNLOCK (src);
7276 gst_task_stop (task);
7278 /* make sure it is not running */
7279 GST_RTSP_STREAM_LOCK (src);
7280 GST_RTSP_STREAM_UNLOCK (src);
7282 /* now wait for the task to finish */
7283 gst_task_join (task);
7285 /* and free the task */
7286 gst_object_unref (GST_OBJECT (task));
7288 GST_OBJECT_LOCK (src);
7290 GST_OBJECT_UNLOCK (src);
7292 /* ensure synchronously all is closed and clean */
7293 gst_rtspsrc_close (src, FALSE, TRUE);
7298 static GstStateChangeReturn
7299 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
7301 GstRTSPSrc *rtspsrc;
7302 GstStateChangeReturn ret;
7304 rtspsrc = GST_RTSPSRC (element);
7306 switch (transition) {
7307 case GST_STATE_CHANGE_NULL_TO_READY:
7308 if (!gst_rtspsrc_start (rtspsrc))
7311 case GST_STATE_CHANGE_READY_TO_PAUSED:
7312 /* init some state */
7313 rtspsrc->cur_protocols = rtspsrc->protocols;
7314 /* first attempt, don't ignore timeouts */
7315 rtspsrc->ignore_timeout = FALSE;
7316 rtspsrc->open_error = FALSE;
7317 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
7319 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
7320 set_manager_buffer_mode (rtspsrc);
7322 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
7323 /* unblock the tcp tasks and make the loop waiting */
7324 if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
7325 /* make sure it is waiting before we send PAUSE or PLAY below */
7326 GST_RTSP_STREAM_LOCK (rtspsrc);
7327 GST_RTSP_STREAM_UNLOCK (rtspsrc);
7330 case GST_STATE_CHANGE_PAUSED_TO_READY:
7336 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
7337 if (ret == GST_STATE_CHANGE_FAILURE)
7340 switch (transition) {
7341 case GST_STATE_CHANGE_NULL_TO_READY:
7342 ret = GST_STATE_CHANGE_SUCCESS;
7344 case GST_STATE_CHANGE_READY_TO_PAUSED:
7345 ret = GST_STATE_CHANGE_NO_PREROLL;
7347 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
7348 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
7349 ret = GST_STATE_CHANGE_SUCCESS;
7351 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
7352 /* send pause request and keep the idle task around */
7353 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
7354 ret = GST_STATE_CHANGE_NO_PREROLL;
7356 case GST_STATE_CHANGE_PAUSED_TO_READY:
7357 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
7358 ret = GST_STATE_CHANGE_SUCCESS;
7360 case GST_STATE_CHANGE_READY_TO_NULL:
7361 gst_rtspsrc_stop (rtspsrc);
7362 ret = GST_STATE_CHANGE_SUCCESS;
7373 GST_DEBUG_OBJECT (rtspsrc, "start failed");
7374 return GST_STATE_CHANGE_FAILURE;
7379 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
7382 GstRTSPSrc *rtspsrc;
7384 rtspsrc = GST_RTSPSRC (element);
7386 if (GST_EVENT_IS_DOWNSTREAM (event)) {
7387 res = gst_rtspsrc_push_event (rtspsrc, event);
7389 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
7396 /*** GSTURIHANDLER INTERFACE *************************************************/
7399 gst_rtspsrc_uri_get_type (GType type)
7404 static const gchar *const *
7405 gst_rtspsrc_uri_get_protocols (GType type)
7407 static const gchar *protocols[] =
7408 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
7409 "rtsps", "rtspsu", "rtspst", "rtspsh", NULL
7416 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
7418 GstRTSPSrc *src = GST_RTSPSRC (handler);
7420 /* FIXME: make thread-safe */
7421 return g_strdup (src->conninfo.location);
7425 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
7430 GstRTSPUrl *newurl = NULL;
7431 GstSDPMessage *sdp = NULL;
7433 src = GST_RTSPSRC (handler);
7435 /* same URI, we're fine */
7436 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
7439 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
7440 if ((res = gst_sdp_message_new (&sdp) < 0))
7443 GST_DEBUG_OBJECT (src, "parsing SDP message");
7444 if ((res = gst_sdp_message_parse_uri (uri, sdp) < 0))
7448 GST_DEBUG_OBJECT (src, "parsing URI");
7449 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
7453 /* if worked, free previous and store new url object along with the original
7455 GST_DEBUG_OBJECT (src, "configuring URI");
7456 g_free (src->conninfo.location);
7457 src->conninfo.location = g_strdup (uri);
7458 gst_rtsp_url_free (src->conninfo.url);
7459 src->conninfo.url = newurl;
7460 g_free (src->conninfo.url_str);
7462 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
7464 src->conninfo.url_str = NULL;
7467 gst_sdp_message_free (src->sdp);
7469 src->from_sdp = sdp != NULL;
7471 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
7472 GST_DEBUG_OBJECT (src, "request uri is: %s",
7473 GST_STR_NULL (src->conninfo.url_str));
7480 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
7485 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", res);
7486 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7487 "Could not create SDP");
7492 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", res,
7493 GST_STR_NULL (uri));
7494 gst_sdp_message_free (sdp);
7495 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7501 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
7502 GST_STR_NULL (uri), res);
7503 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7504 "Invalid RTSP URI");
7510 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
7512 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
7514 iface->get_type = gst_rtspsrc_uri_get_type;
7515 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
7516 iface->get_uri = gst_rtspsrc_uri_get_uri;
7517 iface->set_uri = gst_rtspsrc_uri_set_uri;