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,
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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/sdp/gstmikey.h>
95 #include <gst/rtp/gstrtppayloads.h>
97 #include "gst/gst-i18n-plugin.h"
99 #include "gstrtspsrc.h"
101 GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
102 #define GST_CAT_DEFAULT (rtspsrc_debug)
104 static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
107 GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
109 /* templates used internally */
110 static GstStaticPadTemplate anysrctemplate =
111 GST_STATIC_PAD_TEMPLATE ("internalsrc_%u",
114 GST_STATIC_CAPS_ANY);
116 static GstStaticPadTemplate anysinktemplate =
117 GST_STATIC_PAD_TEMPLATE ("internalsink_%u",
120 GST_STATIC_CAPS_ANY);
124 SIGNAL_HANDLE_REQUEST,
126 SIGNAL_SELECT_STREAM,
131 enum _GstRtspSrcRtcpSyncMode
138 enum _GstRtspSrcBufferMode
147 #define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
149 gst_rtsp_src_buffer_mode_get_type (void)
151 static GType buffer_mode_type = 0;
152 static const GEnumValue buffer_modes[] = {
153 {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
154 {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
155 {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
156 {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
157 {BUFFER_MODE_SYNCED, "Synchronized sender and receiver clocks", "synced"},
161 if (!buffer_mode_type) {
163 g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
165 return buffer_mode_type;
168 #define DEFAULT_LOCATION NULL
169 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
170 #define DEFAULT_DEBUG FALSE
171 #define DEFAULT_RETRY 20
172 #define DEFAULT_TIMEOUT 5000000
173 #define DEFAULT_UDP_BUFFER_SIZE 0x80000
174 #define DEFAULT_TCP_TIMEOUT 20000000
175 #define DEFAULT_LATENCY_MS 2000
176 #define DEFAULT_DROP_ON_LATENCY FALSE
177 #define DEFAULT_CONNECTION_SPEED 0
178 #define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
179 #define DEFAULT_DO_RTCP TRUE
180 #define DEFAULT_DO_RTSP_KEEP_ALIVE TRUE
181 #define DEFAULT_PROXY NULL
182 #define DEFAULT_RTP_BLOCKSIZE 0
183 #define DEFAULT_USER_ID NULL
184 #define DEFAULT_USER_PW NULL
185 #define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO
186 #define DEFAULT_PORT_RANGE NULL
187 #define DEFAULT_SHORT_HEADER FALSE
188 #define DEFAULT_PROBATION 2
189 #define DEFAULT_UDP_RECONNECT TRUE
190 #define DEFAULT_MULTICAST_IFACE NULL
191 #define DEFAULT_NTP_SYNC FALSE
192 #define DEFAULT_USE_PIPELINE_CLOCK FALSE
193 #define DEFAULT_TLS_VALIDATION_FLAGS G_TLS_CERTIFICATE_VALIDATE_ALL
194 #define DEFAULT_TLS_DATABASE NULL
206 PROP_DROP_ON_LATENCY,
207 PROP_CONNECTION_SPEED,
210 PROP_DO_RTSP_KEEP_ALIVE,
219 PROP_UDP_BUFFER_SIZE,
223 PROP_MULTICAST_IFACE,
225 PROP_USE_PIPELINE_CLOCK,
227 PROP_TLS_VALIDATION_FLAGS,
232 #define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
234 gst_rtsp_nat_method_get_type (void)
236 static GType rtsp_nat_method_type = 0;
237 static const GEnumValue rtsp_nat_method[] = {
238 {GST_RTSP_NAT_NONE, "None", "none"},
239 {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
243 if (!rtsp_nat_method_type) {
244 rtsp_nat_method_type =
245 g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
247 return rtsp_nat_method_type;
250 static void gst_rtspsrc_finalize (GObject * object);
252 static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
253 const GValue * value, GParamSpec * pspec);
254 static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
255 GValue * value, GParamSpec * pspec);
257 static GstClock *gst_rtspsrc_provide_clock (GstElement * element);
259 static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
260 gpointer iface_data);
262 static void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
265 static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
266 static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
268 static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
270 static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
271 GstStateChange transition);
272 static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
273 static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
275 static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
276 GstRTSPMessage * response);
278 static gboolean gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd,
280 static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
281 GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
283 static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
284 static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
286 static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
287 static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
288 gboolean only_close);
290 static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
291 const gchar * uri, GError ** error);
292 static gchar *gst_rtspsrc_uri_get_uri (GstURIHandler * handler);
294 static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
295 static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
296 static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
297 GstRTSPStream * stream, GstEvent * event);
298 static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
299 static void gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush);
307 /* commands we send to out loop to notify it of events */
308 #define CMD_OPEN (1 << 0)
309 #define CMD_PLAY (1 << 1)
310 #define CMD_PAUSE (1 << 2)
311 #define CMD_CLOSE (1 << 3)
312 #define CMD_WAIT (1 << 4)
313 #define CMD_RECONNECT (1 << 5)
314 #define CMD_LOOP (1 << 6)
316 /* mask for all commands */
317 #define CMD_ALL ((CMD_LOOP << 1) - 1)
319 #define GST_ELEMENT_PROGRESS(el, type, code, text) \
321 gchar *__txt = _gst_element_error_printf text; \
322 gst_element_post_message (GST_ELEMENT_CAST (el), \
323 gst_message_new_progress (GST_OBJECT_CAST (el), \
324 GST_PROGRESS_TYPE_ ##type, code, __txt)); \
328 static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 };
330 #define gst_rtspsrc_parent_class parent_class
331 G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
332 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
335 default_select_stream (GstRTSPSrc * src, guint id, GstCaps * caps)
337 GST_DEBUG_OBJECT (src, "default handler");
342 select_stream_accum (GSignalInvocationHint * ihint,
343 GValue * return_accu, const GValue * handler_return, gpointer data)
347 myboolean = g_value_get_boolean (handler_return);
348 GST_DEBUG ("accum %d", myboolean);
349 g_value_set_boolean (return_accu, myboolean);
351 /* stop emission if FALSE */
356 gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
358 GObjectClass *gobject_class;
359 GstElementClass *gstelement_class;
360 GstBinClass *gstbin_class;
362 gobject_class = (GObjectClass *) klass;
363 gstelement_class = (GstElementClass *) klass;
364 gstbin_class = (GstBinClass *) klass;
366 GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
368 gobject_class->set_property = gst_rtspsrc_set_property;
369 gobject_class->get_property = gst_rtspsrc_get_property;
371 gobject_class->finalize = gst_rtspsrc_finalize;
373 g_object_class_install_property (gobject_class, PROP_LOCATION,
374 g_param_spec_string ("location", "RTSP Location",
375 "Location of the RTSP url to read",
376 DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
378 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
379 g_param_spec_flags ("protocols", "Protocols",
380 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
381 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
383 g_object_class_install_property (gobject_class, PROP_DEBUG,
384 g_param_spec_boolean ("debug", "Debug",
385 "Dump request and response messages to stdout",
386 DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
388 g_object_class_install_property (gobject_class, PROP_RETRY,
389 g_param_spec_uint ("retry", "Retry",
390 "Max number of retries when allocating RTP ports.",
391 0, G_MAXUINT16, DEFAULT_RETRY,
392 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
394 g_object_class_install_property (gobject_class, PROP_TIMEOUT,
395 g_param_spec_uint64 ("timeout", "Timeout",
396 "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
397 0, G_MAXUINT64, DEFAULT_TIMEOUT,
398 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
400 g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
401 g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
402 "Fail after timeout microseconds on TCP connections (0 = disabled)",
403 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
404 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
406 g_object_class_install_property (gobject_class, PROP_LATENCY,
407 g_param_spec_uint ("latency", "Buffer latency in ms",
408 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
409 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
411 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
412 g_param_spec_boolean ("drop-on-latency",
413 "Drop buffers when maximum latency is reached",
414 "Tells the jitterbuffer to never exceed the given latency in size",
415 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
417 g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
418 g_param_spec_uint64 ("connection-speed", "Connection Speed",
419 "Network connection speed in kbps (0 = unknown)",
420 0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
421 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
423 g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
424 g_param_spec_enum ("nat-method", "NAT Method",
425 "Method to use for traversing firewalls and NAT",
426 GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
427 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
430 * GstRTSPSrc:do-rtcp:
432 * Enable RTCP support. Some old server don't like RTCP and then this property
433 * needs to be set to FALSE.
435 g_object_class_install_property (gobject_class, PROP_DO_RTCP,
436 g_param_spec_boolean ("do-rtcp", "Do RTCP",
437 "Send RTCP packets, disable for old incompatible server.",
438 DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
441 * GstRTSPSrc:do-rtsp-keep-alive:
443 * Enable RTSP keep alive support. Some old server don't like RTSP
444 * keep alive and then this property needs to be set to FALSE.
446 g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
447 g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
448 "Send RTSP keep alive packets, disable for old incompatible server.",
449 DEFAULT_DO_RTSP_KEEP_ALIVE,
450 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
455 * Set the proxy parameters. This has to be a string of the format
456 * [http://][user:passwd@]host[:port].
458 g_object_class_install_property (gobject_class, PROP_PROXY,
459 g_param_spec_string ("proxy", "Proxy",
460 "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
461 DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
463 * GstRTSPSrc:proxy-id:
465 * Sets the proxy URI user id for authentication. If the URI set via the
466 * "proxy" property contains a user-id already, that will take precedence.
470 g_object_class_install_property (gobject_class, PROP_PROXY_ID,
471 g_param_spec_string ("proxy-id", "proxy-id",
472 "HTTP proxy URI user id for authentication", "",
473 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
475 * GstRTSPSrc:proxy-pw:
477 * Sets the proxy URI password for authentication. If the URI set via the
478 * "proxy" property contains a password already, that will take precedence.
482 g_object_class_install_property (gobject_class, PROP_PROXY_PW,
483 g_param_spec_string ("proxy-pw", "proxy-pw",
484 "HTTP proxy URI user password for authentication", "",
485 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
488 * GstRTSPSrc:rtp-blocksize:
490 * RTP package size to suggest to server.
492 g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
493 g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
494 "RTP package size to suggest to server (0 = disabled)",
495 0, 65536, DEFAULT_RTP_BLOCKSIZE,
496 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
498 g_object_class_install_property (gobject_class,
500 g_param_spec_string ("user-id", "user-id",
501 "RTSP location URI user id for authentication", DEFAULT_USER_ID,
502 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
503 g_object_class_install_property (gobject_class, PROP_USER_PW,
504 g_param_spec_string ("user-pw", "user-pw",
505 "RTSP location URI user password for authentication", DEFAULT_USER_PW,
506 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
509 * GstRTSPSrc:buffer-mode:
511 * Control the buffering and timestamping mode used by the jitterbuffer.
513 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
514 g_param_spec_enum ("buffer-mode", "Buffer Mode",
515 "Control the buffering algorithm in use",
516 GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
517 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
520 * GstRTSPSrc:port-range:
522 * Configure the client port numbers that can be used to recieve RTP and
525 g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
526 g_param_spec_string ("port-range", "Port range",
527 "Client port range that can be used to receive RTP and RTCP data, "
528 "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
529 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
532 * GstRTSPSrc:udp-buffer-size:
534 * Size of the kernel UDP receive buffer in bytes.
536 g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
537 g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
538 "Size of the kernel UDP receive buffer in bytes, 0=default",
539 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
540 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
543 * GstRTSPSrc:short-header:
545 * Only send the basic RTSP headers for broken encoders.
547 g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
548 g_param_spec_boolean ("short-header", "Short Header",
549 "Only send the basic RTSP headers for broken encoders",
550 DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
552 g_object_class_install_property (gobject_class, PROP_PROBATION,
553 g_param_spec_uint ("probation", "Number of probations",
554 "Consecutive packet sequence numbers to accept the source",
555 0, G_MAXUINT, DEFAULT_PROBATION,
556 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
558 g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
559 g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
560 "Reconnect to the server if RTSP connection is closed when doing UDP",
561 DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
563 g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
564 g_param_spec_string ("multicast-iface", "Multicast Interface",
565 "The network interface on which to join the multicast group",
566 DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
568 g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
569 g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
570 "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
571 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
573 g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
574 g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
575 "Use the pipeline running-time to set the NTP time in the RTCP SR messages",
576 DEFAULT_USE_PIPELINE_CLOCK,
577 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
579 g_object_class_install_property (gobject_class, PROP_SDES,
580 g_param_spec_boxed ("sdes", "SDES",
581 "The SDES items of this session",
582 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
585 * GstRTSPSrc::tls-validation-flags:
587 * TLS certificate validation flags used to validate server
592 g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
593 g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
594 "TLS certificate validation flags used to validate the server certificate",
595 G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS,
596 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
599 * GstRTSPSrc::tls-database:
601 * TLS database with anchor certificate authorities used to validate
602 * the server certificate.
606 g_object_class_install_property (gobject_class, PROP_TLS_DATABASE,
607 g_param_spec_object ("tls-database", "TLS database",
608 "TLS database with anchor certificate authorities used to validate the server certificate",
609 G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
612 * GstRTSPSrc::handle-request:
613 * @rtspsrc: a #GstRTSPSrc
614 * @request: a #GstRTSPMessage
615 * @response: a #GstRTSPMessage
617 * Handle a server request in @request and prepare @response.
619 * This signal is called from the streaming thread, you should therefore not
620 * do any state changes on @rtspsrc because this might deadlock. If you want
621 * to modify the state as a result of this signal, post a
622 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
627 gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST] =
628 g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
629 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2,
630 G_TYPE_POINTER, G_TYPE_POINTER);
633 * GstRTSPSrc::on-sdp:
634 * @rtspsrc: a #GstRTSPSrc
635 * @sdp: a #GstSDPMessage
637 * Emited when the client has retrieved the SDP and before it configures the
638 * streams in the SDP. @sdp can be inspected and modified.
640 * This signal is called from the streaming thread, you should therefore not
641 * do any state changes on @rtspsrc because this might deadlock. If you want
642 * to modify the state as a result of this signal, post a
643 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
648 gst_rtspsrc_signals[SIGNAL_ON_SDP] =
649 g_signal_new ("on-sdp", G_TYPE_FROM_CLASS (klass), 0,
650 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
651 GST_TYPE_SDP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
654 * GstRTSPSrc::select-stream:
655 * @rtspsrc: a #GstRTSPSrc
656 * @num: the stream number
657 * @caps: the stream caps
659 * Emited before the client decides to configure the stream @num with
662 * Returns: %TRUE when the stream should be selected, %FALSE when the stream
667 gst_rtspsrc_signals[SIGNAL_SELECT_STREAM] =
668 g_signal_new_class_handler ("select-stream", G_TYPE_FROM_CLASS (klass),
669 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP,
670 (GCallback) default_select_stream, select_stream_accum, NULL,
671 g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2, G_TYPE_UINT,
674 * GstRTSPSrc::new-manager:
675 * @rtspsrc: a #GstRTSPSrc
676 * @manager: a #GstElement
678 * Emited after a new manager (like rtpbin) was created and the default
679 * properties were configured.
683 gst_rtspsrc_signals[SIGNAL_NEW_MANAGER] =
684 g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass),
685 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
686 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
688 gstelement_class->send_event = gst_rtspsrc_send_event;
689 gstelement_class->provide_clock = gst_rtspsrc_provide_clock;
690 gstelement_class->change_state = gst_rtspsrc_change_state;
692 gst_element_class_add_pad_template (gstelement_class,
693 gst_static_pad_template_get (&rtptemplate));
695 gst_element_class_set_static_metadata (gstelement_class,
696 "RTSP packet receiver", "Source/Network",
697 "Receive data over the network via RTSP (RFC 2326)",
698 "Wim Taymans <wim@fluendo.com>, "
699 "Thijs Vermeir <thijs.vermeir@barco.com>, "
700 "Lutz Mueller <lutz@topfrose.de>");
702 gstbin_class->handle_message = gst_rtspsrc_handle_message;
704 gst_rtsp_ext_list_init ();
708 gst_rtspsrc_init (GstRTSPSrc * src)
710 src->conninfo.location = g_strdup (DEFAULT_LOCATION);
711 src->protocols = DEFAULT_PROTOCOLS;
712 src->debug = DEFAULT_DEBUG;
713 src->retry = DEFAULT_RETRY;
714 src->udp_timeout = DEFAULT_TIMEOUT;
715 gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
716 src->latency = DEFAULT_LATENCY_MS;
717 src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
718 src->connection_speed = DEFAULT_CONNECTION_SPEED;
719 src->nat_method = DEFAULT_NAT_METHOD;
720 src->do_rtcp = DEFAULT_DO_RTCP;
721 src->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
722 gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
723 src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
724 src->user_id = g_strdup (DEFAULT_USER_ID);
725 src->user_pw = g_strdup (DEFAULT_USER_PW);
726 src->buffer_mode = DEFAULT_BUFFER_MODE;
727 src->client_port_range.min = 0;
728 src->client_port_range.max = 0;
729 src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
730 src->short_header = DEFAULT_SHORT_HEADER;
731 src->probation = DEFAULT_PROBATION;
732 src->udp_reconnect = DEFAULT_UDP_RECONNECT;
733 src->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
734 src->ntp_sync = DEFAULT_NTP_SYNC;
735 src->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
737 src->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
738 src->tls_database = DEFAULT_TLS_DATABASE;
740 /* get a list of all extensions */
741 src->extensions = gst_rtsp_ext_list_get ();
743 /* connect to send signal */
744 gst_rtsp_ext_list_connect (src->extensions, "send",
745 (GCallback) gst_rtspsrc_send_cb, src);
747 /* protects the streaming thread in interleaved mode or the polling
748 * thread in UDP mode. */
749 g_rec_mutex_init (&src->stream_rec_lock);
751 /* protects our state changes from multiple invocations */
752 g_rec_mutex_init (&src->state_rec_lock);
754 src->state = GST_RTSP_STATE_INVALID;
756 GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
760 gst_rtspsrc_finalize (GObject * object)
764 rtspsrc = GST_RTSPSRC (object);
766 gst_rtsp_ext_list_free (rtspsrc->extensions);
767 g_free (rtspsrc->conninfo.location);
768 gst_rtsp_url_free (rtspsrc->conninfo.url);
769 g_free (rtspsrc->conninfo.url_str);
770 g_free (rtspsrc->user_id);
771 g_free (rtspsrc->user_pw);
772 g_free (rtspsrc->multi_iface);
775 gst_sdp_message_free (rtspsrc->sdp);
778 if (rtspsrc->provided_clock)
779 gst_object_unref (rtspsrc->provided_clock);
782 gst_structure_free (rtspsrc->sdes);
784 if (rtspsrc->tls_database)
785 g_object_unref (rtspsrc->tls_database);
788 g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
789 g_rec_mutex_clear (&rtspsrc->state_rec_lock);
791 G_OBJECT_CLASS (parent_class)->finalize (object);
795 gst_rtspsrc_provide_clock (GstElement * element)
797 GstRTSPSrc *src = GST_RTSPSRC (element);
800 if ((clock = src->provided_clock) != NULL)
801 gst_object_ref (clock);
806 /* a proxy string of the format [user:passwd@]host[:port] */
808 gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
812 g_free (rtsp->proxy_user);
813 rtsp->proxy_user = NULL;
814 g_free (rtsp->proxy_passwd);
815 rtsp->proxy_passwd = NULL;
816 g_free (rtsp->proxy_host);
817 rtsp->proxy_host = NULL;
818 rtsp->proxy_port = 0;
825 /* we allow http:// in front but ignore it */
826 if (g_str_has_prefix (p, "http://"))
829 at = strchr (p, '@');
831 /* look for user:passwd */
832 col = strchr (proxy, ':');
833 if (col == NULL || col > at)
836 rtsp->proxy_user = g_strndup (p, col - p);
838 rtsp->proxy_passwd = g_strndup (col, at - col);
843 if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
844 rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
845 if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
846 rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
847 if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
848 GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
849 GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
852 col = strchr (p, ':');
855 /* everything before the colon is the hostname */
856 rtsp->proxy_host = g_strndup (p, col - p);
858 rtsp->proxy_port = strtoul (p, (char **) &p, 10);
860 rtsp->proxy_host = g_strdup (p);
861 rtsp->proxy_port = 8080;
867 gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
869 rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
870 rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
873 rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
875 rtspsrc->ptcp_timeout = NULL;
879 gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
884 rtspsrc = GST_RTSPSRC (object);
888 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
889 g_value_get_string (value), NULL);
892 rtspsrc->protocols = g_value_get_flags (value);
895 rtspsrc->debug = g_value_get_boolean (value);
898 rtspsrc->retry = g_value_get_uint (value);
901 rtspsrc->udp_timeout = g_value_get_uint64 (value);
903 case PROP_TCP_TIMEOUT:
904 gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
907 rtspsrc->latency = g_value_get_uint (value);
909 case PROP_DROP_ON_LATENCY:
910 rtspsrc->drop_on_latency = g_value_get_boolean (value);
912 case PROP_CONNECTION_SPEED:
913 rtspsrc->connection_speed = g_value_get_uint64 (value);
915 case PROP_NAT_METHOD:
916 rtspsrc->nat_method = g_value_get_enum (value);
919 rtspsrc->do_rtcp = g_value_get_boolean (value);
921 case PROP_DO_RTSP_KEEP_ALIVE:
922 rtspsrc->do_rtsp_keep_alive = g_value_get_boolean (value);
925 gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
928 if (rtspsrc->prop_proxy_id)
929 g_free (rtspsrc->prop_proxy_id);
930 rtspsrc->prop_proxy_id = g_value_dup_string (value);
933 if (rtspsrc->prop_proxy_pw)
934 g_free (rtspsrc->prop_proxy_pw);
935 rtspsrc->prop_proxy_pw = g_value_dup_string (value);
937 case PROP_RTP_BLOCKSIZE:
938 rtspsrc->rtp_blocksize = g_value_get_uint (value);
941 if (rtspsrc->user_id)
942 g_free (rtspsrc->user_id);
943 rtspsrc->user_id = g_value_dup_string (value);
946 if (rtspsrc->user_pw)
947 g_free (rtspsrc->user_pw);
948 rtspsrc->user_pw = g_value_dup_string (value);
950 case PROP_BUFFER_MODE:
951 rtspsrc->buffer_mode = g_value_get_enum (value);
953 case PROP_PORT_RANGE:
957 str = g_value_get_string (value);
959 sscanf (str, "%u-%u",
960 &rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
962 rtspsrc->client_port_range.min = 0;
963 rtspsrc->client_port_range.max = 0;
967 case PROP_UDP_BUFFER_SIZE:
968 rtspsrc->udp_buffer_size = g_value_get_int (value);
970 case PROP_SHORT_HEADER:
971 rtspsrc->short_header = g_value_get_boolean (value);
974 rtspsrc->probation = g_value_get_uint (value);
976 case PROP_UDP_RECONNECT:
977 rtspsrc->udp_reconnect = g_value_get_boolean (value);
979 case PROP_MULTICAST_IFACE:
980 g_free (rtspsrc->multi_iface);
982 if (g_value_get_string (value) == NULL)
983 rtspsrc->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
985 rtspsrc->multi_iface = g_value_dup_string (value);
988 rtspsrc->ntp_sync = g_value_get_boolean (value);
990 case PROP_USE_PIPELINE_CLOCK:
991 rtspsrc->use_pipeline_clock = g_value_get_boolean (value);
994 rtspsrc->sdes = g_value_dup_boxed (value);
996 case PROP_TLS_VALIDATION_FLAGS:
997 rtspsrc->tls_validation_flags = g_value_get_flags (value);
999 case PROP_TLS_DATABASE:
1000 g_clear_object (&rtspsrc->tls_database);
1001 rtspsrc->tls_database = g_value_dup_object (value);
1004 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1010 gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
1013 GstRTSPSrc *rtspsrc;
1015 rtspsrc = GST_RTSPSRC (object);
1019 g_value_set_string (value, rtspsrc->conninfo.location);
1021 case PROP_PROTOCOLS:
1022 g_value_set_flags (value, rtspsrc->protocols);
1025 g_value_set_boolean (value, rtspsrc->debug);
1028 g_value_set_uint (value, rtspsrc->retry);
1031 g_value_set_uint64 (value, rtspsrc->udp_timeout);
1033 case PROP_TCP_TIMEOUT:
1037 timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
1038 rtspsrc->tcp_timeout.tv_usec;
1039 g_value_set_uint64 (value, timeout);
1043 g_value_set_uint (value, rtspsrc->latency);
1045 case PROP_DROP_ON_LATENCY:
1046 g_value_set_boolean (value, rtspsrc->drop_on_latency);
1048 case PROP_CONNECTION_SPEED:
1049 g_value_set_uint64 (value, rtspsrc->connection_speed);
1051 case PROP_NAT_METHOD:
1052 g_value_set_enum (value, rtspsrc->nat_method);
1055 g_value_set_boolean (value, rtspsrc->do_rtcp);
1057 case PROP_DO_RTSP_KEEP_ALIVE:
1058 g_value_set_boolean (value, rtspsrc->do_rtsp_keep_alive);
1064 if (rtspsrc->proxy_host) {
1066 g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
1070 g_value_take_string (value, str);
1074 g_value_set_string (value, rtspsrc->prop_proxy_id);
1077 g_value_set_string (value, rtspsrc->prop_proxy_pw);
1079 case PROP_RTP_BLOCKSIZE:
1080 g_value_set_uint (value, rtspsrc->rtp_blocksize);
1083 g_value_set_string (value, rtspsrc->user_id);
1086 g_value_set_string (value, rtspsrc->user_pw);
1088 case PROP_BUFFER_MODE:
1089 g_value_set_enum (value, rtspsrc->buffer_mode);
1091 case PROP_PORT_RANGE:
1095 if (rtspsrc->client_port_range.min != 0) {
1096 str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
1097 rtspsrc->client_port_range.max);
1101 g_value_take_string (value, str);
1104 case PROP_UDP_BUFFER_SIZE:
1105 g_value_set_int (value, rtspsrc->udp_buffer_size);
1107 case PROP_SHORT_HEADER:
1108 g_value_set_boolean (value, rtspsrc->short_header);
1110 case PROP_PROBATION:
1111 g_value_set_uint (value, rtspsrc->probation);
1113 case PROP_UDP_RECONNECT:
1114 g_value_set_boolean (value, rtspsrc->udp_reconnect);
1116 case PROP_MULTICAST_IFACE:
1117 g_value_set_string (value, rtspsrc->multi_iface);
1120 g_value_set_boolean (value, rtspsrc->ntp_sync);
1122 case PROP_USE_PIPELINE_CLOCK:
1123 g_value_set_boolean (value, rtspsrc->use_pipeline_clock);
1126 g_value_set_boxed (value, rtspsrc->sdes);
1128 case PROP_TLS_VALIDATION_FLAGS:
1129 g_value_set_flags (value, rtspsrc->tls_validation_flags);
1131 case PROP_TLS_DATABASE:
1132 g_value_set_object (value, rtspsrc->tls_database);
1135 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1141 find_stream_by_id (GstRTSPStream * stream, gint * id)
1143 if (stream->id == *id)
1150 find_stream_by_channel (GstRTSPStream * stream, gint * channel)
1152 if (stream->channel[0] == *channel || stream->channel[1] == *channel)
1159 find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
1161 GstElement *src = (GstElement *) a;
1163 if (stream->udpsrc[0] == src)
1165 if (stream->udpsrc[1] == src)
1172 find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
1174 if (stream->conninfo.location) {
1175 /* check qualified setup_url */
1176 if (!strcmp (stream->conninfo.location, (gchar *) a))
1179 if (stream->control_url) {
1180 /* check original control_url */
1181 if (!strcmp (stream->control_url, (gchar *) a))
1184 /* check if qualified setup_url ends with string */
1185 if (g_str_has_suffix (stream->control_url, (gchar *) a))
1192 static GstRTSPStream *
1193 find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
1197 /* find and get stream */
1198 if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
1199 return (GstRTSPStream *) lstream->data;
1204 static const GstSDPBandwidth *
1205 gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1206 const GstSDPMedia * media, const gchar * type)
1210 /* first look in the media specific section */
1211 len = gst_sdp_media_bandwidths_len (media);
1212 for (i = 0; i < len; i++) {
1213 const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
1215 if (strcmp (bw->bwtype, type) == 0)
1218 /* then look in the message specific section */
1219 len = gst_sdp_message_bandwidths_len (sdp);
1220 for (i = 0; i < len; i++) {
1221 const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
1223 if (strcmp (bw->bwtype, type) == 0)
1230 gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1231 const GstSDPMedia * media, GstRTSPStream * stream)
1233 const GstSDPBandwidth *bw;
1235 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
1236 stream->as_bandwidth = bw->bandwidth;
1238 stream->as_bandwidth = -1;
1240 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
1241 stream->rr_bandwidth = bw->bandwidth;
1243 stream->rr_bandwidth = -1;
1245 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
1246 stream->rs_bandwidth = bw->bandwidth;
1248 stream->rs_bandwidth = -1;
1252 gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
1253 const GstSDPConnection * conn)
1255 if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
1258 if (conn->addrtype == NULL)
1261 /* check for IPV6 */
1262 if (strcmp (conn->addrtype, "IP4") == 0)
1263 stream->is_ipv6 = FALSE;
1264 else if (strcmp (conn->addrtype, "IP6") == 0)
1265 stream->is_ipv6 = TRUE;
1270 g_free (stream->destination);
1271 stream->destination = g_strdup (conn->address);
1273 /* check for multicast */
1274 stream->is_multicast =
1275 gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
1277 stream->ttl = conn->ttl;
1280 /* Go over the connections for a stream.
1281 * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
1283 * - If we are dealing with a localhost address, we disable multicast
1286 gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
1287 const GstSDPMedia * media, GstRTSPStream * stream)
1289 const GstSDPConnection *conn;
1292 /* first look in the media specific section */
1293 len = gst_sdp_media_connections_len (media);
1294 for (i = 0; i < len; i++) {
1295 conn = gst_sdp_media_get_connection (media, i);
1297 gst_rtspsrc_do_stream_connection (src, stream, conn);
1299 /* then look in the message specific section */
1300 if ((conn = gst_sdp_message_get_connection (sdp))) {
1301 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);
1373 GST_ERROR_OBJECT (src, "can't find proto in media");
1378 GST_ERROR_OBJECT (src, "unknown proto in media %s", proto);
1383 static const gchar *
1384 get_aggregate_control (GstRTSPSrc * src)
1389 base = src->control;
1390 else if (src->content_base)
1391 base = src->content_base;
1392 else if (src->conninfo.url_str)
1393 base = src->conninfo.url_str;
1401 clear_ptmap_item (PtMapItem * item)
1404 gst_caps_unref (item->caps);
1407 static GstRTSPStream *
1408 gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
1410 GstRTSPStream *stream;
1411 const gchar *control_url;
1412 const GstSDPMedia *media;
1414 /* get media, should not return NULL */
1415 media = gst_sdp_message_get_media (sdp, idx);
1419 stream = g_new0 (GstRTSPStream, 1);
1420 stream->parent = src;
1421 /* we mark the pad as not linked, we will mark it as OK when we add the pad to
1423 stream->last_ret = GST_FLOW_NOT_LINKED;
1424 stream->added = FALSE;
1425 stream->setup = FALSE;
1426 stream->skipped = FALSE;
1428 stream->eos = FALSE;
1429 stream->discont = TRUE;
1430 stream->seqbase = -1;
1431 stream->timebase = -1;
1432 stream->profile = GST_RTSP_PROFILE_AVP;
1433 stream->ptmap = g_array_new (FALSE, FALSE, sizeof (PtMapItem));
1434 g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);
1436 /* collect bandwidth information for this steam. FIXME, configure in the RTP
1437 * session manager to scale RTCP. */
1438 gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
1440 /* collect connection info */
1441 gst_rtspsrc_collect_connections (src, sdp, media, stream);
1443 /* make the payload type map */
1444 gst_rtspsrc_collect_payloads (src, sdp, media, stream);
1446 /* collect port number */
1447 stream->port = gst_sdp_media_get_port (media);
1449 /* get control url to construct the setup url. The setup url is used to
1450 * configure the transport of the stream and is used to identity the stream in
1451 * the RTP-Info header field returned from PLAY. */
1452 control_url = gst_sdp_media_get_attribute_val (media, "control");
1453 if (control_url == NULL)
1454 control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
1456 GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
1457 GST_DEBUG_OBJECT (src, " port: %d", stream->port);
1458 GST_DEBUG_OBJECT (src, " container: %d", stream->container);
1459 GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
1461 if (control_url != NULL) {
1462 stream->control_url = g_strdup (control_url);
1463 /* Build a fully qualified url using the content_base if any or by prefixing
1464 * the original request.
1465 * If the control_url starts with a '/' or a non rtsp: protocol we will most
1466 * likely build a URL that the server will fail to understand, this is ok,
1467 * we will fail then. */
1468 if (g_str_has_prefix (control_url, "rtsp://"))
1469 stream->conninfo.location = g_strdup (control_url);
1474 if (g_strcmp0 (control_url, "*") == 0)
1477 base = get_aggregate_control (src);
1479 /* check if the base ends or control starts with / */
1480 has_slash = g_str_has_prefix (control_url, "/");
1481 has_slash = has_slash || g_str_has_suffix (base, "/");
1483 /* concatenate the two strings, insert / when not present */
1484 stream->conninfo.location =
1485 g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
1488 GST_DEBUG_OBJECT (src, " setup: %s",
1489 GST_STR_NULL (stream->conninfo.location));
1491 /* we keep track of all streams */
1492 src->streams = g_list_append (src->streams, stream);
1500 gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
1504 GST_DEBUG_OBJECT (src, "free stream %p", stream);
1506 g_array_free (stream->ptmap, TRUE);
1508 g_free (stream->destination);
1509 g_free (stream->control_url);
1510 g_free (stream->conninfo.location);
1512 for (i = 0; i < 2; i++) {
1513 if (stream->udpsrc[i]) {
1514 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
1515 gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
1516 gst_object_unref (stream->udpsrc[i]);
1518 if (stream->channelpad[i])
1519 gst_object_unref (stream->channelpad[i]);
1520 if (stream->udpsink[i]) {
1521 gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
1522 gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
1523 gst_object_unref (stream->udpsink[i]);
1526 if (stream->fakesrc) {
1527 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
1528 gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
1529 gst_object_unref (stream->fakesrc);
1531 if (stream->srcpad) {
1532 gst_pad_set_active (stream->srcpad, FALSE);
1534 gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
1536 if (stream->rtcppad)
1537 gst_object_unref (stream->rtcppad);
1538 if (stream->session)
1539 g_object_unref (stream->session);
1540 if (stream->srtpdec)
1541 gst_object_unref (stream->srtpdec);
1546 gst_rtspsrc_cleanup (GstRTSPSrc * src)
1550 GST_DEBUG_OBJECT (src, "cleanup");
1552 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1553 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1555 gst_rtspsrc_stream_free (src, stream);
1557 g_list_free (src->streams);
1558 src->streams = NULL;
1560 if (src->manager_sig_id) {
1561 g_signal_handler_disconnect (src->manager, src->manager_sig_id);
1562 src->manager_sig_id = 0;
1564 gst_element_set_state (src->manager, GST_STATE_NULL);
1565 gst_bin_remove (GST_BIN_CAST (src), src->manager);
1566 src->manager = NULL;
1569 gst_structure_free (src->props);
1572 g_free (src->content_base);
1573 src->content_base = NULL;
1575 g_free (src->control);
1576 src->control = NULL;
1579 gst_rtsp_range_free (src->range);
1582 /* don't clear the SDP when it was used in the url */
1583 if (src->sdp && !src->from_sdp) {
1584 gst_sdp_message_free (src->sdp);
1587 if (src->start_segment) {
1588 gst_event_unref (src->start_segment);
1589 src->start_segment = NULL;
1591 if (src->provided_clock) {
1592 gst_object_unref (src->provided_clock);
1593 src->provided_clock = NULL;
1597 #define PARSE_INT(p, del, res) \
1600 p = strstr (p, del); \
1610 #define PARSE_STRING(p, del, res) \
1613 p = strstr (p, del); \
1625 #define SKIP_SPACES(p) \
1626 while (*p && g_ascii_isspace (*p)) \
1631 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1634 gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
1635 gint * rate, gchar ** params)
1639 p = (gchar *) rtpmap;
1641 PARSE_INT (p, " ", *payload);
1649 PARSE_STRING (p, "/", *name);
1650 if (*name == NULL) {
1651 GST_DEBUG ("no rate, name %s", p);
1652 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
1653 * streams seem to omit the rate. */
1660 p = strstr (p, "/");
1678 parse_keymgmt (const gchar * keymgmt, GstCaps * caps)
1680 gboolean res = FALSE;
1684 GstMIKEYMessage *msg;
1685 const GstMIKEYPayload *payload;
1686 const gchar *srtp_cipher;
1687 const gchar *srtp_auth;
1689 p = (gchar *) keymgmt;
1695 PARSE_STRING (p, " ", kmpid);
1696 if (!g_str_equal (kmpid, "mikey"))
1699 data = g_base64_decode (p, &size);
1703 msg = gst_mikey_message_new_from_data (data, size);
1707 srtp_cipher = "aes-128-icm";
1708 srtp_auth = "hmac-sha1-80";
1710 /* check the Security policy if any */
1711 if ((payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, 0))) {
1712 GstMIKEYPayloadSP *p = (GstMIKEYPayloadSP *) payload;
1715 if (p->proto != GST_MIKEY_SEC_PROTO_SRTP)
1718 len = gst_mikey_payload_sp_get_n_params (payload);
1719 for (i = 0; i < len; i++) {
1720 const GstMIKEYPayloadSPParam *param =
1721 gst_mikey_payload_sp_get_param (payload, i);
1723 switch (param->type) {
1724 case GST_MIKEY_SP_SRTP_ENC_ALG:
1725 switch (param->val[0]) {
1727 srtp_cipher = "null";
1731 srtp_cipher = "aes-128-icm";
1737 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1738 switch (param->val[0]) {
1744 srtp_auth = "hmac-sha1-80";
1750 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1752 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1760 if (!(payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_KEMAC, 0)))
1763 GstMIKEYPayloadKEMAC *p = (GstMIKEYPayloadKEMAC *) payload;
1766 if (p->enc_alg != GST_MIKEY_ENC_NULL || p->mac_alg != GST_MIKEY_MAC_NULL)
1770 gst_buffer_new_wrapped (g_memdup (p->enc_data, p->enc_len), p->enc_len);
1771 gst_caps_set_simple (caps, "srtp-key", GST_TYPE_BUFFER, buf, NULL);
1774 gst_caps_set_simple (caps,
1775 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
1776 "srtp-auth", G_TYPE_STRING, srtp_auth,
1777 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
1778 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
1782 gst_mikey_message_free (msg);
1788 * Mapping SDP attributes to caps
1790 * prepend 'a-' to IANA registered sdp attributes names
1791 * (ie: not prefixed with 'x-') in order to avoid
1792 * collision with gstreamer standard caps properties names
1795 gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
1797 if (attributes->len > 0) {
1801 s = gst_caps_get_structure (caps, 0);
1803 for (i = 0; i < attributes->len; i++) {
1804 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
1805 gchar *tofree, *key;
1809 /* skip some of the attribute we already handle */
1810 if (!strcmp (key, "fmtp"))
1812 if (!strcmp (key, "rtpmap"))
1814 if (!strcmp (key, "control"))
1816 if (!strcmp (key, "range"))
1818 if (g_str_equal (key, "key-mgmt")) {
1819 parse_keymgmt (attr->value, caps);
1823 /* string must be valid UTF8 */
1824 if (!g_utf8_validate (attr->value, -1, NULL))
1827 if (!g_str_has_prefix (key, "x-"))
1828 tofree = key = g_strdup_printf ("a-%s", key);
1832 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
1833 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
1839 static const gchar *
1840 rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
1849 if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
1852 if (sscanf (attr, "%d ", &val) != 1)
1862 * Mapping of caps to and from SDP fields:
1864 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1865 * a=fmtp:<payload> <param>[=<value>];...
1868 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
1871 const gchar *rtpmap;
1875 gchar *params = NULL;
1881 /* get and parse rtpmap */
1882 rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
1885 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
1887 g_warning ("error parsing rtpmap, ignoring");
1891 /* dynamic payloads need rtpmap or we fail */
1892 if (rtpmap == NULL && pt >= 96)
1895 /* check if we have a rate, if not, we need to look up the rate from the
1896 * default rates based on the payload types. */
1898 const GstRTPPayloadInfo *info;
1900 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
1901 /* dynamic types, use media and encoding_name */
1902 tmp = g_ascii_strdown (media->media, -1);
1903 info = gst_rtp_payload_info_for_name (tmp, name);
1906 /* static types, use payload type */
1907 info = gst_rtp_payload_info_for_pt (pt);
1911 if ((rate = info->clock_rate) == 0)
1914 /* we fail if we cannot find one */
1919 tmp = g_ascii_strdown (media->media, -1);
1920 caps = gst_caps_new_simple ("application/x-unknown",
1921 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
1923 s = gst_caps_get_structure (caps, 0);
1925 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
1927 /* encoding name must be upper case */
1929 tmp = g_ascii_strup (name, -1);
1930 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
1934 /* params must be lower case */
1935 if (params != NULL) {
1936 tmp = g_ascii_strdown (params, -1);
1937 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
1941 /* parse optional fmtp: field */
1942 if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
1948 /* p is now of the format <payload> <param>[=<value>];... */
1949 PARSE_INT (p, " ", payload);
1950 if (payload != -1 && payload == pt) {
1954 /* <param>[=<value>] are separated with ';' */
1955 pairs = g_strsplit (p, ";", 0);
1956 for (i = 0; pairs[i]; i++) {
1958 const gchar *val, *key;
1960 /* the key may not have a '=', the value can have other '='s */
1961 valpos = strstr (pairs[i], "=");
1963 /* we have a '=' and thus a value, remove the '=' with \0 */
1965 /* value is everything between '=' and ';'. We split the pairs at ;
1966 * boundaries so we can take the remainder of the value. Some servers
1967 * put spaces around the value which we strip off here. Alternatively
1968 * we could strip those spaces in the depayloaders should these spaces
1969 * actually carry any meaning in the future. */
1970 val = g_strstrip (valpos + 1);
1972 /* simple <param>;.. is translated into <param>=1;... */
1975 /* strip the key of spaces, convert key to lowercase but not the value. */
1976 key = g_strstrip (pairs[i]);
1977 if (strlen (key) > 1) {
1978 tmp = g_ascii_strdown (key, -1);
1979 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
1991 g_warning ("rtpmap type not given for dynamic payload %d", pt);
1996 g_warning ("rate unknown for payload type %d", pt);
2002 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
2003 gint * rtpport, gint * rtcpport)
2006 GstStateChangeReturn ret;
2007 GstElement *udpsrc0, *udpsrc1;
2008 gint tmp_rtp, tmp_rtcp;
2012 src = stream->parent;
2018 /* Start at next port */
2019 tmp_rtp = src->next_port_num;
2021 if (stream->is_ipv6)
2022 host = "udp://[::0]";
2024 host = "udp://0.0.0.0";
2026 /* try to allocate 2 UDP ports, the RTP port should be an even
2027 * number and the RTCP port should be the next (uneven) port */
2030 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
2031 tmp_rtp >= src->client_port_range.max)
2034 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2035 if (udpsrc0 == NULL)
2036 goto no_udp_protocol;
2037 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
2039 if (src->udp_buffer_size != 0)
2040 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
2043 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
2044 if (ret == GST_STATE_CHANGE_FAILURE) {
2046 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
2049 if (++count > src->retry)
2052 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2053 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2054 gst_object_unref (udpsrc0);
2057 GST_DEBUG_OBJECT (src, "retry %d", count);
2060 goto no_udp_protocol;
2063 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
2064 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
2066 /* check if port is even */
2067 if ((tmp_rtp & 0x01) != 0) {
2068 /* port not even, close and allocate another */
2069 if (++count > src->retry)
2072 GST_DEBUG_OBJECT (src, "RTP port not even");
2074 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2075 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2076 gst_object_unref (udpsrc0);
2079 GST_DEBUG_OBJECT (src, "retry %d", count);
2084 /* allocate port+1 for RTCP now */
2085 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2086 if (udpsrc1 == NULL)
2087 goto no_udp_rtcp_protocol;
2090 tmp_rtcp = tmp_rtp + 1;
2091 if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
2094 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
2096 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
2097 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
2098 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
2099 if (ret == GST_STATE_CHANGE_FAILURE) {
2100 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
2102 if (++count > src->retry)
2105 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2106 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2107 gst_object_unref (udpsrc0);
2110 GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
2111 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2112 gst_object_unref (udpsrc1);
2116 GST_DEBUG_OBJECT (src, "retry %d", count);
2120 /* all fine, do port check */
2121 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
2122 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
2124 /* this should not happen... */
2125 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
2128 /* we keep these elements, we configure all in configure_transport when the
2129 * server told us to really use the UDP ports. */
2130 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
2131 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
2132 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
2133 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
2135 /* keep track of next available port number when we have a range
2137 if (src->next_port_num != 0)
2138 src->next_port_num = tmp_rtcp + 1;
2145 GST_DEBUG_OBJECT (src, "could not get UDP source");
2150 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
2154 no_udp_rtcp_protocol:
2156 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
2161 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
2162 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
2168 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2169 gst_object_unref (udpsrc0);
2172 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2173 gst_object_unref (udpsrc1);
2180 gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
2185 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
2187 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2188 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2191 for (i = 0; i < 2; i++) {
2192 if (stream->udpsrc[i])
2193 gst_element_set_state (stream->udpsrc[i], state);
2199 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
2206 event = gst_event_new_flush_start ();
2207 GST_DEBUG_OBJECT (src, "start flush");
2209 state = GST_STATE_PAUSED;
2211 event = gst_event_new_flush_stop (FALSE);
2212 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
2215 state = GST_STATE_PLAYING;
2217 state = GST_STATE_PAUSED;
2219 gst_rtspsrc_push_event (src, event);
2220 gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
2221 gst_rtspsrc_set_state (src, state);
2224 static GstRTSPResult
2225 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
2226 GstRTSPMessage * message, GTimeVal * timeout)
2231 ret = gst_rtsp_connection_send (conn, message, timeout);
2233 ret = GST_RTSP_ERROR;
2238 static GstRTSPResult
2239 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
2240 GstRTSPMessage * message, GTimeVal * timeout)
2245 ret = gst_rtsp_connection_receive (conn, message, timeout);
2247 ret = GST_RTSP_ERROR;
2253 gst_rtspsrc_get_position (GstRTSPSrc * src)
2258 query = gst_query_new_position (GST_FORMAT_TIME);
2259 /* should be known somewhere down the stream (e.g. jitterbuffer) */
2260 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2261 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2265 if (stream->srcpad) {
2266 if (gst_pad_query (stream->srcpad, query)) {
2267 gst_query_parse_position (query, &fmt, &pos);
2268 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
2269 GST_TIME_ARGS (pos));
2270 src->last_pos = pos;
2280 gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
2282 src->state = GST_RTSP_STATE_SEEKING;
2283 /* PLAY will add the range header now. */
2284 src->need_range = TRUE;
2290 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
2295 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
2297 gboolean flush, skip;
2300 GstSegment seeksegment = { 0, };
2304 GST_DEBUG_OBJECT (src, "doing seek with event");
2306 gst_event_parse_seek (event, &rate, &format, &flags,
2307 &cur_type, &cur, &stop_type, &stop);
2309 /* no negative rates yet */
2313 /* we need TIME format */
2314 if (format != src->segment.format)
2317 GST_DEBUG_OBJECT (src, "doing seek without event");
2319 cur_type = GST_SEEK_TYPE_SET;
2320 stop_type = GST_SEEK_TYPE_SET;
2323 /* get flush flag */
2324 flush = flags & GST_SEEK_FLAG_FLUSH;
2325 skip = flags & GST_SEEK_FLAG_SKIP;
2327 /* now we need to make sure the streaming thread is stopped. We do this by
2328 * either sending a FLUSH_START event downstream which will cause the
2329 * streaming thread to stop with a WRONG_STATE.
2330 * For a non-flushing seek we simply pause the task, which will happen as soon
2331 * as it completes one iteration (and thus might block when the sink is
2332 * blocking in preroll). */
2334 GST_DEBUG_OBJECT (src, "starting flush");
2335 gst_rtspsrc_flush (src, TRUE, FALSE);
2338 gst_task_pause (src->task);
2342 /* we should now be able to grab the streaming thread because we stopped it
2343 * with the above flush/pause code */
2344 GST_RTSP_STREAM_LOCK (src);
2346 GST_DEBUG_OBJECT (src, "stopped streaming");
2348 /* stop flushing the rtsp connection so we can send PAUSE/PLAY below */
2349 gst_rtspsrc_connection_flush (src, FALSE);
2351 /* copy segment, we need this because we still need the old
2352 * segment when we close the current segment. */
2353 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
2355 /* configure the seek parameters in the seeksegment. We will then have the
2356 * right values in the segment to perform the seek */
2358 GST_DEBUG_OBJECT (src, "configuring seek");
2359 gst_segment_do_seek (&seeksegment, rate, format, flags,
2360 cur_type, cur, stop_type, stop, &update);
2363 /* figure out the last position we need to play. If it's configured (stop !=
2364 * -1), use that, else we play until the total duration of the file */
2365 if ((stop = seeksegment.stop) == -1)
2366 stop = seeksegment.duration;
2368 playing = (src->state == GST_RTSP_STATE_PLAYING);
2370 /* if we were playing, pause first */
2372 /* obtain current position in case seek fails */
2373 gst_rtspsrc_get_position (src);
2374 gst_rtspsrc_pause (src, FALSE);
2378 gst_rtspsrc_do_seek (src, &seeksegment);
2380 /* and continue playing */
2382 gst_rtspsrc_play (src, &seeksegment, FALSE);
2384 /* prepare for streaming again */
2386 /* if we started flush, we stop now */
2387 GST_DEBUG_OBJECT (src, "stopping flush");
2388 gst_rtspsrc_flush (src, FALSE, playing);
2391 /* now we did the seek and can activate the new segment values */
2392 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
2394 /* if we're doing a segment seek, post a SEGMENT_START message */
2395 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2396 gst_element_post_message (GST_ELEMENT_CAST (src),
2397 gst_message_new_segment_start (GST_OBJECT_CAST (src),
2398 src->segment.format, src->segment.position));
2401 /* now create the newsegment */
2402 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
2403 " to %" G_GINT64_FORMAT, src->segment.position, stop);
2406 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
2407 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2408 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2409 stream->discont = TRUE;
2412 GST_RTSP_STREAM_UNLOCK (src);
2419 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
2424 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
2430 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
2434 gboolean res = TRUE;
2437 src = GST_RTSPSRC_CAST (parent);
2439 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
2440 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
2442 switch (GST_EVENT_TYPE (event)) {
2443 case GST_EVENT_SEEK:
2444 res = gst_rtspsrc_perform_seek (src, event);
2448 case GST_EVENT_NAVIGATION:
2449 case GST_EVENT_LATENCY:
2457 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
2458 res = gst_pad_send_event (target, event);
2459 gst_object_unref (target);
2461 gst_event_unref (event);
2464 gst_event_unref (event);
2470 /* this is the final event function we receive on the internal source pad when
2471 * we deal with TCP connections */
2473 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
2478 GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
2480 switch (GST_EVENT_TYPE (event)) {
2481 case GST_EVENT_SEEK:
2483 case GST_EVENT_NAVIGATION:
2484 case GST_EVENT_LATENCY:
2486 gst_event_unref (event);
2493 /* this is the final query function we receive on the internal source pad when
2494 * we deal with TCP connections */
2496 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
2500 gboolean res = TRUE;
2502 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
2504 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2505 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2507 switch (GST_QUERY_TYPE (query)) {
2508 case GST_QUERY_POSITION:
2513 case GST_QUERY_DURATION:
2517 gst_query_parse_duration (query, &format, NULL);
2520 case GST_FORMAT_TIME:
2521 gst_query_set_duration (query, format, src->segment.duration);
2529 case GST_QUERY_LATENCY:
2531 /* we are live with a min latency of 0 and unlimited max latency, this
2532 * result will be updated by the session manager if there is any. */
2533 gst_query_set_latency (query, TRUE, 0, -1);
2543 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2545 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2549 gboolean res = FALSE;
2551 src = GST_RTSPSRC_CAST (parent);
2553 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2554 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2556 switch (GST_QUERY_TYPE (query)) {
2557 case GST_QUERY_DURATION:
2561 gst_query_parse_duration (query, &format, NULL);
2564 case GST_FORMAT_TIME:
2565 gst_query_set_duration (query, format, src->segment.duration);
2573 case GST_QUERY_SEEKING:
2577 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2578 if (format == GST_FORMAT_TIME) {
2580 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2582 /* seeking without duration is unlikely */
2583 seekable = seekable && src->seekable && src->segment.duration &&
2584 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2586 /* FIXME ?? should we have 0 and segment.duration here; see demuxers */
2587 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
2588 src->segment.start, src->segment.stop);
2597 uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
2599 gst_query_set_uri (query, uri);
2607 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2609 /* forward the query to the proxy target pad */
2611 res = gst_pad_query (target, query);
2612 gst_object_unref (target);
2621 /* callback for RTCP messages to be sent to the server when operating in TCP
2623 static GstFlowReturn
2624 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2627 GstRTSPStream *stream;
2628 GstFlowReturn res = GST_FLOW_OK;
2633 GstRTSPMessage message = { 0 };
2634 GstRTSPConnection *conn;
2636 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2637 src = stream->parent;
2639 gst_buffer_map (buffer, &map, GST_MAP_READ);
2643 gst_rtsp_message_init_data (&message, stream->channel[1]);
2645 /* lend the body data to the message */
2646 gst_rtsp_message_take_body (&message, data, size);
2648 if (stream->conninfo.connection)
2649 conn = stream->conninfo.connection;
2651 conn = src->conninfo.connection;
2653 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2654 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2655 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2657 /* and steal it away again because we will free it when unreffing the
2659 gst_rtsp_message_steal_body (&message, &data, &size);
2660 gst_rtsp_message_unset (&message);
2662 gst_buffer_unmap (buffer, &map);
2663 gst_buffer_unref (buffer);
2668 static GstPadProbeReturn
2669 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2671 GstRTSPSrc *src = user_data;
2673 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2674 GST_DEBUG_PAD_NAME (pad));
2676 /* activate the streams */
2677 GST_OBJECT_LOCK (src);
2678 if (!src->need_activate)
2681 src->need_activate = FALSE;
2682 GST_OBJECT_UNLOCK (src);
2684 gst_rtspsrc_activate_streams (src);
2686 return GST_PAD_PROBE_OK;
2690 GST_OBJECT_UNLOCK (src);
2691 return GST_PAD_PROBE_OK;
2696 copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
2698 GstPad *gpad = GST_PAD_CAST (user_data);
2700 GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
2701 gst_pad_store_sticky_event (gpad, *event);
2706 /* this callback is called when the session manager generated a new src pad with
2707 * payloaded RTP packets. We simply ghost the pad here. */
2709 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2712 GstPadTemplate *template;
2715 GstRTSPStream *stream;
2718 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2720 GST_RTSP_STATE_LOCK (src);
2722 name = gst_object_get_name (GST_OBJECT_CAST (pad));
2723 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
2724 goto unknown_stream;
2726 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);
2728 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
2730 goto unknown_stream;
2733 stream->ssrc = ssrc;
2735 /* we'll add it later see below */
2736 stream->added = TRUE;
2738 /* check if we added all streams */
2740 for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
2741 GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;
2743 GST_DEBUG_OBJECT (src, "stream %p, container %d, added %d, setup %d",
2744 ostream, ostream->container, ostream->added, ostream->setup);
2746 /* if we find a stream for which we did a setup that is not added, we
2747 * need to wait some more */
2748 if (ostream->setup && !ostream->added) {
2753 GST_RTSP_STATE_UNLOCK (src);
2755 /* create a new pad we will use to stream to */
2756 template = gst_static_pad_template_get (&rtptemplate);
2757 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
2758 gst_object_unref (template);
2761 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
2762 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
2763 gst_pad_set_active (stream->srcpad, TRUE);
2764 gst_pad_sticky_events_foreach (pad, copy_sticky_events, stream->srcpad);
2765 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
2768 GST_DEBUG_OBJECT (src, "We added all streams");
2769 /* when we get here, all stream are added and we can fire the no-more-pads
2771 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
2779 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
2780 GST_RTSP_STATE_UNLOCK (src);
2787 stream_get_caps_for_pt (GstRTSPStream * stream, guint pt)
2791 len = stream->ptmap->len;
2792 for (i = 0; i < len; i++) {
2793 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
2801 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
2803 GstRTSPStream *stream;
2806 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
2808 GST_RTSP_STATE_LOCK (src);
2809 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2811 goto unknown_stream;
2813 if ((caps = stream_get_caps_for_pt (stream, pt)))
2814 gst_caps_ref (caps);
2815 GST_RTSP_STATE_UNLOCK (src);
2821 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
2822 GST_RTSP_STATE_UNLOCK (src);
2828 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
2830 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
2836 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
2842 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
2848 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2850 GstRTSPSrc *src = stream->parent;
2853 g_object_get (source, "ssrc", &ssrc, NULL);
2855 GST_DEBUG_OBJECT (src, "source %08x, stream %08x, session %u received BYE",
2856 ssrc, stream->ssrc, stream->id);
2858 if (ssrc == stream->ssrc)
2859 gst_rtspsrc_do_stream_eos (src, stream);
2863 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2865 GstRTSPSrc *src = stream->parent;
2868 g_object_get (source, "ssrc", &ssrc, NULL);
2870 GST_WARNING_OBJECT (src, "source %08x, stream %08x in session %u timed out",
2871 ssrc, stream->ssrc, stream->id);
2873 if (ssrc == stream->ssrc)
2874 gst_rtspsrc_do_stream_eos (src, stream);
2878 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
2880 GstRTSPStream *stream;
2882 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
2884 /* get stream for session */
2885 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2887 gst_rtspsrc_do_stream_eos (src, stream);
2892 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
2894 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
2899 set_manager_buffer_mode (GstRTSPSrc * src)
2901 GObjectClass *klass;
2903 if (src->manager == NULL)
2906 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
2908 if (!g_object_class_find_property (klass, "buffer-mode"))
2911 if (src->buffer_mode != BUFFER_MODE_AUTO) {
2912 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
2917 GST_DEBUG_OBJECT (src,
2918 "auto buffering mode, have clock %" GST_PTR_FORMAT, src->provided_clock);
2920 if (src->provided_clock) {
2921 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
2923 if (clock == src->provided_clock) {
2924 GST_DEBUG_OBJECT (src, "selected synced");
2925 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SYNCED, NULL);
2928 gst_object_unref (clock);
2933 /* Otherwise fall-through and use another buffer mode */
2935 gst_object_unref (clock);
2938 GST_DEBUG_OBJECT (src, "auto buffering mode");
2939 if (src->use_buffering) {
2940 GST_DEBUG_OBJECT (src, "selected buffer");
2941 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER, NULL);
2943 GST_DEBUG_OBJECT (src, "selected slave");
2944 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
2949 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
2951 GST_DEBUG ("request key %u", ssrc);
2952 return gst_caps_ref (stream_get_caps_for_pt (stream, stream->default_pt));
2956 request_rtp_decoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
2958 if (stream->id != session)
2961 if (stream->profile != GST_RTSP_PROFILE_SAVP &&
2962 stream->profile != GST_RTSP_PROFILE_SAVPF)
2965 if (stream->srtpdec == NULL) {
2968 name = g_strdup_printf ("srtpdec_%u", session);
2969 stream->srtpdec = gst_element_factory_make ("srtpdec", name);
2972 g_signal_connect (stream->srtpdec, "request-key",
2973 (GCallback) request_key, stream);
2975 return gst_object_ref (stream->srtpdec);
2978 /* try to get and configure a manager */
2980 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
2981 GstRTSPTransport * transport)
2983 const gchar *manager;
2985 GstStateChangeReturn ret;
2987 /* find a manager */
2988 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
2992 GST_DEBUG_OBJECT (src, "using manager %s", manager);
2994 /* configure the manager */
2995 if (src->manager == NULL) {
2996 GObjectClass *klass;
2998 if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
3000 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
3004 goto use_no_manager;
3006 if (!(src->manager = gst_element_factory_make (manager, "manager")))
3007 goto manager_failed;
3010 /* we manage this element */
3011 gst_element_set_locked_state (src->manager, TRUE);
3012 gst_bin_add (GST_BIN_CAST (src), src->manager);
3014 ret = gst_element_set_state (src->manager, GST_STATE_PAUSED);
3015 if (ret == GST_STATE_CHANGE_FAILURE)
3016 goto start_manager_failure;
3018 g_object_set (src->manager, "latency", src->latency, NULL);
3020 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
3022 if (g_object_class_find_property (klass, "ntp-sync")) {
3023 g_object_set (src->manager, "ntp-sync", src->ntp_sync, NULL);
3026 if (g_object_class_find_property (klass, "use-pipeline-clock")) {
3027 g_object_set (src->manager, "use-pipeline-clock",
3028 src->use_pipeline_clock, NULL);
3031 if (src->sdes && g_object_class_find_property (klass, "sdes")) {
3032 g_object_set (src->manager, "sdes", src->sdes, NULL);
3035 if (g_object_class_find_property (klass, "drop-on-latency")) {
3036 g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
3040 /* buffer mode pauses are handled by adding offsets to buffer times,
3041 * but some depayloaders may have a hard time syncing output times
3042 * with such input times, e.g. container ones, most notably ASF */
3043 /* TODO alternatives are having an event that indicates these shifts,
3044 * or having rtsp extensions provide suggestion on buffer mode */
3045 /* valid duration implies not likely live pipeline,
3046 * so slaving in jitterbuffer does not make much sense
3047 * (and might mess things up due to bursts) */
3048 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
3049 src->segment.duration && !stream->container) {
3050 src->use_buffering = TRUE;
3052 src->use_buffering = FALSE;
3055 set_manager_buffer_mode (src);
3057 /* connect to signals */
3058 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
3060 src->manager_sig_id =
3061 g_signal_connect (src->manager, "pad-added",
3062 (GCallback) new_manager_pad, src);
3063 src->manager_ptmap_id =
3064 g_signal_connect (src->manager, "request-pt-map",
3065 (GCallback) request_pt_map, src);
3067 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
3070 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_NEW_MANAGER], 0,
3073 g_signal_connect (src->manager, "request-rtp-decoder",
3074 (GCallback) request_rtp_decoder, stream);
3075 g_signal_connect (src->manager, "request-rtcp-decoder",
3076 (GCallback) request_rtp_decoder, stream);
3078 /* we stream directly to the manager, get some pads. Each RTSP stream goes
3079 * into a separate RTP session. */
3080 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
3081 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
3083 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
3084 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
3087 /* now configure the bandwidth in the manager */
3088 if (g_signal_lookup ("get-internal-session",
3089 G_OBJECT_TYPE (src->manager)) != 0) {
3090 GObject *rtpsession;
3092 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
3095 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
3097 stream->session = rtpsession;
3099 if (stream->as_bandwidth != -1) {
3100 GST_INFO_OBJECT (src, "setting AS: %f",
3101 (gdouble) (stream->as_bandwidth * 1000));
3102 g_object_set (rtpsession, "bandwidth",
3103 (gdouble) (stream->as_bandwidth * 1000), NULL);
3105 if (stream->rr_bandwidth != -1) {
3106 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
3107 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
3110 if (stream->rs_bandwidth != -1) {
3111 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
3112 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
3116 g_object_set (rtpsession, "probation", src->probation, NULL);
3118 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
3120 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
3122 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
3124 g_signal_connect (rtpsession, "on-ssrc-active",
3125 (GCallback) on_ssrc_active, stream);
3136 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3141 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
3144 start_manager_failure:
3146 GST_DEBUG_OBJECT (src, "could not start session manager");
3151 /* free the UDP sources allocated when negotiating a transport.
3152 * This function is called when the server negotiated to a transport where the
3153 * UDP sources are not needed anymore, such as TCP or multicast. */
3155 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
3159 for (i = 0; i < 2; i++) {
3160 if (stream->udpsrc[i]) {
3161 GST_DEBUG ("free UDP source %d for stream %p", i, stream);
3162 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
3163 gst_object_unref (stream->udpsrc[i]);
3164 stream->udpsrc[i] = NULL;
3169 /* for TCP, create pads to send and receive data to and from the manager and to
3170 * intercept various events and queries
3173 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
3174 GstRTSPTransport * transport, GstPad ** outpad)
3177 GstPadTemplate *template;
3178 GstPad *pad0, *pad1;
3180 /* configure for interleaved delivery, nothing needs to be done
3181 * here, the loop function will call the chain functions of the
3182 * session manager. */
3183 stream->channel[0] = transport->interleaved.min;
3184 stream->channel[1] = transport->interleaved.max;
3185 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
3186 stream->channel[0], stream->channel[1]);
3188 /* we can remove the allocated UDP ports now */
3189 gst_rtspsrc_stream_free_udp (stream);
3191 /* no session manager, send data to srcpad directly */
3192 if (!stream->channelpad[0]) {
3193 GST_DEBUG_OBJECT (src, "no manager, creating pad");
3195 /* create a new pad we will use to stream to */
3196 name = g_strdup_printf ("stream_%u", stream->id);
3197 template = gst_static_pad_template_get (&rtptemplate);
3198 stream->channelpad[0] = gst_pad_new_from_template (template, name);
3199 gst_object_unref (template);
3202 /* set caps and activate */
3203 gst_pad_use_fixed_caps (stream->channelpad[0]);
3204 gst_pad_set_active (stream->channelpad[0], TRUE);
3206 *outpad = gst_object_ref (stream->channelpad[0]);
3208 GST_DEBUG_OBJECT (src, "using manager source pad");
3210 template = gst_static_pad_template_get (&anysrctemplate);
3212 /* allocate pads for sending the channel data into the manager */
3213 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
3214 gst_pad_link_full (pad0, stream->channelpad[0], GST_PAD_LINK_CHECK_NOTHING);
3215 gst_object_unref (stream->channelpad[0]);
3216 stream->channelpad[0] = pad0;
3217 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
3218 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
3219 gst_pad_set_element_private (pad0, src);
3220 gst_pad_set_active (pad0, TRUE);
3222 if (stream->channelpad[1]) {
3223 /* if we have a sinkpad for the other channel, create a pad and link to the
3225 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
3226 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
3227 gst_pad_link_full (pad1, stream->channelpad[1],
3228 GST_PAD_LINK_CHECK_NOTHING);
3229 gst_object_unref (stream->channelpad[1]);
3230 stream->channelpad[1] = pad1;
3231 gst_pad_set_active (pad1, TRUE);
3233 gst_object_unref (template);
3235 /* setup RTCP transport back to the server if we have to. */
3236 if (src->manager && src->do_rtcp) {
3239 template = gst_static_pad_template_get (&anysinktemplate);
3241 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
3242 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
3243 gst_pad_set_element_private (stream->rtcppad, stream);
3244 gst_pad_set_active (stream->rtcppad, TRUE);
3246 /* get session RTCP pad */
3247 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3248 pad = gst_element_get_request_pad (src->manager, name);
3253 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3254 gst_object_unref (pad);
3257 gst_object_unref (template);
3263 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
3264 GstRTSPTransport * transport, const gchar ** destination, gint * min,
3265 gint * max, guint * ttl)
3267 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3269 if (!(*destination = transport->destination))
3270 *destination = stream->destination;
3273 /* transport first */
3274 *min = transport->port.min;
3275 *max = transport->port.max;
3276 if (*min == -1 && *max == -1) {
3277 /* then try from SDP */
3278 if (stream->port != 0) {
3279 *min = stream->port;
3280 *max = stream->port + 1;
3286 if (!(*ttl = transport->ttl))
3291 /* first take the source, then the endpoint to figure out where to send
3293 if (!(*destination = transport->source)) {
3294 if (src->conninfo.connection)
3295 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
3296 else if (stream->conninfo.connection)
3298 gst_rtsp_connection_get_ip (stream->conninfo.connection);
3302 /* for unicast we only expect the ports here */
3303 *min = transport->server_port.min;
3304 *max = transport->server_port.max;
3309 /* For multicast create UDP sources and join the multicast group. */
3311 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
3312 GstRTSPTransport * transport, GstPad ** outpad)
3315 const gchar *destination;
3318 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
3320 /* we can remove the allocated UDP ports now */
3321 gst_rtspsrc_stream_free_udp (stream);
3323 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
3326 /* we need a destination now */
3327 if (destination == NULL)
3328 goto no_destination;
3330 /* we really need ports now or we won't be able to receive anything at all */
3331 if (min == -1 && max == -1)
3334 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
3335 destination, min, max);
3337 /* creating UDP source for RTP */
3339 uri = g_strdup_printf ("udp://%s:%d", destination, min);
3341 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3343 if (stream->udpsrc[0] == NULL)
3346 /* take ownership */
3347 gst_object_ref_sink (stream->udpsrc[0]);
3349 if (src->udp_buffer_size != 0)
3350 g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
3351 src->udp_buffer_size, NULL);
3353 if (src->multi_iface != NULL)
3354 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3355 src->multi_iface, NULL);
3358 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3359 gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
3362 /* creating another UDP source for RTCP */
3366 uri = g_strdup_printf ("udp://%s:%d", destination, max);
3368 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3370 if (stream->udpsrc[1] == NULL)
3373 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3374 stream->profile == GST_RTSP_PROFILE_SAVPF)
3375 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3377 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3378 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3379 gst_caps_unref (caps);
3381 /* take ownership */
3382 gst_object_ref_sink (stream->udpsrc[1]);
3384 if (src->multi_iface != NULL)
3385 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3386 src->multi_iface, NULL);
3388 gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
3395 GST_DEBUG_OBJECT (src, "no UDP source element found");
3400 GST_DEBUG_OBJECT (src, "no destination found");
3405 GST_DEBUG_OBJECT (src, "no ports found");
3410 /* configure the remainder of the UDP ports */
3412 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
3413 GstRTSPTransport * transport, GstPad ** outpad)
3415 /* we manage the UDP elements now. For unicast, the UDP sources where
3416 * allocated in the stream when we suggested a transport. */
3417 if (stream->udpsrc[0]) {
3420 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3421 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
3423 GST_DEBUG_OBJECT (src, "setting up UDP source");
3425 /* configure a timeout on the UDP port. When the timeout message is
3426 * posted, we assume UDP transport is not possible. We reconnect using TCP
3428 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
3429 src->udp_timeout * 1000, NULL);
3431 if ((caps = stream_get_caps_for_pt (stream, stream->default_pt)))
3432 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
3434 /* get output pad of the UDP source. */
3435 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
3437 /* save it so we can unblock */
3438 stream->blockedpad = *outpad;
3440 /* configure pad block on the pad. As soon as there is dataflow on the
3441 * UDP source, we know that UDP is not blocked by a firewall and we can
3442 * configure all the streams to let the application autoplug decoders. */
3444 gst_pad_add_probe (stream->blockedpad,
3445 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3446 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocked, src, NULL);
3448 if (stream->channelpad[0]) {
3449 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
3450 /* configure for UDP delivery, we need to connect the UDP pads to
3451 * the session plugin. */
3452 gst_pad_link_full (*outpad, stream->channelpad[0],
3453 GST_PAD_LINK_CHECK_NOTHING);
3454 gst_object_unref (*outpad);
3456 /* we connected to pad-added signal to get pads from the manager */
3458 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
3463 if (stream->udpsrc[1]) {
3466 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
3467 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
3469 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3470 stream->profile == GST_RTSP_PROFILE_SAVPF)
3471 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3473 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3474 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3475 gst_caps_unref (caps);
3477 if (stream->channelpad[1]) {
3480 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
3482 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
3483 gst_pad_link_full (pad, stream->channelpad[1],
3484 GST_PAD_LINK_CHECK_NOTHING);
3485 gst_object_unref (pad);
3487 /* leave unlinked */
3493 /* configure the UDP sink back to the server for status reports */
3495 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
3496 GstRTSPStream * stream, GstRTSPTransport * transport)
3499 gint rtp_port, rtcp_port;
3500 gboolean do_rtp, do_rtcp;
3501 const gchar *destination;
3506 /* get transport info */
3507 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
3508 &rtp_port, &rtcp_port, &ttl);
3510 /* see what we need to do */
3511 do_rtp = (rtp_port != -1);
3512 /* it's possible that the server does not want us to send RTCP in which case
3514 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
3516 /* we need a destination when we have RTP or RTCP ports */
3517 if (destination == NULL && (do_rtp || do_rtcp))
3518 goto no_destination;
3520 /* try to construct the fakesrc to the RTP port of the server to open up any
3523 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
3526 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
3527 stream->udpsink[0] =
3528 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3530 if (stream->udpsink[0] == NULL)
3531 goto no_sink_element;
3533 /* don't join multicast group, we will have the source socket do that */
3534 /* no sync or async state changes needed */
3535 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
3536 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3538 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3540 if (stream->udpsrc[0]) {
3541 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
3542 * so that NAT firewalls will open a hole for us */
3543 g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
3544 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
3545 /* configure socket and make sure udpsink does not close it when shutting
3546 * down, it belongs to udpsrc after all. */
3547 g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
3548 "close-socket", FALSE, NULL);
3549 g_object_unref (socket);
3552 /* the source for the dummy packets to open up NAT */
3553 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
3554 if (stream->fakesrc == NULL)
3555 goto no_fakesrc_element;
3557 /* random data in 5 buffers, a size of 200 bytes should be fine */
3558 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
3559 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
3561 /* we don't want to consider this a sink */
3562 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
3564 /* keep everything locked */
3565 gst_element_set_locked_state (stream->udpsink[0], TRUE);
3566 gst_element_set_locked_state (stream->fakesrc, TRUE);
3568 gst_object_ref (stream->udpsink[0]);
3569 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
3570 gst_object_ref (stream->fakesrc);
3571 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
3573 gst_element_link_pads_full (stream->fakesrc, "src", stream->udpsink[0],
3574 "sink", GST_PAD_LINK_CHECK_NOTHING);
3577 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
3580 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
3581 stream->udpsink[1] =
3582 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3584 if (stream->udpsink[1] == NULL)
3585 goto no_sink_element;
3587 /* don't join multicast group, we will have the source socket do that */
3588 /* no sync or async state changes needed */
3589 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
3590 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3592 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3594 if (stream->udpsrc[1]) {
3595 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
3596 * because some servers check the port number of where it sends RTCP to identify
3597 * the RTCP packets it receives */
3598 g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
3599 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
3600 /* configure socket and make sure udpsink does not close it when shutting
3601 * down, it belongs to udpsrc after all. */
3602 g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
3603 "close-socket", FALSE, NULL);
3604 g_object_unref (socket);
3607 /* we don't want to consider this a sink */
3608 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
3610 /* we keep this playing always */
3611 gst_element_set_locked_state (stream->udpsink[1], TRUE);
3612 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
3614 gst_object_ref (stream->udpsink[1]);
3615 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
3617 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
3619 /* get session RTCP pad */
3620 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3621 pad = gst_element_get_request_pad (src->manager, name);
3626 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3627 gst_object_unref (pad);
3636 GST_DEBUG_OBJECT (src, "no destination address specified");
3641 GST_DEBUG_OBJECT (src, "no UDP sink element found");
3646 GST_DEBUG_OBJECT (src, "no fakesrc element found");
3651 /* sets up all elements needed for streaming over the specified transport.
3652 * Does not yet expose the element pads, this will be done when there is actuall
3653 * dataflow detected, which might never happen when UDP is blocked in a
3654 * firewall, for example.
3657 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
3658 GstRTSPTransport * transport)
3661 GstPad *outpad = NULL;
3662 GstPadTemplate *template;
3664 const gchar *media_type;
3667 src = stream->parent;
3669 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
3671 /* get the proper media type for this stream now */
3672 if (gst_rtsp_transport_get_media_type (transport, &media_type) < 0)
3673 goto unknown_transport;
3675 goto unknown_transport;
3677 /* configure the final media type */
3678 GST_DEBUG_OBJECT (src, "setting media type to %s", media_type);
3680 len = stream->ptmap->len;
3681 for (i = 0; i < len; i++) {
3683 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
3685 if (item->caps == NULL)
3688 s = gst_caps_get_structure (item->caps, 0);
3689 gst_structure_set_name (s, media_type);
3690 /* set ssrc if known */
3691 if (transport->ssrc)
3692 gst_structure_set (s, "ssrc", G_TYPE_UINT, transport->ssrc, NULL);
3695 /* try to get and configure a manager, channelpad[0-1] will be configured with
3696 * the pads for the manager, or NULL when no manager is needed. */
3697 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
3700 switch (transport->lower_transport) {
3701 case GST_RTSP_LOWER_TRANS_TCP:
3702 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
3703 goto transport_failed;
3705 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3706 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
3707 goto transport_failed;
3708 /* fallthrough, the rest is the same for UDP and MCAST */
3709 case GST_RTSP_LOWER_TRANS_UDP:
3710 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
3711 goto transport_failed;
3712 /* configure udpsinks back to the server for RTCP messages and for the
3713 * dummy RTP messages to open NAT. */
3714 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
3715 goto transport_failed;
3718 goto unknown_transport;
3722 GST_DEBUG_OBJECT (src, "creating ghostpad");
3724 gst_pad_use_fixed_caps (outpad);
3726 /* create ghostpad, don't add just yet, this will be done when we activate
3728 name = g_strdup_printf ("stream_%u", stream->id);
3729 template = gst_static_pad_template_get (&rtptemplate);
3730 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
3731 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
3732 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
3733 gst_object_unref (template);
3736 gst_object_unref (outpad);
3738 /* mark pad as ok */
3739 stream->last_ret = GST_FLOW_OK;
3746 GST_DEBUG_OBJECT (src, "failed to configure transport");
3751 GST_DEBUG_OBJECT (src, "unknown transport");
3756 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3761 /* send a couple of dummy random packets on the receiver RTP port to the server,
3762 * this should make a firewall think we initiated the data transfer and
3763 * hopefully allow packets to go from the sender port to our RTP receiver port */
3765 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
3769 if (src->nat_method != GST_RTSP_NAT_DUMMY)
3772 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3773 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3775 if (stream->fakesrc && stream->udpsink[0]) {
3776 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
3777 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
3778 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
3779 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
3780 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
3786 /* Adds the source pads of all configured streams to the element.
3787 * This code is performed when we detected dataflow.
3789 * We detect dataflow from either the _loop function or with pad probes on the
3793 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
3797 GST_DEBUG_OBJECT (src, "activating streams");
3799 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3800 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3802 if (stream->udpsrc[0]) {
3803 /* remove timeout, we are streaming now and timeouts will be handled by
3804 * the session manager and jitter buffer */
3805 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
3807 if (stream->srcpad) {
3808 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
3809 gst_pad_set_active (stream->srcpad, TRUE);
3811 /* if we don't have a session manager, set the caps now. If we have a
3812 * session, we will get a notification of the pad and the caps. */
3813 if (!src->manager) {
3816 caps = stream_get_caps_for_pt (stream, stream->default_pt);
3817 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
3818 gst_pad_set_caps (stream->srcpad, caps);
3821 if (!stream->added) {
3822 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
3823 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
3824 stream->added = TRUE;
3829 /* unblock all pads */
3830 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3831 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3833 if (stream->blockid) {
3834 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
3835 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
3836 stream->blockid = 0;
3844 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
3845 gboolean reset_manager)
3848 guint64 start, stop;
3849 gdouble play_speed, play_scale;
3851 GST_DEBUG_OBJECT (src, "configuring stream caps");
3853 start = segment->position;
3854 stop = segment->duration;
3855 play_speed = segment->rate;
3856 play_scale = segment->applied_rate;
3858 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3859 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3865 len = stream->ptmap->len;
3866 for (j = 0; j < len; j++) {
3868 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, j);
3870 if (item->caps == NULL)
3873 caps = gst_caps_make_writable (item->caps);
3875 if (stream->timebase != -1)
3876 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
3877 (guint) stream->timebase, NULL);
3878 if (stream->seqbase != -1)
3879 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
3880 (guint) stream->seqbase, NULL);
3881 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
3883 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
3884 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
3885 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
3888 GST_DEBUG_OBJECT (src, "stream %p, pt %d, caps %" GST_PTR_FORMAT, stream,
3891 if (item->pt == stream->default_pt && stream->udpsrc[0]) {
3892 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
3896 if (reset_manager && src->manager) {
3897 GST_DEBUG_OBJECT (src, "clear session");
3898 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
3902 static GstFlowReturn
3903 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
3908 /* store the value */
3909 stream->last_ret = ret;
3911 /* if it's success we can return the value right away */
3912 if (ret == GST_FLOW_OK)
3915 /* any other error that is not-linked can be returned right
3917 if (ret != GST_FLOW_NOT_LINKED)
3920 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
3921 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3922 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3924 ret = ostream->last_ret;
3925 /* some other return value (must be SUCCESS but we can return
3926 * other values as well) */
3927 if (ret != GST_FLOW_NOT_LINKED)
3930 /* if we get here, all other pads were unlinked and we return
3931 * NOT_LINKED then */
3937 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
3940 gboolean res = TRUE;
3942 /* only streams that have a connection to the outside world */
3946 if (stream->udpsrc[0]) {
3947 gst_event_ref (event);
3948 res = gst_element_send_event (stream->udpsrc[0], event);
3949 } else if (stream->channelpad[0]) {
3950 gst_event_ref (event);
3951 if (GST_PAD_IS_SRC (stream->channelpad[0]))
3952 res = gst_pad_push_event (stream->channelpad[0], event);
3954 res = gst_pad_send_event (stream->channelpad[0], event);
3957 if (stream->udpsrc[1]) {
3958 gst_event_ref (event);
3959 res &= gst_element_send_event (stream->udpsrc[1], event);
3960 } else if (stream->channelpad[1]) {
3961 gst_event_ref (event);
3962 if (GST_PAD_IS_SRC (stream->channelpad[1]))
3963 res &= gst_pad_push_event (stream->channelpad[1], event);
3965 res &= gst_pad_send_event (stream->channelpad[1], event);
3969 gst_event_unref (event);
3975 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
3978 gboolean res = TRUE;
3980 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3981 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3983 gst_event_ref (event);
3984 res &= gst_rtspsrc_stream_push_event (src, ostream, event);
3986 gst_event_unref (event);
3991 static GstRTSPResult
3992 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3997 if (info->connection == NULL) {
3998 if (info->url == NULL) {
3999 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
4000 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
4004 /* create connection */
4005 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
4006 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
4007 goto could_not_create;
4010 g_free (info->url_str);
4011 info->url_str = gst_rtsp_url_get_request_uri (info->url);
4013 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
4015 if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
4016 if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
4017 src->tls_validation_flags))
4018 GST_WARNING_OBJECT (src, "Unable to set TLS validation flags");
4020 if (src->tls_database)
4021 gst_rtsp_connection_set_tls_database (info->connection,
4025 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
4026 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
4028 if (src->proxy_host) {
4029 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
4031 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
4036 if (!info->connected) {
4039 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
4040 ("Connecting to %s", info->location));
4041 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
4043 gst_rtsp_connection_connect (info->connection,
4044 src->ptcp_timeout)) < 0)
4045 goto could_not_connect;
4047 info->connected = TRUE;
4054 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
4059 gchar *str = gst_rtsp_strresult (res);
4060 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
4066 gchar *str = gst_rtsp_strresult (res);
4067 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
4073 static GstRTSPResult
4074 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
4077 GST_RTSP_STATE_LOCK (src);
4078 if (info->connected) {
4079 GST_DEBUG_OBJECT (src, "closing connection...");
4080 gst_rtsp_connection_close (info->connection);
4081 info->connected = FALSE;
4083 if (free && info->connection) {
4084 /* free connection */
4085 GST_DEBUG_OBJECT (src, "freeing connection...");
4086 gst_rtsp_connection_free (info->connection);
4087 info->connection = NULL;
4089 GST_RTSP_STATE_UNLOCK (src);
4093 static GstRTSPResult
4094 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
4099 GST_DEBUG_OBJECT (src, "reconnecting connection...");
4100 gst_rtsp_conninfo_close (src, info, FALSE);
4101 res = gst_rtsp_conninfo_connect (src, info, async);
4107 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
4111 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
4112 GST_RTSP_STATE_LOCK (src);
4113 if (src->conninfo.connection && src->conninfo.flushing != flush) {
4114 GST_DEBUG_OBJECT (src, "connection flush");
4115 gst_rtsp_connection_flush (src->conninfo.connection, flush);
4116 src->conninfo.flushing = flush;
4118 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4119 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4120 if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
4121 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
4122 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
4123 stream->conninfo.flushing = flush;
4126 GST_RTSP_STATE_UNLOCK (src);
4129 /* FIXME, handle server request, reply with OK, for now */
4130 static GstRTSPResult
4131 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
4132 GstRTSPMessage * request)
4134 GstRTSPMessage response = { 0 };
4137 GST_DEBUG_OBJECT (src, "got server request message");
4140 gst_rtsp_message_dump (request);
4142 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
4144 if (res == GST_RTSP_ENOTIMPL) {
4145 /* default implementation, send OK */
4146 GST_DEBUG_OBJECT (src, "prepare OK reply");
4148 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
4153 /* let app parse and reply */
4154 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST],
4155 0, request, &response);
4158 gst_rtsp_message_dump (&response);
4160 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
4164 gst_rtsp_message_unset (&response);
4165 } else if (res == GST_RTSP_EEOF)
4173 gst_rtsp_message_unset (&response);
4178 /* send server keep-alive */
4179 static GstRTSPResult
4180 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
4182 GstRTSPMessage request = { 0 };
4184 GstRTSPMethod method;
4185 const gchar *control;
4187 if (src->do_rtsp_keep_alive == FALSE) {
4188 GST_DEBUG_OBJECT (src, "do-rtsp-keep-alive is FALSE, not sending.");
4189 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4193 GST_DEBUG_OBJECT (src, "creating server keep-alive");
4195 /* find a method to use for keep-alive */
4196 if (src->methods & GST_RTSP_GET_PARAMETER)
4197 method = GST_RTSP_GET_PARAMETER;
4199 method = GST_RTSP_OPTIONS;
4201 control = get_aggregate_control (src);
4202 if (control == NULL)
4205 res = gst_rtsp_message_init_request (&request, method, control);
4210 gst_rtsp_message_dump (&request);
4213 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
4218 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4219 gst_rtsp_message_unset (&request);
4226 GST_WARNING_OBJECT (src, "no control url to send keepalive");
4231 gchar *str = gst_rtsp_strresult (res);
4233 gst_rtsp_message_unset (&request);
4234 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
4235 ("Could not send keep-alive. (%s)", str));
4241 static GstFlowReturn
4242 gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
4244 GstFlowReturn ret = GST_FLOW_OK;
4246 GstRTSPStream *stream;
4247 GstPad *outpad = NULL;
4254 channel = message->type_data.data.channel;
4256 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
4258 goto unknown_stream;
4260 if (channel == stream->channel[0]) {
4261 outpad = stream->channelpad[0];
4263 } else if (channel == stream->channel[1]) {
4264 outpad = stream->channelpad[1];
4270 /* take a look at the body to figure out what we have */
4271 gst_rtsp_message_get_body (message, &data, &size);
4273 goto invalid_length;
4275 /* channels are not correct on some servers, do extra check */
4276 if (data[1] >= 200 && data[1] <= 204) {
4277 /* hmm RTCP message switch to the RTCP pad of the same stream. */
4278 outpad = stream->channelpad[1];
4282 /* we have no clue what this is, just ignore then. */
4284 goto unknown_stream;
4286 /* take the message body for further processing */
4287 gst_rtsp_message_steal_body (message, &data, &size);
4289 /* strip the trailing \0 */
4292 buf = gst_buffer_new ();
4293 gst_buffer_append_memory (buf,
4294 gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
4296 /* don't need message anymore */
4297 gst_rtsp_message_unset (message);
4299 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
4302 if (src->need_activate) {
4308 guint group_id = gst_util_group_id_next ();
4310 /* generate an SHA256 sum of the URI */
4311 cs = g_checksum_new (G_CHECKSUM_SHA256);
4312 uri = src->conninfo.location;
4313 g_checksum_update (cs, (const guchar *) uri, strlen (uri));
4315 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4316 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4319 g_strdup_printf ("%s/%d", g_checksum_get_string (cs), ostream->id);
4320 event = gst_event_new_stream_start (stream_id);
4321 gst_event_set_group_id (event, group_id);
4324 gst_rtspsrc_stream_push_event (src, ostream, event);
4326 g_checksum_free (cs);
4328 gst_rtspsrc_activate_streams (src);
4329 src->need_activate = FALSE;
4331 if ((event = src->start_segment) != NULL) {
4332 src->start_segment = NULL;
4333 gst_rtspsrc_push_event (src, event);
4336 if (src->base_time == -1) {
4337 /* Take current running_time. This timestamp will be put on
4338 * the first buffer of each stream because we are a live source and so we
4339 * timestamp with the running_time. When we are dealing with TCP, we also
4340 * only timestamp the first buffer (using the DISCONT flag) because a server
4341 * typically bursts data, for which we don't want to compensate by speeding
4342 * up the media. The other timestamps will be interpollated from this one
4343 * using the RTP timestamps. */
4344 GST_OBJECT_LOCK (src);
4345 if (GST_ELEMENT_CLOCK (src)) {
4347 GstClockTime base_time;
4349 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
4350 base_time = GST_ELEMENT_CAST (src)->base_time;
4352 src->base_time = now - base_time;
4354 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
4355 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
4357 GST_OBJECT_UNLOCK (src);
4360 if (stream->discont && !is_rtcp) {
4361 /* mark first RTP buffer as discont */
4362 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
4363 stream->discont = FALSE;
4364 /* first buffer gets the timestamp, other buffers are not timestamped and
4365 * their presentation time will be interpollated from the rtp timestamps. */
4366 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
4367 GST_TIME_ARGS (src->base_time));
4369 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
4372 /* chain to the peer pad */
4373 if (GST_PAD_IS_SINK (outpad))
4374 ret = gst_pad_chain (outpad, buf);
4376 ret = gst_pad_push (outpad, buf);
4379 /* combine all stream flows for the data transport */
4380 ret = gst_rtspsrc_combine_flows (src, stream, ret);
4387 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
4388 gst_rtsp_message_unset (message);
4393 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4394 ("Short message received, ignoring."));
4395 gst_rtsp_message_unset (message);
4400 static GstFlowReturn
4401 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
4403 GstRTSPMessage message = { 0 };
4405 GstFlowReturn ret = GST_FLOW_OK;
4406 GTimeVal tv_timeout;
4409 /* get the next timeout interval */
4410 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4412 /* see if the timeout period expired */
4413 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
4414 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
4415 /* send keep-alive, only act on interrupt, a warning will be posted for
4417 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4419 /* get new timeout */
4420 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4423 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
4424 tv_timeout.tv_sec, tv_timeout.tv_usec);
4426 /* protect the connection with the connection lock so that we can see when
4427 * we are finished doing server communication */
4429 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4430 &message, src->ptcp_timeout);
4434 GST_DEBUG_OBJECT (src, "we received a server message");
4436 case GST_RTSP_EINTR:
4437 /* we got interrupted this means we need to stop */
4439 case GST_RTSP_ETIMEOUT:
4440 /* no reply, send keep alive */
4441 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4442 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4446 /* go EOS when the server closed the connection */
4452 switch (message.type) {
4453 case GST_RTSP_MESSAGE_REQUEST:
4454 /* server sends us a request message, handle it */
4456 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4458 if (res == GST_RTSP_EEOF)
4461 goto handle_request_failed;
4463 case GST_RTSP_MESSAGE_RESPONSE:
4464 /* we ignore response messages */
4465 GST_DEBUG_OBJECT (src, "ignoring response message");
4467 gst_rtsp_message_dump (&message);
4469 case GST_RTSP_MESSAGE_DATA:
4470 GST_DEBUG_OBJECT (src, "got data message");
4471 ret = gst_rtspsrc_handle_data (src, &message);
4472 if (ret != GST_FLOW_OK)
4473 goto handle_data_failed;
4476 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4481 g_assert_not_reached ();
4486 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4487 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4488 ("The server closed the connection."));
4489 src->conninfo.connected = FALSE;
4490 gst_rtsp_message_unset (&message);
4491 return GST_FLOW_EOS;
4495 gst_rtsp_message_unset (&message);
4496 GST_DEBUG_OBJECT (src, "got interrupted");
4497 return GST_FLOW_FLUSHING;
4501 gchar *str = gst_rtsp_strresult (res);
4503 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4504 ("Could not receive message. (%s)", str));
4507 gst_rtsp_message_unset (&message);
4508 return GST_FLOW_ERROR;
4510 handle_request_failed:
4512 gchar *str = gst_rtsp_strresult (res);
4514 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4515 ("Could not handle server message. (%s)", str));
4517 gst_rtsp_message_unset (&message);
4518 return GST_FLOW_ERROR;
4522 GST_DEBUG_OBJECT (src, "could no handle data message");
4527 static GstFlowReturn
4528 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
4531 GstRTSPMessage message = { 0 };
4535 GTimeVal tv_timeout;
4537 /* get the next timeout interval */
4538 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4540 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
4541 (gint) tv_timeout.tv_sec);
4543 gst_rtsp_message_unset (&message);
4545 /* we should continue reading the TCP socket because the server might
4546 * send us requests. When the session timeout expires, we need to send a
4547 * keep-alive request to keep the session open. */
4548 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4549 &message, &tv_timeout);
4553 GST_DEBUG_OBJECT (src, "we received a server message");
4555 case GST_RTSP_EINTR:
4556 /* we got interrupted, see what we have to do */
4558 case GST_RTSP_ETIMEOUT:
4559 /* send keep-alive, ignore the result, a warning will be posted. */
4560 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4561 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4565 /* server closed the connection. not very fatal for UDP, reconnect and
4566 * see what happens. */
4567 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4568 ("The server closed the connection."));
4569 if (src->udp_reconnect) {
4571 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
4578 GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
4580 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4581 ("Unhandled return value %d.", res));
4585 switch (message.type) {
4586 case GST_RTSP_MESSAGE_REQUEST:
4587 /* server sends us a request message, handle it */
4589 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4591 if (res == GST_RTSP_EEOF)
4594 goto handle_request_failed;
4596 case GST_RTSP_MESSAGE_RESPONSE:
4597 /* we ignore response and data messages */
4598 GST_DEBUG_OBJECT (src, "ignoring response message");
4600 gst_rtsp_message_dump (&message);
4601 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
4602 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
4603 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
4604 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
4605 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4612 case GST_RTSP_MESSAGE_DATA:
4613 /* we ignore response and data messages */
4614 GST_DEBUG_OBJECT (src, "ignoring data message");
4617 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4622 g_assert_not_reached ();
4624 /* we get here when the connection got interrupted */
4627 gst_rtsp_message_unset (&message);
4628 GST_DEBUG_OBJECT (src, "got interrupted");
4629 return GST_FLOW_FLUSHING;
4633 gchar *str = gst_rtsp_strresult (res);
4636 src->conninfo.connected = FALSE;
4637 if (res != GST_RTSP_EINTR) {
4638 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
4639 ("Could not connect to server. (%s)", str));
4641 ret = GST_FLOW_ERROR;
4643 ret = GST_FLOW_FLUSHING;
4649 gchar *str = gst_rtsp_strresult (res);
4651 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4652 ("Could not receive message. (%s)", str));
4654 return GST_FLOW_ERROR;
4656 handle_request_failed:
4658 gchar *str = gst_rtsp_strresult (res);
4661 gst_rtsp_message_unset (&message);
4662 if (res != GST_RTSP_EINTR) {
4663 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4664 ("Could not handle server message. (%s)", str));
4666 ret = GST_FLOW_ERROR;
4668 ret = GST_FLOW_FLUSHING;
4674 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4675 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4676 ("The server closed the connection."));
4677 src->conninfo.connected = FALSE;
4678 gst_rtsp_message_unset (&message);
4679 return GST_FLOW_EOS;
4683 static GstRTSPResult
4684 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
4686 GstRTSPResult res = GST_RTSP_OK;
4689 GST_DEBUG_OBJECT (src, "doing reconnect");
4691 GST_OBJECT_LOCK (src);
4692 /* only restart when the pads were not yet activated, else we were
4693 * streaming over UDP */
4694 restart = src->need_activate;
4695 GST_OBJECT_UNLOCK (src);
4697 /* no need to restart, we're done */
4701 /* we can try only TCP now */
4702 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
4704 /* close and cleanup our state */
4705 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
4708 /* see if we have TCP left to try. Also don't try TCP when we were configured
4710 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
4713 /* We post a warning message now to inform the user
4714 * that nothing happened. It's most likely a firewall thing. */
4715 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4716 ("Could not receive any UDP packets for %.4f seconds, maybe your "
4717 "firewall is blocking it. Retrying using a TCP connection.",
4718 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
4720 /* open new connection using tcp */
4721 if (gst_rtspsrc_open (src, async) < 0)
4724 /* start playback */
4725 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
4734 src->cur_protocols = 0;
4735 /* no transport possible, post an error and stop */
4736 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4737 ("Could not receive any UDP packets for %.4f seconds, maybe your "
4738 "firewall is blocking it. No other protocols to try.",
4739 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
4740 return GST_RTSP_ERROR;
4744 GST_DEBUG_OBJECT (src, "open failed");
4749 GST_DEBUG_OBJECT (src, "play failed");
4755 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
4759 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
4762 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
4765 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
4768 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
4776 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
4780 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
4783 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
4786 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
4789 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
4797 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
4801 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
4804 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
4807 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
4810 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
4818 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
4822 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
4825 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
4828 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
4831 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
4839 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
4841 if (ret == GST_RTSP_OK)
4842 gst_rtspsrc_loop_complete_cmd (src, cmd);
4843 else if (ret == GST_RTSP_EINTR)
4844 gst_rtspsrc_loop_cancel_cmd (src, cmd);
4846 gst_rtspsrc_loop_error_cmd (src, cmd);
4850 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
4853 gboolean flushed = FALSE;
4855 /* start new request */
4856 gst_rtspsrc_loop_start_cmd (src, cmd);
4858 GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
4860 GST_OBJECT_LOCK (src);
4861 old = src->pending_cmd;
4862 if (old == CMD_RECONNECT) {
4863 GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
4864 cmd = CMD_RECONNECT;
4866 if (old != CMD_WAIT) {
4867 src->pending_cmd = CMD_WAIT;
4868 GST_OBJECT_UNLOCK (src);
4869 /* cancel previous request */
4870 GST_DEBUG_OBJECT (src, "cancel previous request %d", old);
4871 gst_rtspsrc_loop_cancel_cmd (src, old);
4872 GST_OBJECT_LOCK (src);
4874 src->pending_cmd = cmd;
4875 /* interrupt if allowed */
4876 if (src->busy_cmd & mask) {
4877 GST_DEBUG_OBJECT (src, "connection flush busy %d", src->busy_cmd);
4878 gst_rtspsrc_connection_flush (src, TRUE);
4881 GST_DEBUG_OBJECT (src, "not interrupting busy cmd %d", src->busy_cmd);
4884 gst_task_start (src->task);
4885 GST_OBJECT_UNLOCK (src);
4891 gst_rtspsrc_loop (GstRTSPSrc * src)
4895 if (!src->conninfo.connection || !src->conninfo.connected)
4898 if (src->interleaved)
4899 ret = gst_rtspsrc_loop_interleaved (src);
4901 ret = gst_rtspsrc_loop_udp (src);
4903 if (ret != GST_FLOW_OK)
4911 GST_WARNING_OBJECT (src, "we are not connected");
4912 ret = GST_FLOW_FLUSHING;
4917 const gchar *reason = gst_flow_get_name (ret);
4919 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
4920 src->running = FALSE;
4921 if (ret == GST_FLOW_EOS) {
4922 /* perform EOS logic */
4923 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
4924 gst_element_post_message (GST_ELEMENT_CAST (src),
4925 gst_message_new_segment_done (GST_OBJECT_CAST (src),
4926 src->segment.format, src->segment.position));
4927 gst_rtspsrc_push_event (src,
4928 gst_event_new_segment_done (src->segment.format,
4929 src->segment.position));
4931 gst_rtspsrc_push_event (src, gst_event_new_eos ());
4933 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
4934 /* for fatal errors we post an error message, post the error before the
4935 * EOS so the app knows about the error first. */
4936 GST_ELEMENT_ERROR (src, STREAM, FAILED,
4937 ("Internal data flow error."),
4938 ("streaming task paused, reason %s (%d)", reason, ret));
4939 gst_rtspsrc_push_event (src, gst_event_new_eos ());
4941 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_LOOP);
4946 #ifndef GST_DISABLE_GST_DEBUG
4947 static const gchar *
4948 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
4952 while (method != 0) {
4969 static const gchar *
4970 gst_rtspsrc_skip_lws (const gchar * s)
4972 while (g_ascii_isspace (*s))
4977 static const gchar *
4978 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
4980 while (s > start && g_ascii_isspace (*(s - 1)))
4985 static const gchar *
4986 gst_rtspsrc_skip_commas (const gchar * s)
4988 /* The grammar allows for multiple commas */
4989 while (g_ascii_isspace (*s) || *s == ',')
4994 static const gchar *
4995 gst_rtspsrc_skip_item (const gchar * s)
4997 gboolean quoted = FALSE;
4998 const gchar *start = s;
5000 /* A list item ends at the last non-whitespace character
5001 * before a comma which is not inside a quoted-string. Or at
5002 * the end of the string.
5008 if (*s == '\\' && *(s + 1))
5017 return gst_rtspsrc_unskip_lws (s, start);
5021 gst_rtsp_decode_quoted_string (gchar * quoted_string)
5025 src = quoted_string + 1;
5026 dst = quoted_string;
5027 while (*src && *src != '"') {
5028 if (*src == '\\' && *(src + 1))
5035 /* Extract the authentication tokens that the server provided for each method
5036 * into an array of structures and give those to the connection object.
5039 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
5040 const gchar * header, gboolean * stale)
5042 GSList *list = NULL, *iter;
5044 gchar *item, *eq, *name_end, *value;
5046 g_return_if_fail (stale != NULL);
5048 gst_rtsp_connection_clear_auth_params (conn);
5051 /* Parse a header whose content is described by RFC2616 as
5052 * "#something", where "something" does not itself contain commas,
5053 * except as part of quoted-strings, into a list of allocated strings.
5055 header = gst_rtspsrc_skip_commas (header);
5057 end = gst_rtspsrc_skip_item (header);
5058 list = g_slist_prepend (list, g_strndup (header, end - header));
5059 header = gst_rtspsrc_skip_commas (end);
5064 list = g_slist_reverse (list);
5065 for (iter = list; iter; iter = iter->next) {
5068 eq = strchr (item, '=');
5070 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
5071 if (name_end == item) {
5072 /* That's no good... */
5079 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
5081 gst_rtsp_decode_quoted_string (value);
5085 if (item && (strcmp (item, "stale") == 0) &&
5086 value && (strcmp (value, "TRUE") == 0))
5088 gst_rtsp_connection_set_auth_param (conn, item, value);
5092 g_slist_free (list);
5095 /* Parse a WWW-Authenticate Response header and determine the
5096 * available authentication methods
5098 * This code should also cope with the fact that each WWW-Authenticate
5099 * header can contain multiple challenge methods + tokens
5101 * At the moment, for Basic auth, we just do a minimal check and don't
5102 * even parse out the realm */
5104 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
5105 GstRTSPConnection * conn, gboolean * stale)
5109 g_return_if_fail (hdr != NULL);
5110 g_return_if_fail (methods != NULL);
5111 g_return_if_fail (stale != NULL);
5113 /* Skip whitespace at the start of the string */
5114 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
5116 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
5117 *methods |= GST_RTSP_AUTH_BASIC;
5118 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
5119 *methods |= GST_RTSP_AUTH_DIGEST;
5120 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
5125 * gst_rtspsrc_setup_auth:
5126 * @src: the rtsp source
5128 * Configure a username and password and auth method on the
5129 * connection object based on a response we received from the
5132 * Currently, this requires that a username and password were supplied
5133 * in the uri. In the future, they may be requested on demand by sending
5134 * a message up the bus.
5136 * Returns: TRUE if authentication information could be set up correctly.
5139 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
5143 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
5144 GstRTSPAuthMethod method;
5145 GstRTSPResult auth_result;
5147 GstRTSPConnection *conn;
5149 gboolean stale = FALSE;
5151 conn = src->conninfo.connection;
5153 /* Identify the available auth methods and see if any are supported */
5154 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
5155 &hdr, 0) == GST_RTSP_OK) {
5156 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
5159 if (avail_methods == GST_RTSP_AUTH_NONE)
5160 goto no_auth_available;
5162 /* For digest auth, if the response indicates that the session
5163 * data are stale, we just update them in the connection object and
5164 * return TRUE to retry the request */
5166 src->tried_url_auth = FALSE;
5168 url = gst_rtsp_connection_get_url (conn);
5170 /* Do we have username and password available? */
5171 if (url != NULL && !src->tried_url_auth && url->user != NULL
5172 && url->passwd != NULL) {
5175 src->tried_url_auth = TRUE;
5176 GST_DEBUG_OBJECT (src,
5177 "Attempting authentication using credentials from the URL");
5179 user = src->user_id;
5180 pass = src->user_pw;
5181 GST_DEBUG_OBJECT (src,
5182 "Attempting authentication using credentials from the properties");
5185 /* FIXME: If the url didn't contain username and password or we tried them
5186 * already, request a username and passwd from the application via some kind
5187 * of credentials request message */
5189 /* If we don't have a username and passwd at this point, bail out. */
5190 if (user == NULL || pass == NULL)
5193 /* Try to configure for each available authentication method, strongest to
5195 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
5196 /* Check if this method is available on the server */
5197 if ((method & avail_methods) == 0)
5200 /* Pass the credentials to the connection to try on the next request */
5201 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
5202 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
5203 * ignore it and end up retrying later */
5204 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
5205 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
5206 gst_rtsp_auth_method_to_string (method));
5211 if (method == GST_RTSP_AUTH_NONE)
5212 goto no_auth_available;
5218 /* Output an error indicating that we couldn't connect because there were
5219 * no supported authentication protocols */
5220 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5221 ("No supported authentication protocol was found"));
5226 /* We don't fire an error message, we just return FALSE and let the
5227 * normal NOT_AUTHORIZED error be propagated */
5232 static GstRTSPResult
5233 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5234 GstRTSPMessage * request, GstRTSPMessage * response,
5235 GstRTSPStatusCode * code)
5238 GstRTSPStatusCode thecode;
5239 gchar *content_base = NULL;
5243 if (!src->short_header)
5244 gst_rtsp_ext_list_before_send (src->extensions, request);
5246 GST_DEBUG_OBJECT (src, "sending message");
5249 gst_rtsp_message_dump (request);
5251 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
5255 gst_rtsp_connection_reset_timeout (conn);
5258 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
5263 gst_rtsp_message_dump (response);
5265 switch (response->type) {
5266 case GST_RTSP_MESSAGE_REQUEST:
5267 res = gst_rtspsrc_handle_request (src, conn, response);
5268 if (res == GST_RTSP_EEOF)
5271 goto handle_request_failed;
5273 case GST_RTSP_MESSAGE_RESPONSE:
5274 /* ok, a response is good */
5275 GST_DEBUG_OBJECT (src, "received response message");
5277 case GST_RTSP_MESSAGE_DATA:
5278 /* get next response */
5279 GST_DEBUG_OBJECT (src, "handle data response message");
5280 gst_rtspsrc_handle_data (src, response);
5283 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
5288 thecode = response->type_data.response.code;
5290 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
5292 /* if the caller wanted the result code, we store it. */
5296 /* If the request didn't succeed, bail out before doing any more */
5297 if (thecode != GST_RTSP_STS_OK)
5300 /* store new content base if any */
5301 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
5304 g_free (src->content_base);
5305 src->content_base = g_strdup (content_base);
5307 gst_rtsp_ext_list_after_send (src->extensions, request, response);
5314 gchar *str = gst_rtsp_strresult (res);
5316 if (res != GST_RTSP_EINTR) {
5317 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5318 ("Could not send message. (%s)", str));
5320 GST_WARNING_OBJECT (src, "send interrupted");
5329 GST_WARNING_OBJECT (src, "server closed connection");
5330 if ((try == 0) && !src->interleaved && src->udp_reconnect) {
5332 /* if reconnect succeeds, try again */
5334 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
5338 /* only try once after reconnect, then fallthrough and error out */
5341 gchar *str = gst_rtsp_strresult (res);
5343 if (res != GST_RTSP_EINTR) {
5344 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5345 ("Could not receive message. (%s)", str));
5347 GST_WARNING_OBJECT (src, "receive interrupted");
5355 handle_request_failed:
5357 /* ERROR was posted */
5358 gst_rtsp_message_unset (response);
5363 GST_DEBUG_OBJECT (src, "we got an eof from the server");
5364 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5365 ("The server closed the connection."));
5366 gst_rtsp_message_unset (response);
5373 * @src: the rtsp source
5374 * @conn: the connection to send on
5375 * @request: must point to a valid request
5376 * @response: must point to an empty #GstRTSPMessage
5377 * @code: an optional code result
5379 * send @request and retrieve the response in @response. optionally @code can be
5380 * non-NULL in which case it will contain the status code of the response.
5382 * If This function returns #GST_RTSP_OK, @response will contain a valid response
5383 * message that should be cleaned with gst_rtsp_message_unset() after usage.
5385 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
5386 * @response message) if the response code was not 200 (OK).
5388 * If the attempt results in an authentication failure, then this will attempt
5389 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
5392 * Returns: #GST_RTSP_OK if the processing was successful.
5394 static GstRTSPResult
5395 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5396 GstRTSPMessage * request, GstRTSPMessage * response,
5397 GstRTSPStatusCode * code)
5399 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
5400 GstRTSPResult res = GST_RTSP_ERROR;
5403 GstRTSPMethod method = GST_RTSP_INVALID;
5409 /* make sure we don't loop forever */
5413 /* save method so we can disable it when the server complains */
5414 method = request->type_data.request.method;
5417 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
5421 case GST_RTSP_STS_UNAUTHORIZED:
5422 if (gst_rtspsrc_setup_auth (src, response)) {
5423 /* Try the request/response again after configuring the auth info
5431 } while (retry == TRUE);
5433 /* If the user requested the code, let them handle errors, otherwise
5434 * post an error below */
5437 else if (int_code != GST_RTSP_STS_OK)
5438 goto error_response;
5445 GST_DEBUG_OBJECT (src, "got error %d", res);
5450 res = GST_RTSP_ERROR;
5452 switch (response->type_data.response.code) {
5453 case GST_RTSP_STS_NOT_FOUND:
5454 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
5455 response->type_data.response.reason));
5457 case GST_RTSP_STS_MOVED_PERMANENTLY:
5458 case GST_RTSP_STS_MOVE_TEMPORARILY:
5460 gchar *new_location;
5461 GstRTSPLowerTrans transports;
5463 GST_DEBUG_OBJECT (src, "got redirection");
5464 /* if we don't have a Location Header, we must error */
5465 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
5466 &new_location, 0) < 0)
5469 /* When we receive a redirect result, we go back to the INIT state after
5470 * parsing the new URI. The caller should do the needed steps to issue
5471 * a new setup when it detects this state change. */
5472 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
5474 /* save current transports */
5475 if (src->conninfo.url)
5476 transports = src->conninfo.url->transports;
5478 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
5480 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
5482 /* set old transports */
5483 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
5484 src->conninfo.url->transports = transports;
5486 src->need_redirect = TRUE;
5487 src->state = GST_RTSP_STATE_INIT;
5491 case GST_RTSP_STS_NOT_ACCEPTABLE:
5492 case GST_RTSP_STS_NOT_IMPLEMENTED:
5493 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
5494 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
5495 gst_rtsp_method_as_text (method));
5496 src->methods &= ~method;
5500 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5501 ("Got error response: %d (%s).", response->type_data.response.code,
5502 response->type_data.response.reason));
5505 /* if we return ERROR we should unset the response ourselves */
5506 if (res == GST_RTSP_ERROR)
5507 gst_rtsp_message_unset (response);
5513 static GstRTSPResult
5514 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
5515 GstRTSPMessage * response, GstRTSPSrc * src)
5517 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
5522 /* parse the response and collect all the supported methods. We need this
5523 * information so that we don't try to send an unsupported request to the
5527 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
5529 GstRTSPHeaderField field;
5533 /* reset supported methods */
5536 /* Try Allow Header first */
5537 field = GST_RTSP_HDR_ALLOW;
5540 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5541 if (indx == 0 && !respoptions) {
5542 /* if no Allow header was found then try the Public header... */
5543 field = GST_RTSP_HDR_PUBLIC;
5544 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5549 src->methods |= gst_rtsp_options_from_text (respoptions);
5554 if (src->methods == 0) {
5555 /* neither Allow nor Public are required, assume the server supports
5556 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
5558 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
5559 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
5561 /* always assume PLAY, FIXME, extensions should be able to override
5563 src->methods |= GST_RTSP_PLAY;
5564 /* also assume it will support Range */
5565 src->seekable = TRUE;
5567 /* we need describe and setup */
5568 if (!(src->methods & GST_RTSP_DESCRIBE))
5570 if (!(src->methods & GST_RTSP_SETUP))
5578 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5579 ("Server does not support DESCRIBE."));
5584 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5585 ("Server does not support SETUP."));
5590 /* masks to be kept in sync with the hardcoded protocol order of preference
5592 static guint protocol_masks[] = {
5593 GST_RTSP_LOWER_TRANS_UDP,
5594 GST_RTSP_LOWER_TRANS_UDP_MCAST,
5595 GST_RTSP_LOWER_TRANS_TCP,
5599 static GstRTSPResult
5600 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
5601 GstRTSPLowerTrans protocols, GstRTSPProfile profile, gchar ** transports)
5605 gboolean add_udp_str;
5610 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
5615 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
5617 /* extension listed transports, use those */
5618 if (*transports != NULL)
5621 /* it's the default */
5622 add_udp_str = FALSE;
5624 /* the default RTSP transports */
5625 result = g_string_new ("RTP");
5628 case GST_RTSP_PROFILE_AVP:
5629 g_string_append (result, "/AVP");
5631 case GST_RTSP_PROFILE_SAVP:
5632 g_string_append (result, "/SAVP");
5634 case GST_RTSP_PROFILE_AVPF:
5635 g_string_append (result, "/AVPF");
5637 case GST_RTSP_PROFILE_SAVPF:
5638 g_string_append (result, "/SAVPF");
5644 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
5645 GST_DEBUG_OBJECT (src, "adding UDP unicast");
5647 g_string_append (result, "/UDP");
5648 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
5649 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
5650 GST_DEBUG_OBJECT (src, "adding UDP multicast");
5651 /* we don't have to allocate any UDP ports yet, if the selected transport
5652 * turns out to be multicast we can create them and join the multicast
5653 * group indicated in the transport reply */
5655 g_string_append (result, "/UDP");
5656 g_string_append (result, ";multicast");
5657 if (src->next_port_num != 0) {
5658 if (src->client_port_range.max > 0 &&
5659 src->next_port_num >= src->client_port_range.max)
5662 g_string_append_printf (result, ";client_port=%d-%d",
5663 src->next_port_num, src->next_port_num + 1);
5665 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
5666 GST_DEBUG_OBJECT (src, "adding TCP");
5668 g_string_append (result, "/TCP;unicast;interleaved=%%i1-%%i2");
5670 *transports = g_string_free (result, FALSE);
5672 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
5679 GST_ERROR ("extension gave error %d", res);
5684 GST_ERROR ("no more ports available");
5685 return GST_RTSP_ERROR;
5689 static GstRTSPResult
5690 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
5691 gint orig_rtpport, gint orig_rtcpport)
5694 gint nr_udp, nr_int;
5696 gint rtpport = 0, rtcpport = 0;
5699 src = stream->parent;
5701 /* find number of placeholders first */
5702 if (strstr (*transports, "%%i2"))
5704 else if (strstr (*transports, "%%i1"))
5709 if (strstr (*transports, "%%u2"))
5711 else if (strstr (*transports, "%%u1"))
5716 if (nr_udp == 0 && nr_int == 0)
5720 if (!orig_rtpport || !orig_rtcpport) {
5721 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
5724 rtpport = orig_rtpport;
5725 rtcpport = orig_rtcpport;
5729 str = g_string_new ("");
5731 while ((next = strstr (p, "%%"))) {
5732 g_string_append_len (str, p, next - p);
5733 if (next[2] == 'u') {
5735 g_string_append_printf (str, "%d", rtpport);
5736 else if (next[3] == '2')
5737 g_string_append_printf (str, "%d", rtcpport);
5739 if (next[2] == 'i') {
5741 g_string_append_printf (str, "%d", src->free_channel);
5742 else if (next[3] == '2')
5743 g_string_append_printf (str, "%d", src->free_channel + 1);
5748 /* append final part */
5749 g_string_append (str, p);
5751 g_free (*transports);
5752 *transports = g_string_free (str, FALSE);
5760 GST_ERROR ("failed to allocate udp ports");
5761 return GST_RTSP_ERROR;
5765 /* Perform the SETUP request for all the streams.
5767 * We ask the server for a specific transport, which initially includes all the
5768 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
5769 * two local UDP ports that we send to the server.
5771 * Once the server replied with a transport, we configure the other streams
5772 * with the same transport.
5774 * This function will also configure the stream for the selected transport,
5775 * which basically means creating the pipeline.
5777 static GstRTSPResult
5778 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
5781 GstRTSPResult res = GST_RTSP_ERROR;
5782 GstRTSPMessage request = { 0 };
5783 GstRTSPMessage response = { 0 };
5784 GstRTSPStream *stream = NULL;
5785 GstRTSPLowerTrans protocols;
5786 GstRTSPStatusCode code;
5787 gboolean unsupported_real = FALSE;
5788 gint rtpport, rtcpport;
5792 if (src->conninfo.connection) {
5793 url = gst_rtsp_connection_get_url (src->conninfo.connection);
5794 /* we initially allow all configured lower transports. based on the URL
5795 * transports and the replies from the server we narrow them down. */
5796 protocols = url->transports & src->cur_protocols;
5799 protocols = src->cur_protocols;
5805 /* reset some state */
5806 src->free_channel = 0;
5807 src->interleaved = FALSE;
5808 src->need_activate = FALSE;
5809 /* keep track of next port number, 0 is random */
5810 src->next_port_num = src->client_port_range.min;
5811 rtpport = rtcpport = 0;
5813 if (G_UNLIKELY (src->streams == NULL))
5816 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5817 GstRTSPConnection *conn;
5824 stream = (GstRTSPStream *) walk->data;
5826 caps = stream_get_caps_for_pt (stream, stream->default_pt);
5828 GST_DEBUG_OBJECT (src, "skipping stream %p, no caps", stream);
5832 if (stream->skipped) {
5833 GST_DEBUG_OBJECT (src, "skipping stream %p", stream);
5837 /* see if we need to configure this stream */
5838 if (!gst_rtsp_ext_list_configure_stream (src->extensions, caps)) {
5839 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
5844 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_SELECT_STREAM], 0,
5845 stream->id, caps, &selected);
5847 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by signal", stream);
5851 /* merge/overwrite global caps */
5856 s = gst_caps_get_structure (caps, 0);
5858 num = gst_structure_n_fields (src->props);
5859 for (j = 0; j < num; j++) {
5863 name = gst_structure_nth_field_name (src->props, j);
5864 val = gst_structure_get_value (src->props, name);
5865 gst_structure_set_value (s, name, val);
5867 GST_DEBUG_OBJECT (src, "copied %s", name);
5871 /* skip setup if we have no URL for it */
5872 if (stream->conninfo.location == NULL) {
5873 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
5877 if (src->conninfo.connection == NULL) {
5878 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
5879 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
5882 conn = stream->conninfo.connection;
5884 conn = src->conninfo.connection;
5886 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
5887 stream->conninfo.location);
5889 /* if we have a multicast connection, only suggest multicast from now on */
5890 if (stream->is_multicast)
5891 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
5894 /* first selectable protocol */
5895 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5897 if (!protocol_masks[mask])
5901 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
5902 protocol_masks[mask]);
5903 /* create a string with first transport in line */
5905 res = gst_rtspsrc_create_transports_string (src,
5906 protocols & protocol_masks[mask], stream->profile, &transports);
5907 if (res < 0 || transports == NULL)
5908 goto setup_transport_failed;
5910 if (strlen (transports) == 0) {
5911 g_free (transports);
5912 GST_DEBUG_OBJECT (src, "no transports found");
5917 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
5919 /* replace placeholders with real values, this function will optionally
5920 * allocate UDP ports and other info needed to execute the setup request */
5921 res = gst_rtspsrc_prepare_transports (stream, &transports,
5922 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
5924 g_free (transports);
5925 goto setup_transport_failed;
5928 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
5930 /* create SETUP request */
5932 gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
5933 stream->conninfo.location);
5935 g_free (transports);
5936 goto create_request_failed;
5939 /* select transport */
5940 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
5942 /* if the user wants a non default RTP packet size we add the blocksize
5944 if (src->rtp_blocksize > 0) {
5945 hval = g_strdup_printf ("%d", src->rtp_blocksize);
5946 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
5950 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
5953 /* handle the code ourselves */
5954 if ((res = gst_rtspsrc_send (src, conn, &request, &response, &code) < 0))
5958 case GST_RTSP_STS_OK:
5960 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
5961 gst_rtsp_message_unset (&request);
5962 gst_rtsp_message_unset (&response);
5963 /* cleanup of leftover transport */
5964 gst_rtspsrc_stream_free_udp (stream);
5965 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
5966 * we might be in this case */
5967 if (stream->container && rtpport && rtcpport && !retry) {
5968 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
5973 /* this transport did not go down well, but we may have others to try
5974 * that we did not send yet, try those and only give up then
5975 * but not without checking for lost cause/extension so we can
5976 * post a nicer/more useful error message later */
5977 if (!unsupported_real)
5978 unsupported_real = stream->is_real;
5979 /* select next available protocol, give up on this stream if none */
5981 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5983 if (!protocol_masks[mask] || unsupported_real)
5988 /* cleanup of leftover transport and move to the next stream */
5989 gst_rtspsrc_stream_free_udp (stream);
5990 goto response_error;
5993 /* parse response transport */
5995 gchar *resptrans = NULL;
5996 GstRTSPTransport transport = { 0 };
5998 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
6001 gst_rtspsrc_stream_free_udp (stream);
6005 /* parse transport, go to next stream on parse error */
6006 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
6007 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
6011 /* update allowed transports for other streams. once the transport of
6012 * one stream has been determined, we make sure that all other streams
6013 * are configured in the same way */
6014 switch (transport.lower_transport) {
6015 case GST_RTSP_LOWER_TRANS_TCP:
6016 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
6017 protocols = GST_RTSP_LOWER_TRANS_TCP;
6018 src->interleaved = TRUE;
6019 /* update free channels */
6021 MAX (transport.interleaved.min, src->free_channel);
6023 MAX (transport.interleaved.max, src->free_channel);
6024 src->free_channel++;
6026 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
6027 /* only allow multicast for other streams */
6028 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
6029 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
6030 /* if the server selected our ports, increment our counters so that
6031 * we select a new port later */
6032 if (src->next_port_num == transport.port.min &&
6033 src->next_port_num + 1 == transport.port.max) {
6034 src->next_port_num += 2;
6037 case GST_RTSP_LOWER_TRANS_UDP:
6038 /* only allow unicast for other streams */
6039 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
6040 protocols = GST_RTSP_LOWER_TRANS_UDP;
6043 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
6044 transport.lower_transport);
6048 if (!stream->container || (!src->interleaved && !retry)) {
6049 /* now configure the stream with the selected transport */
6050 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
6051 GST_DEBUG_OBJECT (src,
6052 "could not configure stream %p transport, skipping stream",
6055 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
6056 /* retain the first allocated UDP port pair */
6057 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
6058 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
6061 /* we need to activate at least one streams when we detect activity */
6062 src->need_activate = TRUE;
6064 /* stream is setup now */
6065 stream->setup = TRUE;
6070 GstRTSPStream *sskip;
6072 skip = g_list_next (skip);
6076 sskip = (GstRTSPStream *) skip->data;
6078 /* skip all streams with the same control url */
6079 if (g_str_equal (stream->conninfo.location, sskip->conninfo.location)) {
6080 GST_DEBUG_OBJECT (src, "found stream %p with same control %s",
6081 sskip, sskip->conninfo.location);
6082 sskip->skipped = TRUE;
6087 /* clean up our transport struct */
6088 gst_rtsp_transport_init (&transport);
6089 /* clean up used RTSP messages */
6090 gst_rtsp_message_unset (&request);
6091 gst_rtsp_message_unset (&response);
6095 /* store the transport protocol that was configured */
6096 src->cur_protocols = protocols;
6098 gst_rtsp_ext_list_stream_select (src->extensions, url);
6100 /* if there is nothing to activate, error out */
6101 if (!src->need_activate)
6102 goto nothing_to_activate;
6109 /* no transport possible, post an error and stop */
6110 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
6111 ("Could not connect to server, no protocols left"));
6112 return GST_RTSP_ERROR;
6116 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6117 ("SDP contains no streams"));
6118 return GST_RTSP_ERROR;
6120 create_request_failed:
6122 gchar *str = gst_rtsp_strresult (res);
6124 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6125 ("Could not create request. (%s)", str));
6129 setup_transport_failed:
6131 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6132 ("Could not setup transport."));
6133 res = GST_RTSP_ERROR;
6138 const gchar *str = gst_rtsp_status_as_text (code);
6140 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6141 ("Error (%d): %s", code, GST_STR_NULL (str)));
6142 res = GST_RTSP_ERROR;
6147 gchar *str = gst_rtsp_strresult (res);
6149 if (res != GST_RTSP_EINTR) {
6150 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6151 ("Could not send message. (%s)", str));
6153 GST_WARNING_OBJECT (src, "send interrupted");
6160 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6161 ("Server did not select transport."));
6162 res = GST_RTSP_ERROR;
6165 nothing_to_activate:
6167 /* none of the available error codes is really right .. */
6168 if (unsupported_real) {
6169 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6170 (_("No supported stream was found. You might need to install a "
6171 "GStreamer RTSP extension plugin for Real media streams.")),
6174 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6175 (_("No supported stream was found. You might need to allow "
6176 "more transport protocols or may otherwise be missing "
6177 "the right GStreamer RTSP extension plugin.")), (NULL));
6179 return GST_RTSP_ERROR;
6183 gst_rtsp_message_unset (&request);
6184 gst_rtsp_message_unset (&response);
6190 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
6191 GstSegment * segment)
6194 GstRTSPTimeRange *therange;
6197 gst_rtsp_range_free (src->range);
6199 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
6200 GST_DEBUG_OBJECT (src, "parsed range %s", range);
6201 src->range = therange;
6203 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
6205 gst_segment_init (segment, GST_FORMAT_TIME);
6209 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
6210 therange->min.type, therange->min.seconds, therange->max.type,
6211 therange->max.seconds);
6213 if (therange->min.type == GST_RTSP_TIME_NOW)
6215 else if (therange->min.type == GST_RTSP_TIME_END)
6218 seconds = therange->min.seconds * GST_SECOND;
6220 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
6221 GST_TIME_ARGS (seconds));
6223 /* we need to start playback without clipping from the position reported by
6225 segment->start = seconds;
6226 segment->position = seconds;
6228 if (therange->max.type == GST_RTSP_TIME_NOW)
6230 else if (therange->max.type == GST_RTSP_TIME_END)
6233 seconds = therange->max.seconds * GST_SECOND;
6235 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
6236 GST_TIME_ARGS (seconds));
6238 /* live (WMS) server might send overflowed large max as its idea of infinity,
6239 * compensate to prevent problems later on */
6240 if (seconds != -1 && seconds < 0) {
6242 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
6245 /* live (WMS) might send min == max, which is not worth recording */
6246 if (segment->duration == -1 && seconds == segment->start)
6249 /* don't change duration with unknown value, we might have a valid value
6250 * there that we want to keep. */
6252 segment->duration = seconds;
6257 /* Parse clock profived by the server with following syntax:
6259 * "GstNetTimeProvider <wrapped-clock> <server-IP:port> <clock-time>"
6262 gst_rtspsrc_parse_gst_clock (GstRTSPSrc * src, const gchar * gstclock)
6264 gboolean res = FALSE;
6266 if (g_str_has_prefix (gstclock, "GstNetTimeProvider ")) {
6267 gchar **fields = NULL, **parts = NULL;
6268 gchar *remote_ip, *str;
6270 GstClockTime base_time;
6273 fields = g_strsplit (gstclock, " ", 0);
6275 /* wrapped clock, not very interesting for now */
6276 if (fields[1] == NULL)
6279 /* remote IP address and port */
6280 if ((str = fields[2]) == NULL)
6283 parts = g_strsplit (str, ":", 0);
6285 if ((remote_ip = parts[0]) == NULL)
6288 if ((str = parts[1]) == NULL)
6296 if ((str = fields[3]) == NULL)
6299 base_time = g_ascii_strtoull (str, NULL, 10);
6302 gst_net_client_clock_new ((gchar *) "GstRTSPClock", remote_ip, port,
6305 if (src->provided_clock)
6306 gst_object_unref (src->provided_clock);
6307 src->provided_clock = netclock;
6309 gst_element_post_message (GST_ELEMENT_CAST (src),
6310 gst_message_new_clock_provide (GST_OBJECT_CAST (src),
6311 src->provided_clock, TRUE));
6315 g_strfreev (fields);
6321 /* must be called with the RTSP state lock */
6322 static GstRTSPResult
6323 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
6329 /* prepare global stream caps properties */
6331 gst_structure_remove_all_fields (src->props);
6333 src->props = gst_structure_new_empty ("RTSPProperties");
6336 gst_sdp_message_dump (sdp);
6338 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
6340 /* let the app inspect and change the SDP */
6341 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_ON_SDP], 0, sdp);
6343 gst_segment_init (&src->segment, GST_FORMAT_TIME);
6345 /* parse range for duration reporting. */
6350 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
6354 /* keep track of the range and configure it in the segment */
6355 if (gst_rtspsrc_parse_range (src, range, &src->segment))
6359 /* parse clock information. This is GStreamer specific, a server can tell the
6360 * client what clock it is using and wrap that in a network clock. The
6361 * advantage of that is that we can slave to it. */
6363 const gchar *gstclock;
6366 gstclock = gst_sdp_message_get_attribute_val_n (sdp, "x-gst-clock", i);
6367 if (gstclock == NULL)
6370 /* parse the clock and expose it in the provide_clock method */
6371 if (gst_rtspsrc_parse_gst_clock (src, gstclock))
6375 /* try to find a global control attribute. Note that a '*' means that we should
6376 * do aggregate control with the current url (so we don't do anything and
6377 * leave the current connection as is) */
6379 const gchar *control;
6382 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
6383 if (control == NULL)
6386 /* only take fully qualified urls */
6387 if (g_str_has_prefix (control, "rtsp://"))
6391 g_free (src->conninfo.location);
6392 src->conninfo.location = g_strdup (control);
6393 /* make a connection for this, if there was a connection already, nothing
6395 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
6396 GST_ERROR_OBJECT (src, "could not connect");
6399 /* we need to keep the control url separate from the connection url because
6400 * the rules for constructing the media control url need it */
6401 g_free (src->control);
6402 src->control = g_strdup (control);
6405 /* create streams */
6406 n_streams = gst_sdp_message_medias_len (sdp);
6407 for (i = 0; i < n_streams; i++) {
6408 gst_rtspsrc_create_stream (src, sdp, i);
6411 src->state = GST_RTSP_STATE_INIT;
6414 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
6417 /* reset our state */
6418 src->need_range = TRUE;
6421 src->state = GST_RTSP_STATE_READY;
6428 GST_ERROR_OBJECT (src, "setup failed");
6429 gst_rtspsrc_cleanup (src);
6434 static GstRTSPResult
6435 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
6439 GstRTSPMessage request = { 0 };
6440 GstRTSPMessage response = { 0 };
6443 gchar *respcont = NULL;
6446 src->need_redirect = FALSE;
6448 /* can't continue without a valid url */
6449 if (G_UNLIKELY (src->conninfo.url == NULL)) {
6450 res = GST_RTSP_EINVAL;
6453 src->tried_url_auth = FALSE;
6455 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
6456 goto connect_failed;
6458 /* create OPTIONS */
6459 GST_DEBUG_OBJECT (src, "create options...");
6461 gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
6462 src->conninfo.url_str);
6464 goto create_request_failed;
6467 GST_DEBUG_OBJECT (src, "send options...");
6470 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
6473 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6478 if (!gst_rtspsrc_parse_methods (src, &response))
6481 /* create DESCRIBE */
6482 GST_DEBUG_OBJECT (src, "create describe...");
6484 gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
6485 src->conninfo.url_str);
6487 goto create_request_failed;
6489 /* we only accept SDP for now */
6490 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
6494 GST_DEBUG_OBJECT (src, "send describe...");
6497 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
6500 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6504 /* we only perform redirect for the describe, currently */
6505 if (src->need_redirect) {
6506 /* close connection, we don't have to send a TEARDOWN yet, ignore the
6508 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6510 gst_rtsp_message_unset (&request);
6511 gst_rtsp_message_unset (&response);
6517 /* it could be that the DESCRIBE method was not implemented */
6518 if (!src->methods & GST_RTSP_DESCRIBE)
6521 /* check if reply is SDP */
6522 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
6524 /* could not be set but since the request returned OK, we assume it
6525 * was SDP, else check it. */
6527 if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
6528 goto wrong_content_type;
6531 /* get message body and parse as SDP */
6532 gst_rtsp_message_get_body (&response, &data, &size);
6533 if (data == NULL || size == 0)
6536 GST_DEBUG_OBJECT (src, "parse SDP...");
6537 gst_sdp_message_new (sdp);
6538 gst_sdp_message_parse_buffer (data, size, *sdp);
6540 /* clean up any messages */
6541 gst_rtsp_message_unset (&request);
6542 gst_rtsp_message_unset (&response);
6549 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
6550 ("No valid RTSP URL was provided"));
6555 gchar *str = gst_rtsp_strresult (res);
6557 if (res != GST_RTSP_EINTR) {
6558 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
6559 ("Failed to connect. (%s)", str));
6561 GST_WARNING_OBJECT (src, "connect interrupted");
6566 create_request_failed:
6568 gchar *str = gst_rtsp_strresult (res);
6570 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6571 ("Could not create request. (%s)", str));
6577 /* Don't post a message - the rtsp_send method will have
6578 * taken care of it because we passed NULL for the response code */
6583 /* error was posted */
6584 res = GST_RTSP_ERROR;
6589 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6590 ("Server does not support SDP, got %s.", respcont));
6591 res = GST_RTSP_ERROR;
6596 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6597 ("Server can not provide an SDP."));
6598 res = GST_RTSP_ERROR;
6603 if (src->conninfo.connection) {
6604 GST_DEBUG_OBJECT (src, "free connection");
6605 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6607 gst_rtsp_message_unset (&request);
6608 gst_rtsp_message_unset (&response);
6613 static GstRTSPResult
6614 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
6619 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
6621 if (src->sdp == NULL) {
6622 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
6626 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
6631 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
6638 GST_WARNING_OBJECT (src, "can't get sdp");
6639 src->open_error = TRUE;
6644 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
6645 src->open_error = TRUE;
6650 static GstRTSPResult
6651 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
6653 GstRTSPMessage request = { 0 };
6654 GstRTSPMessage response = { 0 };
6655 GstRTSPResult res = GST_RTSP_OK;
6657 const gchar *control;
6659 GST_DEBUG_OBJECT (src, "TEARDOWN...");
6661 gst_rtspsrc_set_state (src, GST_STATE_READY);
6663 if (src->state < GST_RTSP_STATE_READY) {
6664 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
6671 /* construct a control url */
6672 control = get_aggregate_control (src);
6674 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
6677 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6678 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6679 const gchar *setup_url;
6680 GstRTSPConnInfo *info;
6682 /* try aggregate control first but do non-aggregate control otherwise */
6684 setup_url = control;
6685 else if ((setup_url = stream->conninfo.location) == NULL)
6688 if (src->conninfo.connection) {
6689 info = &src->conninfo;
6690 } else if (stream->conninfo.connection) {
6691 info = &stream->conninfo;
6695 if (!info->connected)
6700 gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
6702 goto create_request_failed;
6705 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
6708 gst_rtspsrc_send (src, info->connection, &request, &response,
6712 /* FIXME, parse result? */
6713 gst_rtsp_message_unset (&request);
6714 gst_rtsp_message_unset (&response);
6717 /* early exit when we did aggregate control */
6723 /* close connections */
6724 GST_DEBUG_OBJECT (src, "closing connection...");
6725 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6726 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6727 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6728 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
6732 gst_rtspsrc_cleanup (src);
6734 src->state = GST_RTSP_STATE_INVALID;
6737 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
6742 create_request_failed:
6744 gchar *str = gst_rtsp_strresult (res);
6746 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6747 ("Could not create request. (%s)", str));
6753 gchar *str = gst_rtsp_strresult (res);
6755 gst_rtsp_message_unset (&request);
6756 if (res != GST_RTSP_EINTR) {
6757 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6758 ("Could not send message. (%s)", str));
6760 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
6767 GST_DEBUG_OBJECT (src,
6768 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
6773 /* RTP-Info is of the format:
6775 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
6777 * rtptime corresponds to the timestamp for the NPT time given in the header
6778 * seqbase corresponds to the next sequence number we received. This number
6779 * indicates the first seqnum after the seek and should be used to discard
6780 * packets that are from before the seek.
6783 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
6788 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
6790 infos = g_strsplit (rtpinfo, ",", 0);
6791 for (i = 0; infos[i]; i++) {
6793 GstRTSPStream *stream;
6797 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
6799 /* init values, types of seqbase and timebase are bigger than needed so we
6800 * can store -1 as uninitialized values */
6805 /* parse url, find stream for url.
6806 * parse seq and rtptime. The seq number should be configured in the rtp
6807 * depayloader or session manager to detect gaps. Same for the rtptime, it
6808 * should be used to create an initial time newsegment. */
6809 fields = g_strsplit (infos[i], ";", 0);
6810 for (j = 0; fields[j]; j++) {
6811 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
6812 /* remove leading whitespace */
6813 fields[j] = g_strchug (fields[j]);
6814 if (g_str_has_prefix (fields[j], "url=")) {
6815 /* get the url and the stream */
6817 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
6818 } else if (g_str_has_prefix (fields[j], "seq=")) {
6819 seqbase = atoi (fields[j] + 4);
6820 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
6821 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
6824 g_strfreev (fields);
6825 /* now we need to store the values for the caps of the stream */
6826 if (stream != NULL) {
6827 GST_DEBUG_OBJECT (src,
6828 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
6829 stream, seqbase, timebase);
6831 /* we have a stream, configure detected params */
6832 stream->seqbase = seqbase;
6833 stream->timebase = timebase;
6842 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
6847 interval = strtoul (rtcp, NULL, 10);
6848 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
6853 interval *= GST_MSECOND;
6855 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6856 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6858 /* already (optionally) retrieved this when configuring manager */
6859 if (stream->session) {
6860 GObject *rtpsession = stream->session;
6862 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
6864 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
6868 /* now it happens that (Xenon) server sending this may also provide bogus
6869 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
6870 * and just use RTP-Info to sync */
6872 GObjectClass *klass;
6874 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
6875 if (g_object_class_find_property (klass, "rtcp-sync")) {
6876 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
6877 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
6883 gst_rtspsrc_get_float (const gchar * dstr)
6885 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
6887 /* canonicalise floating point string so we can handle float strings
6888 * in the form "24.930" or "24,930" irrespective of the current locale */
6889 g_strlcpy (s, dstr, sizeof (s));
6890 g_strdelimit (s, ",", '.');
6891 return g_ascii_strtod (s, NULL);
6895 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
6897 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
6899 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
6900 g_strlcpy (val_str, "now", sizeof (val_str));
6902 if (segment->position == 0) {
6903 g_strlcpy (val_str, "0", sizeof (val_str));
6905 g_ascii_dtostr (val_str, sizeof (val_str),
6906 ((gdouble) segment->position) / GST_SECOND);
6909 return g_strdup_printf ("npt=%s-", val_str);
6913 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
6917 stream->timebase = -1;
6918 stream->seqbase = -1;
6920 len = stream->ptmap->len;
6921 for (i = 0; i < len; i++) {
6922 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
6925 if (item->caps == NULL)
6928 item->caps = gst_caps_make_writable (item->caps);
6929 s = gst_caps_get_structure (item->caps, 0);
6930 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
6934 static GstRTSPResult
6935 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
6937 GstRTSPResult res = GST_RTSP_OK;
6939 if (src->state < GST_RTSP_STATE_READY) {
6940 res = GST_RTSP_ERROR;
6941 if (src->open_error) {
6942 GST_DEBUG_OBJECT (src, "the stream was in error");
6946 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
6948 if ((res = gst_rtspsrc_open (src, async)) < 0) {
6949 GST_DEBUG_OBJECT (src, "failed to open stream");
6958 static GstRTSPResult
6959 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
6961 GstRTSPMessage request = { 0 };
6962 GstRTSPMessage response = { 0 };
6963 GstRTSPResult res = GST_RTSP_OK;
6967 const gchar *control;
6969 GST_DEBUG_OBJECT (src, "PLAY...");
6971 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6974 if (!(src->methods & GST_RTSP_PLAY))
6977 if (src->state == GST_RTSP_STATE_PLAYING)
6980 if (!src->conninfo.connection || !src->conninfo.connected)
6983 /* send some dummy packets before we activate the receive in the
6985 gst_rtspsrc_send_dummy_packets (src);
6987 /* require new SR packets */
6989 g_signal_emit_by_name (src->manager, "reset-sync", NULL);
6991 gst_rtspsrc_set_state (src, GST_STATE_PLAYING);
6993 /* construct a control url */
6994 control = get_aggregate_control (src);
6996 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6997 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6998 const gchar *setup_url;
6999 GstRTSPConnection *conn;
7001 /* try aggregate control first but do non-aggregate control otherwise */
7003 setup_url = control;
7004 else if ((setup_url = stream->conninfo.location) == NULL)
7007 if (src->conninfo.connection) {
7008 conn = src->conninfo.connection;
7009 } else if (stream->conninfo.connection) {
7010 conn = stream->conninfo.connection;
7016 res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
7018 goto create_request_failed;
7020 if (src->need_range) {
7021 hval = gen_range_header (src, segment);
7023 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
7025 /* store the newsegment event so it can be sent from the streaming thread. */
7026 if (src->start_segment)
7027 gst_event_unref (src->start_segment);
7028 src->start_segment = gst_event_new_segment (&src->segment);
7031 if (segment->rate != 1.0) {
7032 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
7034 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
7036 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
7038 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
7042 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
7044 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7047 /* seek may have silently failed as it is not supported */
7048 if (!(src->methods & GST_RTSP_PLAY)) {
7049 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
7050 /* obviously it is supported as we made it here */
7051 src->methods |= GST_RTSP_PLAY;
7052 src->seekable = FALSE;
7053 /* but there is nothing to parse in the response,
7054 * so convey we have no idea and not to expect anything particular */
7055 clear_rtp_base (src, stream);
7059 /* need to do for all streams */
7060 for (run = src->streams; run; run = g_list_next (run))
7061 clear_rtp_base (src, (GstRTSPStream *) run->data);
7063 /* NOTE the above also disables npt based eos detection */
7064 /* and below forces position to 0,
7065 * which is visible feedback we lost the plot */
7066 segment->start = segment->position = src->last_pos;
7069 gst_rtsp_message_unset (&request);
7071 /* parse RTP npt field. This is the current position in the stream (Normal
7072 * Play Time) and should be put in the NEWSEGMENT position field. */
7073 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
7075 gst_rtspsrc_parse_range (src, hval, segment);
7077 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
7078 segment->rate = 1.0;
7080 /* parse Speed header. This is the intended playback rate of the stream
7081 * and should be put in the NEWSEGMENT rate field. */
7082 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
7083 0) == GST_RTSP_OK) {
7084 segment->rate = gst_rtspsrc_get_float (hval);
7085 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
7086 &hval, 0) == GST_RTSP_OK) {
7087 segment->rate = gst_rtspsrc_get_float (hval);
7090 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
7091 * for the RTP packets. If this is not present, we assume all starts from 0...
7092 * This is info for the RTP session manager that we pass to it in caps. */
7094 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
7095 &hval, hval_idx++) == GST_RTSP_OK)
7096 gst_rtspsrc_parse_rtpinfo (src, hval);
7098 /* some servers indicate RTCP parameters in PLAY response,
7099 * rather than properly in SDP */
7100 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
7101 &hval, 0) == GST_RTSP_OK)
7102 gst_rtspsrc_handle_rtcp_interval (src, hval);
7104 gst_rtsp_message_unset (&response);
7106 /* early exit when we did aggregate control */
7110 /* configure the caps of the streams after we parsed all headers. Only reset
7111 * the manager object when we set a new Range header (we did a seek) */
7112 gst_rtspsrc_configure_caps (src, segment, src->need_range);
7114 /* set again when needed */
7115 src->need_range = FALSE;
7117 src->running = TRUE;
7118 src->base_time = -1;
7119 src->state = GST_RTSP_STATE_PLAYING;
7122 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
7123 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7124 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7125 stream->discont = TRUE;
7130 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
7137 GST_DEBUG_OBJECT (src, "failed to open stream");
7142 GST_DEBUG_OBJECT (src, "PLAY is not supported");
7147 GST_DEBUG_OBJECT (src, "we were already PLAYING");
7150 create_request_failed:
7152 gchar *str = gst_rtsp_strresult (res);
7154 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7155 ("Could not create request. (%s)", str));
7161 gchar *str = gst_rtsp_strresult (res);
7163 gst_rtsp_message_unset (&request);
7164 if (res != GST_RTSP_EINTR) {
7165 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7166 ("Could not send message. (%s)", str));
7168 GST_WARNING_OBJECT (src, "PLAY interrupted");
7175 static GstRTSPResult
7176 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
7178 GstRTSPResult res = GST_RTSP_OK;
7179 GstRTSPMessage request = { 0 };
7180 GstRTSPMessage response = { 0 };
7182 const gchar *control;
7184 GST_DEBUG_OBJECT (src, "PAUSE...");
7186 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
7189 if (!(src->methods & GST_RTSP_PAUSE))
7192 if (src->state == GST_RTSP_STATE_READY)
7195 if (!src->conninfo.connection || !src->conninfo.connected)
7198 /* construct a control url */
7199 control = get_aggregate_control (src);
7201 /* loop over the streams. We might exit the loop early when we could do an
7202 * aggregate control */
7203 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7204 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7205 GstRTSPConnection *conn;
7206 const gchar *setup_url;
7208 /* try aggregate control first but do non-aggregate control otherwise */
7210 setup_url = control;
7211 else if ((setup_url = stream->conninfo.location) == NULL)
7214 if (src->conninfo.connection) {
7215 conn = src->conninfo.connection;
7216 } else if (stream->conninfo.connection) {
7217 conn = stream->conninfo.connection;
7223 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
7224 ("Sending PAUSE request"));
7227 gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
7229 goto create_request_failed;
7231 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7234 gst_rtsp_message_unset (&request);
7235 gst_rtsp_message_unset (&response);
7237 /* exit early when we did agregate control */
7242 /* change element states now */
7243 gst_rtspsrc_set_state (src, GST_STATE_PAUSED);
7246 src->state = GST_RTSP_STATE_READY;
7250 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
7257 GST_DEBUG_OBJECT (src, "failed to open stream");
7262 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
7267 GST_DEBUG_OBJECT (src, "we were already PAUSED");
7270 create_request_failed:
7272 gchar *str = gst_rtsp_strresult (res);
7274 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7275 ("Could not create request. (%s)", str));
7281 gchar *str = gst_rtsp_strresult (res);
7283 gst_rtsp_message_unset (&request);
7284 if (res != GST_RTSP_EINTR) {
7285 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7286 ("Could not send message. (%s)", str));
7288 GST_WARNING_OBJECT (src, "PAUSE interrupted");
7296 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
7298 GstRTSPSrc *rtspsrc;
7300 rtspsrc = GST_RTSPSRC (bin);
7302 switch (GST_MESSAGE_TYPE (message)) {
7303 case GST_MESSAGE_EOS:
7304 gst_message_unref (message);
7306 case GST_MESSAGE_ELEMENT:
7308 const GstStructure *s = gst_message_get_structure (message);
7310 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
7311 gboolean ignore_timeout;
7313 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
7315 GST_OBJECT_LOCK (rtspsrc);
7316 ignore_timeout = rtspsrc->ignore_timeout;
7317 rtspsrc->ignore_timeout = TRUE;
7318 GST_OBJECT_UNLOCK (rtspsrc);
7320 /* we only act on the first udp timeout message, others are irrelevant
7321 * and can be ignored. */
7322 if (!ignore_timeout)
7323 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
7325 gst_message_unref (message);
7328 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7331 case GST_MESSAGE_ERROR:
7334 GstRTSPStream *stream;
7337 udpsrc = GST_MESSAGE_SRC (message);
7339 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
7340 GST_ELEMENT_NAME (udpsrc));
7342 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
7346 /* we ignore the RTCP udpsrc */
7347 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
7350 /* if we get error messages from the udp sources, that's not a problem as
7351 * long as not all of them error out. We also don't really know what the
7352 * problem is, the message does not give enough detail... */
7353 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
7354 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
7355 if (ret != GST_FLOW_OK)
7359 gst_message_unref (message);
7363 /* fatal but not our message, forward */
7364 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7369 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7375 /* the thread where everything happens */
7377 gst_rtspsrc_thread (GstRTSPSrc * src)
7381 GST_OBJECT_LOCK (src);
7382 cmd = src->pending_cmd;
7383 if (cmd == CMD_RECONNECT || cmd == CMD_PLAY || cmd == CMD_PAUSE
7384 || cmd == CMD_LOOP || cmd == CMD_OPEN)
7385 src->pending_cmd = CMD_LOOP;
7387 src->pending_cmd = CMD_WAIT;
7388 GST_DEBUG_OBJECT (src, "got command %d", cmd);
7390 /* we got the message command, so ensure communication is possible again */
7391 gst_rtspsrc_connection_flush (src, FALSE);
7393 src->busy_cmd = cmd;
7394 GST_OBJECT_UNLOCK (src);
7398 gst_rtspsrc_open (src, TRUE);
7401 gst_rtspsrc_play (src, &src->segment, TRUE);
7404 gst_rtspsrc_pause (src, TRUE);
7407 gst_rtspsrc_close (src, TRUE, FALSE);
7410 gst_rtspsrc_loop (src);
7413 gst_rtspsrc_reconnect (src, FALSE);
7419 GST_OBJECT_LOCK (src);
7420 /* and go back to sleep */
7421 if (src->pending_cmd == CMD_WAIT) {
7423 gst_task_pause (src->task);
7426 src->busy_cmd = CMD_WAIT;
7427 GST_OBJECT_UNLOCK (src);
7431 gst_rtspsrc_start (GstRTSPSrc * src)
7433 GST_DEBUG_OBJECT (src, "starting");
7435 GST_OBJECT_LOCK (src);
7437 src->pending_cmd = CMD_WAIT;
7439 if (src->task == NULL) {
7440 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src, NULL);
7441 if (src->task == NULL)
7444 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
7446 GST_OBJECT_UNLOCK (src);
7453 GST_OBJECT_UNLOCK (src);
7454 GST_ERROR_OBJECT (src, "failed to create task");
7460 gst_rtspsrc_stop (GstRTSPSrc * src)
7464 GST_DEBUG_OBJECT (src, "stopping");
7466 /* also cancels pending task */
7467 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_ALL);
7469 GST_OBJECT_LOCK (src);
7470 if ((task = src->task)) {
7472 GST_OBJECT_UNLOCK (src);
7474 gst_task_stop (task);
7476 /* make sure it is not running */
7477 GST_RTSP_STREAM_LOCK (src);
7478 GST_RTSP_STREAM_UNLOCK (src);
7480 /* now wait for the task to finish */
7481 gst_task_join (task);
7483 /* and free the task */
7484 gst_object_unref (GST_OBJECT (task));
7486 GST_OBJECT_LOCK (src);
7488 GST_OBJECT_UNLOCK (src);
7490 /* ensure synchronously all is closed and clean */
7491 gst_rtspsrc_close (src, FALSE, TRUE);
7496 static GstStateChangeReturn
7497 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
7499 GstRTSPSrc *rtspsrc;
7500 GstStateChangeReturn ret;
7502 rtspsrc = GST_RTSPSRC (element);
7504 switch (transition) {
7505 case GST_STATE_CHANGE_NULL_TO_READY:
7506 if (!gst_rtspsrc_start (rtspsrc))
7509 case GST_STATE_CHANGE_READY_TO_PAUSED:
7510 /* init some state */
7511 rtspsrc->cur_protocols = rtspsrc->protocols;
7512 /* first attempt, don't ignore timeouts */
7513 rtspsrc->ignore_timeout = FALSE;
7514 rtspsrc->open_error = FALSE;
7515 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
7517 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
7518 set_manager_buffer_mode (rtspsrc);
7520 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
7521 /* unblock the tcp tasks and make the loop waiting */
7522 if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
7523 /* make sure it is waiting before we send PAUSE or PLAY below */
7524 GST_RTSP_STREAM_LOCK (rtspsrc);
7525 GST_RTSP_STREAM_UNLOCK (rtspsrc);
7528 case GST_STATE_CHANGE_PAUSED_TO_READY:
7534 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
7535 if (ret == GST_STATE_CHANGE_FAILURE)
7538 switch (transition) {
7539 case GST_STATE_CHANGE_NULL_TO_READY:
7540 ret = GST_STATE_CHANGE_SUCCESS;
7542 case GST_STATE_CHANGE_READY_TO_PAUSED:
7543 ret = GST_STATE_CHANGE_NO_PREROLL;
7545 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
7546 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
7547 ret = GST_STATE_CHANGE_SUCCESS;
7549 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
7550 /* send pause request and keep the idle task around */
7551 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
7552 ret = GST_STATE_CHANGE_NO_PREROLL;
7554 case GST_STATE_CHANGE_PAUSED_TO_READY:
7555 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
7556 ret = GST_STATE_CHANGE_SUCCESS;
7558 case GST_STATE_CHANGE_READY_TO_NULL:
7559 gst_rtspsrc_stop (rtspsrc);
7560 ret = GST_STATE_CHANGE_SUCCESS;
7571 GST_DEBUG_OBJECT (rtspsrc, "start failed");
7572 return GST_STATE_CHANGE_FAILURE;
7577 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
7580 GstRTSPSrc *rtspsrc;
7582 rtspsrc = GST_RTSPSRC (element);
7584 if (GST_EVENT_IS_DOWNSTREAM (event)) {
7585 res = gst_rtspsrc_push_event (rtspsrc, event);
7587 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
7594 /*** GSTURIHANDLER INTERFACE *************************************************/
7597 gst_rtspsrc_uri_get_type (GType type)
7602 static const gchar *const *
7603 gst_rtspsrc_uri_get_protocols (GType type)
7605 static const gchar *protocols[] =
7606 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
7607 "rtsps", "rtspsu", "rtspst", "rtspsh", NULL
7614 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
7616 GstRTSPSrc *src = GST_RTSPSRC (handler);
7618 /* FIXME: make thread-safe */
7619 return g_strdup (src->conninfo.location);
7623 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
7628 GstRTSPUrl *newurl = NULL;
7629 GstSDPMessage *sdp = NULL;
7631 src = GST_RTSPSRC (handler);
7633 /* same URI, we're fine */
7634 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
7637 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
7638 if ((res = gst_sdp_message_new (&sdp) < 0))
7641 GST_DEBUG_OBJECT (src, "parsing SDP message");
7642 if ((res = gst_sdp_message_parse_uri (uri, sdp) < 0))
7646 GST_DEBUG_OBJECT (src, "parsing URI");
7647 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
7651 /* if worked, free previous and store new url object along with the original
7653 GST_DEBUG_OBJECT (src, "configuring URI");
7654 g_free (src->conninfo.location);
7655 src->conninfo.location = g_strdup (uri);
7656 gst_rtsp_url_free (src->conninfo.url);
7657 src->conninfo.url = newurl;
7658 g_free (src->conninfo.url_str);
7660 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
7662 src->conninfo.url_str = NULL;
7665 gst_sdp_message_free (src->sdp);
7667 src->from_sdp = sdp != NULL;
7669 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
7670 GST_DEBUG_OBJECT (src, "request uri is: %s",
7671 GST_STR_NULL (src->conninfo.url_str));
7678 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
7683 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", res);
7684 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7685 "Could not create SDP");
7690 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", res,
7691 GST_STR_NULL (uri));
7692 gst_sdp_message_free (sdp);
7693 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7699 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
7700 GST_STR_NULL (uri), res);
7701 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7702 "Invalid RTSP URI");
7708 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
7710 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
7712 iface->get_type = gst_rtspsrc_uri_get_type;
7713 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
7714 iface->get_uri = gst_rtspsrc_uri_get_uri;
7715 iface->set_uri = gst_rtspsrc_uri_set_uri;