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]);
1517 stream->udpsrc[i] = NULL;
1519 if (stream->channelpad[i]) {
1520 gst_object_unref (stream->channelpad[i]);
1521 stream->channelpad[i] = NULL;
1523 if (stream->udpsink[i]) {
1524 gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
1525 gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
1526 gst_object_unref (stream->udpsink[i]);
1527 stream->udpsink[i] = NULL;
1530 if (stream->fakesrc) {
1531 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
1532 gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
1533 gst_object_unref (stream->fakesrc);
1534 stream->fakesrc = NULL;
1536 if (stream->srcpad) {
1537 gst_pad_set_active (stream->srcpad, FALSE);
1538 if (stream->added) {
1539 gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
1540 stream->added = FALSE;
1542 stream->srcpad = NULL;
1544 if (stream->rtcppad) {
1545 gst_object_unref (stream->rtcppad);
1546 stream->rtcppad = NULL;
1548 if (stream->session) {
1549 g_object_unref (stream->session);
1550 stream->session = NULL;
1556 gst_rtspsrc_cleanup (GstRTSPSrc * src)
1560 GST_DEBUG_OBJECT (src, "cleanup");
1562 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1563 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1565 gst_rtspsrc_stream_free (src, stream);
1567 g_list_free (src->streams);
1568 src->streams = NULL;
1570 if (src->manager_sig_id) {
1571 g_signal_handler_disconnect (src->manager, src->manager_sig_id);
1572 src->manager_sig_id = 0;
1574 gst_element_set_state (src->manager, GST_STATE_NULL);
1575 gst_bin_remove (GST_BIN_CAST (src), src->manager);
1576 src->manager = NULL;
1579 gst_structure_free (src->props);
1582 g_free (src->content_base);
1583 src->content_base = NULL;
1585 g_free (src->control);
1586 src->control = NULL;
1589 gst_rtsp_range_free (src->range);
1592 /* don't clear the SDP when it was used in the url */
1593 if (src->sdp && !src->from_sdp) {
1594 gst_sdp_message_free (src->sdp);
1597 if (src->start_segment) {
1598 gst_event_unref (src->start_segment);
1599 src->start_segment = NULL;
1601 if (src->provided_clock) {
1602 gst_object_unref (src->provided_clock);
1603 src->provided_clock = NULL;
1607 #define PARSE_INT(p, del, res) \
1610 p = strstr (p, del); \
1620 #define PARSE_STRING(p, del, res) \
1623 p = strstr (p, del); \
1635 #define SKIP_SPACES(p) \
1636 while (*p && g_ascii_isspace (*p)) \
1641 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1644 gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
1645 gint * rate, gchar ** params)
1649 p = (gchar *) rtpmap;
1651 PARSE_INT (p, " ", *payload);
1659 PARSE_STRING (p, "/", *name);
1660 if (*name == NULL) {
1661 GST_DEBUG ("no rate, name %s", p);
1662 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
1663 * streams seem to omit the rate. */
1670 p = strstr (p, "/");
1688 parse_keymgmt (const gchar * keymgmt, GstCaps * caps)
1690 gboolean res = FALSE;
1694 GstMIKEYMessage *msg;
1695 const GstMIKEYPayload *payload;
1696 const gchar *srtp_cipher;
1697 const gchar *srtp_auth;
1699 p = (gchar *) keymgmt;
1705 PARSE_STRING (p, " ", kmpid);
1706 if (!g_str_equal (kmpid, "mikey"))
1709 data = g_base64_decode (p, &size);
1713 msg = gst_mikey_message_new_from_data (data, size);
1717 srtp_cipher = "aes-128-icm";
1718 srtp_auth = "hmac-sha1-80";
1720 /* check the Security policy if any */
1721 if ((payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, 0))) {
1722 GstMIKEYPayloadSP *p = (GstMIKEYPayloadSP *) payload;
1725 if (p->proto != GST_MIKEY_SEC_PROTO_SRTP)
1728 len = gst_mikey_payload_sp_get_n_params (payload);
1729 for (i = 0; i < len; i++) {
1730 const GstMIKEYPayloadSPParam *param =
1731 gst_mikey_payload_sp_get_param (payload, i);
1733 switch (param->type) {
1734 case GST_MIKEY_SP_SRTP_ENC_ALG:
1735 switch (param->val[0]) {
1737 srtp_cipher = "null";
1741 srtp_cipher = "aes-128-icm";
1747 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1748 switch (param->val[0]) {
1754 srtp_auth = "hmac-sha1-80";
1760 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1762 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1770 if (!(payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_KEMAC, 0)))
1773 GstMIKEYPayloadKEMAC *p = (GstMIKEYPayloadKEMAC *) payload;
1776 if (p->enc_alg != GST_MIKEY_ENC_NULL || p->mac_alg != GST_MIKEY_MAC_NULL)
1780 gst_buffer_new_wrapped (g_memdup (p->enc_data, p->enc_len), p->enc_len);
1781 gst_caps_set_simple (caps, "srtp-key", GST_TYPE_BUFFER, buf, NULL);
1784 gst_caps_set_simple (caps,
1785 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
1786 "srtp-auth", G_TYPE_STRING, srtp_auth,
1787 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
1788 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
1792 gst_mikey_message_free (msg);
1798 * Mapping SDP attributes to caps
1800 * prepend 'a-' to IANA registered sdp attributes names
1801 * (ie: not prefixed with 'x-') in order to avoid
1802 * collision with gstreamer standard caps properties names
1805 gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
1807 if (attributes->len > 0) {
1811 s = gst_caps_get_structure (caps, 0);
1813 for (i = 0; i < attributes->len; i++) {
1814 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
1815 gchar *tofree, *key;
1819 /* skip some of the attribute we already handle */
1820 if (!strcmp (key, "fmtp"))
1822 if (!strcmp (key, "rtpmap"))
1824 if (!strcmp (key, "control"))
1826 if (!strcmp (key, "range"))
1828 if (g_str_equal (key, "key-mgmt")) {
1829 parse_keymgmt (attr->value, caps);
1833 /* string must be valid UTF8 */
1834 if (!g_utf8_validate (attr->value, -1, NULL))
1837 if (!g_str_has_prefix (key, "x-"))
1838 tofree = key = g_strdup_printf ("a-%s", key);
1842 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
1843 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
1849 static const gchar *
1850 rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
1859 if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
1862 if (sscanf (attr, "%d ", &val) != 1)
1872 * Mapping of caps to and from SDP fields:
1874 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1875 * a=fmtp:<payload> <param>[=<value>];...
1878 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
1881 const gchar *rtpmap;
1885 gchar *params = NULL;
1891 /* get and parse rtpmap */
1892 rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
1895 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
1897 g_warning ("error parsing rtpmap, ignoring");
1901 /* dynamic payloads need rtpmap or we fail */
1902 if (rtpmap == NULL && pt >= 96)
1905 /* check if we have a rate, if not, we need to look up the rate from the
1906 * default rates based on the payload types. */
1908 const GstRTPPayloadInfo *info;
1910 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
1911 /* dynamic types, use media and encoding_name */
1912 tmp = g_ascii_strdown (media->media, -1);
1913 info = gst_rtp_payload_info_for_name (tmp, name);
1916 /* static types, use payload type */
1917 info = gst_rtp_payload_info_for_pt (pt);
1921 if ((rate = info->clock_rate) == 0)
1924 /* we fail if we cannot find one */
1929 tmp = g_ascii_strdown (media->media, -1);
1930 caps = gst_caps_new_simple ("application/x-unknown",
1931 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
1933 s = gst_caps_get_structure (caps, 0);
1935 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
1937 /* encoding name must be upper case */
1939 tmp = g_ascii_strup (name, -1);
1940 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
1944 /* params must be lower case */
1945 if (params != NULL) {
1946 tmp = g_ascii_strdown (params, -1);
1947 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
1951 /* parse optional fmtp: field */
1952 if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
1958 /* p is now of the format <payload> <param>[=<value>];... */
1959 PARSE_INT (p, " ", payload);
1960 if (payload != -1 && payload == pt) {
1964 /* <param>[=<value>] are separated with ';' */
1965 pairs = g_strsplit (p, ";", 0);
1966 for (i = 0; pairs[i]; i++) {
1968 const gchar *val, *key;
1970 /* the key may not have a '=', the value can have other '='s */
1971 valpos = strstr (pairs[i], "=");
1973 /* we have a '=' and thus a value, remove the '=' with \0 */
1975 /* value is everything between '=' and ';'. We split the pairs at ;
1976 * boundaries so we can take the remainder of the value. Some servers
1977 * put spaces around the value which we strip off here. Alternatively
1978 * we could strip those spaces in the depayloaders should these spaces
1979 * actually carry any meaning in the future. */
1980 val = g_strstrip (valpos + 1);
1982 /* simple <param>;.. is translated into <param>=1;... */
1985 /* strip the key of spaces, convert key to lowercase but not the value. */
1986 key = g_strstrip (pairs[i]);
1987 if (strlen (key) > 1) {
1988 tmp = g_ascii_strdown (key, -1);
1989 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
2001 g_warning ("rtpmap type not given for dynamic payload %d", pt);
2006 g_warning ("rate unknown for payload type %d", pt);
2012 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
2013 gint * rtpport, gint * rtcpport)
2016 GstStateChangeReturn ret;
2017 GstElement *udpsrc0, *udpsrc1;
2018 gint tmp_rtp, tmp_rtcp;
2022 src = stream->parent;
2028 /* Start at next port */
2029 tmp_rtp = src->next_port_num;
2031 if (stream->is_ipv6)
2032 host = "udp://[::0]";
2034 host = "udp://0.0.0.0";
2036 /* try to allocate 2 UDP ports, the RTP port should be an even
2037 * number and the RTCP port should be the next (uneven) port */
2040 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
2041 tmp_rtp >= src->client_port_range.max)
2044 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2045 if (udpsrc0 == NULL)
2046 goto no_udp_protocol;
2047 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
2049 if (src->udp_buffer_size != 0)
2050 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
2053 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
2054 if (ret == GST_STATE_CHANGE_FAILURE) {
2056 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
2059 if (++count > src->retry)
2062 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2063 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2064 gst_object_unref (udpsrc0);
2067 GST_DEBUG_OBJECT (src, "retry %d", count);
2070 goto no_udp_protocol;
2073 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
2074 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
2076 /* check if port is even */
2077 if ((tmp_rtp & 0x01) != 0) {
2078 /* port not even, close and allocate another */
2079 if (++count > src->retry)
2082 GST_DEBUG_OBJECT (src, "RTP port not even");
2084 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2085 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2086 gst_object_unref (udpsrc0);
2089 GST_DEBUG_OBJECT (src, "retry %d", count);
2094 /* allocate port+1 for RTCP now */
2095 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2096 if (udpsrc1 == NULL)
2097 goto no_udp_rtcp_protocol;
2100 tmp_rtcp = tmp_rtp + 1;
2101 if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
2104 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
2106 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
2107 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
2108 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
2109 if (ret == GST_STATE_CHANGE_FAILURE) {
2110 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
2112 if (++count > src->retry)
2115 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2116 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2117 gst_object_unref (udpsrc0);
2120 GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
2121 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2122 gst_object_unref (udpsrc1);
2126 GST_DEBUG_OBJECT (src, "retry %d", count);
2130 /* all fine, do port check */
2131 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
2132 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
2134 /* this should not happen... */
2135 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
2138 /* we keep these elements, we configure all in configure_transport when the
2139 * server told us to really use the UDP ports. */
2140 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
2141 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
2142 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
2143 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
2145 /* keep track of next available port number when we have a range
2147 if (src->next_port_num != 0)
2148 src->next_port_num = tmp_rtcp + 1;
2155 GST_DEBUG_OBJECT (src, "could not get UDP source");
2160 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
2164 no_udp_rtcp_protocol:
2166 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
2171 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
2172 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
2178 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2179 gst_object_unref (udpsrc0);
2182 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2183 gst_object_unref (udpsrc1);
2190 gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
2195 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
2197 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2198 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2201 for (i = 0; i < 2; i++) {
2202 if (stream->udpsrc[i])
2203 gst_element_set_state (stream->udpsrc[i], state);
2209 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
2216 event = gst_event_new_flush_start ();
2217 GST_DEBUG_OBJECT (src, "start flush");
2219 state = GST_STATE_PAUSED;
2221 event = gst_event_new_flush_stop (FALSE);
2222 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
2225 state = GST_STATE_PLAYING;
2227 state = GST_STATE_PAUSED;
2229 gst_rtspsrc_push_event (src, event);
2230 gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
2231 gst_rtspsrc_set_state (src, state);
2234 static GstRTSPResult
2235 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
2236 GstRTSPMessage * message, GTimeVal * timeout)
2241 ret = gst_rtsp_connection_send (conn, message, timeout);
2243 ret = GST_RTSP_ERROR;
2248 static GstRTSPResult
2249 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
2250 GstRTSPMessage * message, GTimeVal * timeout)
2255 ret = gst_rtsp_connection_receive (conn, message, timeout);
2257 ret = GST_RTSP_ERROR;
2263 gst_rtspsrc_get_position (GstRTSPSrc * src)
2268 query = gst_query_new_position (GST_FORMAT_TIME);
2269 /* should be known somewhere down the stream (e.g. jitterbuffer) */
2270 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2271 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2275 if (stream->srcpad) {
2276 if (gst_pad_query (stream->srcpad, query)) {
2277 gst_query_parse_position (query, &fmt, &pos);
2278 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
2279 GST_TIME_ARGS (pos));
2280 src->last_pos = pos;
2290 gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
2292 src->state = GST_RTSP_STATE_SEEKING;
2293 /* PLAY will add the range header now. */
2294 src->need_range = TRUE;
2300 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
2305 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
2307 gboolean flush, skip;
2310 GstSegment seeksegment = { 0, };
2314 GST_DEBUG_OBJECT (src, "doing seek with event");
2316 gst_event_parse_seek (event, &rate, &format, &flags,
2317 &cur_type, &cur, &stop_type, &stop);
2319 /* no negative rates yet */
2323 /* we need TIME format */
2324 if (format != src->segment.format)
2327 GST_DEBUG_OBJECT (src, "doing seek without event");
2329 cur_type = GST_SEEK_TYPE_SET;
2330 stop_type = GST_SEEK_TYPE_SET;
2333 /* get flush flag */
2334 flush = flags & GST_SEEK_FLAG_FLUSH;
2335 skip = flags & GST_SEEK_FLAG_SKIP;
2337 /* now we need to make sure the streaming thread is stopped. We do this by
2338 * either sending a FLUSH_START event downstream which will cause the
2339 * streaming thread to stop with a WRONG_STATE.
2340 * For a non-flushing seek we simply pause the task, which will happen as soon
2341 * as it completes one iteration (and thus might block when the sink is
2342 * blocking in preroll). */
2344 GST_DEBUG_OBJECT (src, "starting flush");
2345 gst_rtspsrc_flush (src, TRUE, FALSE);
2348 gst_task_pause (src->task);
2352 /* we should now be able to grab the streaming thread because we stopped it
2353 * with the above flush/pause code */
2354 GST_RTSP_STREAM_LOCK (src);
2356 GST_DEBUG_OBJECT (src, "stopped streaming");
2358 /* stop flushing the rtsp connection so we can send PAUSE/PLAY below */
2359 gst_rtspsrc_connection_flush (src, FALSE);
2361 /* copy segment, we need this because we still need the old
2362 * segment when we close the current segment. */
2363 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
2365 /* configure the seek parameters in the seeksegment. We will then have the
2366 * right values in the segment to perform the seek */
2368 GST_DEBUG_OBJECT (src, "configuring seek");
2369 gst_segment_do_seek (&seeksegment, rate, format, flags,
2370 cur_type, cur, stop_type, stop, &update);
2373 /* figure out the last position we need to play. If it's configured (stop !=
2374 * -1), use that, else we play until the total duration of the file */
2375 if ((stop = seeksegment.stop) == -1)
2376 stop = seeksegment.duration;
2378 playing = (src->state == GST_RTSP_STATE_PLAYING);
2380 /* if we were playing, pause first */
2382 /* obtain current position in case seek fails */
2383 gst_rtspsrc_get_position (src);
2384 gst_rtspsrc_pause (src, FALSE);
2388 gst_rtspsrc_do_seek (src, &seeksegment);
2390 /* and continue playing */
2392 gst_rtspsrc_play (src, &seeksegment, FALSE);
2394 /* prepare for streaming again */
2396 /* if we started flush, we stop now */
2397 GST_DEBUG_OBJECT (src, "stopping flush");
2398 gst_rtspsrc_flush (src, FALSE, playing);
2401 /* now we did the seek and can activate the new segment values */
2402 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
2404 /* if we're doing a segment seek, post a SEGMENT_START message */
2405 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2406 gst_element_post_message (GST_ELEMENT_CAST (src),
2407 gst_message_new_segment_start (GST_OBJECT_CAST (src),
2408 src->segment.format, src->segment.position));
2411 /* now create the newsegment */
2412 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
2413 " to %" G_GINT64_FORMAT, src->segment.position, stop);
2416 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
2417 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2418 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2419 stream->discont = TRUE;
2422 GST_RTSP_STREAM_UNLOCK (src);
2429 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
2434 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
2440 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
2444 gboolean res = TRUE;
2447 src = GST_RTSPSRC_CAST (parent);
2449 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
2450 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
2452 switch (GST_EVENT_TYPE (event)) {
2453 case GST_EVENT_SEEK:
2454 res = gst_rtspsrc_perform_seek (src, event);
2458 case GST_EVENT_NAVIGATION:
2459 case GST_EVENT_LATENCY:
2467 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
2468 res = gst_pad_send_event (target, event);
2469 gst_object_unref (target);
2471 gst_event_unref (event);
2474 gst_event_unref (event);
2480 /* this is the final event function we receive on the internal source pad when
2481 * we deal with TCP connections */
2483 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
2488 GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
2490 switch (GST_EVENT_TYPE (event)) {
2491 case GST_EVENT_SEEK:
2493 case GST_EVENT_NAVIGATION:
2494 case GST_EVENT_LATENCY:
2496 gst_event_unref (event);
2503 /* this is the final query function we receive on the internal source pad when
2504 * we deal with TCP connections */
2506 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
2510 gboolean res = TRUE;
2512 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
2514 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2515 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2517 switch (GST_QUERY_TYPE (query)) {
2518 case GST_QUERY_POSITION:
2523 case GST_QUERY_DURATION:
2527 gst_query_parse_duration (query, &format, NULL);
2530 case GST_FORMAT_TIME:
2531 gst_query_set_duration (query, format, src->segment.duration);
2539 case GST_QUERY_LATENCY:
2541 /* we are live with a min latency of 0 and unlimited max latency, this
2542 * result will be updated by the session manager if there is any. */
2543 gst_query_set_latency (query, TRUE, 0, -1);
2553 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2555 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2559 gboolean res = FALSE;
2561 src = GST_RTSPSRC_CAST (parent);
2563 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2564 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2566 switch (GST_QUERY_TYPE (query)) {
2567 case GST_QUERY_DURATION:
2571 gst_query_parse_duration (query, &format, NULL);
2574 case GST_FORMAT_TIME:
2575 gst_query_set_duration (query, format, src->segment.duration);
2583 case GST_QUERY_SEEKING:
2587 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2588 if (format == GST_FORMAT_TIME) {
2590 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2592 /* seeking without duration is unlikely */
2593 seekable = seekable && src->seekable && src->segment.duration &&
2594 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2596 /* FIXME ?? should we have 0 and segment.duration here; see demuxers */
2597 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
2598 src->segment.start, src->segment.stop);
2607 uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
2609 gst_query_set_uri (query, uri);
2617 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2619 /* forward the query to the proxy target pad */
2621 res = gst_pad_query (target, query);
2622 gst_object_unref (target);
2631 /* callback for RTCP messages to be sent to the server when operating in TCP
2633 static GstFlowReturn
2634 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2637 GstRTSPStream *stream;
2638 GstFlowReturn res = GST_FLOW_OK;
2643 GstRTSPMessage message = { 0 };
2644 GstRTSPConnection *conn;
2646 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2647 src = stream->parent;
2649 gst_buffer_map (buffer, &map, GST_MAP_READ);
2653 gst_rtsp_message_init_data (&message, stream->channel[1]);
2655 /* lend the body data to the message */
2656 gst_rtsp_message_take_body (&message, data, size);
2658 if (stream->conninfo.connection)
2659 conn = stream->conninfo.connection;
2661 conn = src->conninfo.connection;
2663 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2664 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2665 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2667 /* and steal it away again because we will free it when unreffing the
2669 gst_rtsp_message_steal_body (&message, &data, &size);
2670 gst_rtsp_message_unset (&message);
2672 gst_buffer_unmap (buffer, &map);
2673 gst_buffer_unref (buffer);
2678 static GstPadProbeReturn
2679 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2681 GstRTSPSrc *src = user_data;
2683 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2684 GST_DEBUG_PAD_NAME (pad));
2686 /* activate the streams */
2687 GST_OBJECT_LOCK (src);
2688 if (!src->need_activate)
2691 src->need_activate = FALSE;
2692 GST_OBJECT_UNLOCK (src);
2694 gst_rtspsrc_activate_streams (src);
2696 return GST_PAD_PROBE_OK;
2700 GST_OBJECT_UNLOCK (src);
2701 return GST_PAD_PROBE_OK;
2706 copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
2708 GstPad *gpad = GST_PAD_CAST (user_data);
2710 GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
2711 gst_pad_store_sticky_event (gpad, *event);
2716 /* this callback is called when the session manager generated a new src pad with
2717 * payloaded RTP packets. We simply ghost the pad here. */
2719 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2722 GstPadTemplate *template;
2725 GstRTSPStream *stream;
2728 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2730 GST_RTSP_STATE_LOCK (src);
2732 name = gst_object_get_name (GST_OBJECT_CAST (pad));
2733 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
2734 goto unknown_stream;
2736 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);
2738 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
2740 goto unknown_stream;
2743 stream->ssrc = ssrc;
2745 /* we'll add it later see below */
2746 stream->added = TRUE;
2748 /* check if we added all streams */
2750 for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
2751 GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;
2753 GST_DEBUG_OBJECT (src, "stream %p, container %d, added %d, setup %d",
2754 ostream, ostream->container, ostream->added, ostream->setup);
2756 /* if we find a stream for which we did a setup that is not added, we
2757 * need to wait some more */
2758 if (ostream->setup && !ostream->added) {
2763 GST_RTSP_STATE_UNLOCK (src);
2765 /* create a new pad we will use to stream to */
2766 template = gst_static_pad_template_get (&rtptemplate);
2767 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
2768 gst_object_unref (template);
2771 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
2772 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
2773 gst_pad_set_active (stream->srcpad, TRUE);
2774 gst_pad_sticky_events_foreach (pad, copy_sticky_events, stream->srcpad);
2775 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
2778 GST_DEBUG_OBJECT (src, "We added all streams");
2779 /* when we get here, all stream are added and we can fire the no-more-pads
2781 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
2789 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
2790 GST_RTSP_STATE_UNLOCK (src);
2797 stream_get_caps_for_pt (GstRTSPStream * stream, guint pt)
2801 len = stream->ptmap->len;
2802 for (i = 0; i < len; i++) {
2803 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
2811 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
2813 GstRTSPStream *stream;
2816 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
2818 GST_RTSP_STATE_LOCK (src);
2819 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2821 goto unknown_stream;
2823 if ((caps = stream_get_caps_for_pt (stream, pt)))
2824 gst_caps_ref (caps);
2825 GST_RTSP_STATE_UNLOCK (src);
2831 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
2832 GST_RTSP_STATE_UNLOCK (src);
2838 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
2840 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
2846 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
2852 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
2858 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2860 GstRTSPSrc *src = stream->parent;
2863 g_object_get (source, "ssrc", &ssrc, NULL);
2865 GST_DEBUG_OBJECT (src, "source %08x, stream %08x, session %u received BYE",
2866 ssrc, stream->ssrc, stream->id);
2868 if (ssrc == stream->ssrc)
2869 gst_rtspsrc_do_stream_eos (src, stream);
2873 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2875 GstRTSPSrc *src = stream->parent;
2878 g_object_get (source, "ssrc", &ssrc, NULL);
2880 GST_WARNING_OBJECT (src, "source %08x, stream %08x in session %u timed out",
2881 ssrc, stream->ssrc, stream->id);
2883 if (ssrc == stream->ssrc)
2884 gst_rtspsrc_do_stream_eos (src, stream);
2888 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
2890 GstRTSPStream *stream;
2892 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
2894 /* get stream for session */
2895 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2897 gst_rtspsrc_do_stream_eos (src, stream);
2902 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
2904 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
2909 set_manager_buffer_mode (GstRTSPSrc * src)
2911 GObjectClass *klass;
2913 if (src->manager == NULL)
2916 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
2918 if (!g_object_class_find_property (klass, "buffer-mode"))
2921 if (src->buffer_mode != BUFFER_MODE_AUTO) {
2922 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
2927 GST_DEBUG_OBJECT (src,
2928 "auto buffering mode, have clock %" GST_PTR_FORMAT, src->provided_clock);
2930 if (src->provided_clock) {
2931 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
2933 if (clock == src->provided_clock) {
2934 GST_DEBUG_OBJECT (src, "selected synced");
2935 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SYNCED, NULL);
2938 gst_object_unref (clock);
2943 /* Otherwise fall-through and use another buffer mode */
2945 gst_object_unref (clock);
2948 GST_DEBUG_OBJECT (src, "auto buffering mode");
2949 if (src->use_buffering) {
2950 GST_DEBUG_OBJECT (src, "selected buffer");
2951 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER, NULL);
2953 GST_DEBUG_OBJECT (src, "selected slave");
2954 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
2959 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
2961 GST_DEBUG ("request key %u", ssrc);
2962 return gst_caps_ref (stream_get_caps_for_pt (stream, stream->default_pt));
2966 request_rtp_decoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
2968 if (stream->id != session)
2971 if (stream->profile != GST_RTSP_PROFILE_SAVP &&
2972 stream->profile != GST_RTSP_PROFILE_SAVPF)
2975 if (stream->srtpdec == NULL) {
2978 name = g_strdup_printf ("srtpdec_%u", session);
2979 stream->srtpdec = gst_element_factory_make ("srtpdec", name);
2982 g_signal_connect (stream->srtpdec, "request-key",
2983 (GCallback) request_key, stream);
2985 return gst_object_ref (stream->srtpdec);
2988 /* try to get and configure a manager */
2990 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
2991 GstRTSPTransport * transport)
2993 const gchar *manager;
2995 GstStateChangeReturn ret;
2997 /* find a manager */
2998 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
3002 GST_DEBUG_OBJECT (src, "using manager %s", manager);
3004 /* configure the manager */
3005 if (src->manager == NULL) {
3006 GObjectClass *klass;
3008 if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
3010 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
3014 goto use_no_manager;
3016 if (!(src->manager = gst_element_factory_make (manager, "manager")))
3017 goto manager_failed;
3020 /* we manage this element */
3021 gst_element_set_locked_state (src->manager, TRUE);
3022 gst_bin_add (GST_BIN_CAST (src), src->manager);
3024 ret = gst_element_set_state (src->manager, GST_STATE_PAUSED);
3025 if (ret == GST_STATE_CHANGE_FAILURE)
3026 goto start_manager_failure;
3028 g_object_set (src->manager, "latency", src->latency, NULL);
3030 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
3032 if (g_object_class_find_property (klass, "ntp-sync")) {
3033 g_object_set (src->manager, "ntp-sync", src->ntp_sync, NULL);
3036 if (g_object_class_find_property (klass, "use-pipeline-clock")) {
3037 g_object_set (src->manager, "use-pipeline-clock",
3038 src->use_pipeline_clock, NULL);
3041 if (src->sdes && g_object_class_find_property (klass, "sdes")) {
3042 g_object_set (src->manager, "sdes", src->sdes, NULL);
3045 if (g_object_class_find_property (klass, "drop-on-latency")) {
3046 g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
3050 /* buffer mode pauses are handled by adding offsets to buffer times,
3051 * but some depayloaders may have a hard time syncing output times
3052 * with such input times, e.g. container ones, most notably ASF */
3053 /* TODO alternatives are having an event that indicates these shifts,
3054 * or having rtsp extensions provide suggestion on buffer mode */
3055 /* valid duration implies not likely live pipeline,
3056 * so slaving in jitterbuffer does not make much sense
3057 * (and might mess things up due to bursts) */
3058 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
3059 src->segment.duration && !stream->container) {
3060 src->use_buffering = TRUE;
3062 src->use_buffering = FALSE;
3065 set_manager_buffer_mode (src);
3067 /* connect to signals */
3068 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
3070 src->manager_sig_id =
3071 g_signal_connect (src->manager, "pad-added",
3072 (GCallback) new_manager_pad, src);
3073 src->manager_ptmap_id =
3074 g_signal_connect (src->manager, "request-pt-map",
3075 (GCallback) request_pt_map, src);
3077 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
3080 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_NEW_MANAGER], 0,
3083 g_signal_connect (src->manager, "request-rtp-decoder",
3084 (GCallback) request_rtp_decoder, stream);
3085 g_signal_connect (src->manager, "request-rtcp-decoder",
3086 (GCallback) request_rtp_decoder, stream);
3088 /* we stream directly to the manager, get some pads. Each RTSP stream goes
3089 * into a separate RTP session. */
3090 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
3091 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
3093 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
3094 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
3097 /* now configure the bandwidth in the manager */
3098 if (g_signal_lookup ("get-internal-session",
3099 G_OBJECT_TYPE (src->manager)) != 0) {
3100 GObject *rtpsession;
3102 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
3105 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
3107 stream->session = rtpsession;
3109 if (stream->as_bandwidth != -1) {
3110 GST_INFO_OBJECT (src, "setting AS: %f",
3111 (gdouble) (stream->as_bandwidth * 1000));
3112 g_object_set (rtpsession, "bandwidth",
3113 (gdouble) (stream->as_bandwidth * 1000), NULL);
3115 if (stream->rr_bandwidth != -1) {
3116 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
3117 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
3120 if (stream->rs_bandwidth != -1) {
3121 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
3122 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
3126 g_object_set (rtpsession, "probation", src->probation, NULL);
3128 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
3130 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
3132 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
3134 g_signal_connect (rtpsession, "on-ssrc-active",
3135 (GCallback) on_ssrc_active, stream);
3146 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3151 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
3154 start_manager_failure:
3156 GST_DEBUG_OBJECT (src, "could not start session manager");
3161 /* free the UDP sources allocated when negotiating a transport.
3162 * This function is called when the server negotiated to a transport where the
3163 * UDP sources are not needed anymore, such as TCP or multicast. */
3165 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
3169 for (i = 0; i < 2; i++) {
3170 if (stream->udpsrc[i]) {
3171 GST_DEBUG ("free UDP source %d for stream %p", i, stream);
3172 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
3173 gst_object_unref (stream->udpsrc[i]);
3174 stream->udpsrc[i] = NULL;
3179 /* for TCP, create pads to send and receive data to and from the manager and to
3180 * intercept various events and queries
3183 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
3184 GstRTSPTransport * transport, GstPad ** outpad)
3187 GstPadTemplate *template;
3188 GstPad *pad0, *pad1;
3190 /* configure for interleaved delivery, nothing needs to be done
3191 * here, the loop function will call the chain functions of the
3192 * session manager. */
3193 stream->channel[0] = transport->interleaved.min;
3194 stream->channel[1] = transport->interleaved.max;
3195 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
3196 stream->channel[0], stream->channel[1]);
3198 /* we can remove the allocated UDP ports now */
3199 gst_rtspsrc_stream_free_udp (stream);
3201 /* no session manager, send data to srcpad directly */
3202 if (!stream->channelpad[0]) {
3203 GST_DEBUG_OBJECT (src, "no manager, creating pad");
3205 /* create a new pad we will use to stream to */
3206 name = g_strdup_printf ("stream_%u", stream->id);
3207 template = gst_static_pad_template_get (&rtptemplate);
3208 stream->channelpad[0] = gst_pad_new_from_template (template, name);
3209 gst_object_unref (template);
3212 /* set caps and activate */
3213 gst_pad_use_fixed_caps (stream->channelpad[0]);
3214 gst_pad_set_active (stream->channelpad[0], TRUE);
3216 *outpad = gst_object_ref (stream->channelpad[0]);
3218 GST_DEBUG_OBJECT (src, "using manager source pad");
3220 template = gst_static_pad_template_get (&anysrctemplate);
3222 /* allocate pads for sending the channel data into the manager */
3223 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
3224 gst_pad_link_full (pad0, stream->channelpad[0], GST_PAD_LINK_CHECK_NOTHING);
3225 gst_object_unref (stream->channelpad[0]);
3226 stream->channelpad[0] = pad0;
3227 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
3228 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
3229 gst_pad_set_element_private (pad0, src);
3230 gst_pad_set_active (pad0, TRUE);
3232 if (stream->channelpad[1]) {
3233 /* if we have a sinkpad for the other channel, create a pad and link to the
3235 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
3236 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
3237 gst_pad_link_full (pad1, stream->channelpad[1],
3238 GST_PAD_LINK_CHECK_NOTHING);
3239 gst_object_unref (stream->channelpad[1]);
3240 stream->channelpad[1] = pad1;
3241 gst_pad_set_active (pad1, TRUE);
3243 gst_object_unref (template);
3245 /* setup RTCP transport back to the server if we have to. */
3246 if (src->manager && src->do_rtcp) {
3249 template = gst_static_pad_template_get (&anysinktemplate);
3251 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
3252 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
3253 gst_pad_set_element_private (stream->rtcppad, stream);
3254 gst_pad_set_active (stream->rtcppad, TRUE);
3256 /* get session RTCP pad */
3257 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3258 pad = gst_element_get_request_pad (src->manager, name);
3263 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3264 gst_object_unref (pad);
3267 gst_object_unref (template);
3273 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
3274 GstRTSPTransport * transport, const gchar ** destination, gint * min,
3275 gint * max, guint * ttl)
3277 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3279 if (!(*destination = transport->destination))
3280 *destination = stream->destination;
3283 /* transport first */
3284 *min = transport->port.min;
3285 *max = transport->port.max;
3286 if (*min == -1 && *max == -1) {
3287 /* then try from SDP */
3288 if (stream->port != 0) {
3289 *min = stream->port;
3290 *max = stream->port + 1;
3296 if (!(*ttl = transport->ttl))
3301 /* first take the source, then the endpoint to figure out where to send
3303 if (!(*destination = transport->source)) {
3304 if (src->conninfo.connection)
3305 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
3306 else if (stream->conninfo.connection)
3308 gst_rtsp_connection_get_ip (stream->conninfo.connection);
3312 /* for unicast we only expect the ports here */
3313 *min = transport->server_port.min;
3314 *max = transport->server_port.max;
3319 /* For multicast create UDP sources and join the multicast group. */
3321 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
3322 GstRTSPTransport * transport, GstPad ** outpad)
3325 const gchar *destination;
3328 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
3330 /* we can remove the allocated UDP ports now */
3331 gst_rtspsrc_stream_free_udp (stream);
3333 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
3336 /* we need a destination now */
3337 if (destination == NULL)
3338 goto no_destination;
3340 /* we really need ports now or we won't be able to receive anything at all */
3341 if (min == -1 && max == -1)
3344 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
3345 destination, min, max);
3347 /* creating UDP source for RTP */
3349 uri = g_strdup_printf ("udp://%s:%d", destination, min);
3351 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3353 if (stream->udpsrc[0] == NULL)
3356 /* take ownership */
3357 gst_object_ref_sink (stream->udpsrc[0]);
3359 if (src->udp_buffer_size != 0)
3360 g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
3361 src->udp_buffer_size, NULL);
3363 if (src->multi_iface != NULL)
3364 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3365 src->multi_iface, NULL);
3368 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3369 gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
3372 /* creating another UDP source for RTCP */
3376 uri = g_strdup_printf ("udp://%s:%d", destination, max);
3378 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3380 if (stream->udpsrc[1] == NULL)
3383 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3384 stream->profile == GST_RTSP_PROFILE_SAVPF)
3385 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3387 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3388 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3389 gst_caps_unref (caps);
3391 /* take ownership */
3392 gst_object_ref_sink (stream->udpsrc[1]);
3394 if (src->multi_iface != NULL)
3395 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3396 src->multi_iface, NULL);
3398 gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
3405 GST_DEBUG_OBJECT (src, "no UDP source element found");
3410 GST_DEBUG_OBJECT (src, "no destination found");
3415 GST_DEBUG_OBJECT (src, "no ports found");
3420 /* configure the remainder of the UDP ports */
3422 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
3423 GstRTSPTransport * transport, GstPad ** outpad)
3425 /* we manage the UDP elements now. For unicast, the UDP sources where
3426 * allocated in the stream when we suggested a transport. */
3427 if (stream->udpsrc[0]) {
3430 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3431 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
3433 GST_DEBUG_OBJECT (src, "setting up UDP source");
3435 /* configure a timeout on the UDP port. When the timeout message is
3436 * posted, we assume UDP transport is not possible. We reconnect using TCP
3438 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
3439 src->udp_timeout * 1000, NULL);
3441 if ((caps = stream_get_caps_for_pt (stream, stream->default_pt)))
3442 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
3444 /* get output pad of the UDP source. */
3445 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
3447 /* save it so we can unblock */
3448 stream->blockedpad = *outpad;
3450 /* configure pad block on the pad. As soon as there is dataflow on the
3451 * UDP source, we know that UDP is not blocked by a firewall and we can
3452 * configure all the streams to let the application autoplug decoders. */
3454 gst_pad_add_probe (stream->blockedpad,
3455 GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, pad_blocked, src, NULL);
3457 if (stream->channelpad[0]) {
3458 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
3459 /* configure for UDP delivery, we need to connect the UDP pads to
3460 * the session plugin. */
3461 gst_pad_link_full (*outpad, stream->channelpad[0],
3462 GST_PAD_LINK_CHECK_NOTHING);
3463 gst_object_unref (*outpad);
3465 /* we connected to pad-added signal to get pads from the manager */
3467 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
3472 if (stream->udpsrc[1]) {
3475 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
3476 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
3478 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3479 stream->profile == GST_RTSP_PROFILE_SAVPF)
3480 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3482 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3483 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3484 gst_caps_unref (caps);
3486 if (stream->channelpad[1]) {
3489 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
3491 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
3492 gst_pad_link_full (pad, stream->channelpad[1],
3493 GST_PAD_LINK_CHECK_NOTHING);
3494 gst_object_unref (pad);
3496 /* leave unlinked */
3502 /* configure the UDP sink back to the server for status reports */
3504 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
3505 GstRTSPStream * stream, GstRTSPTransport * transport)
3508 gint rtp_port, rtcp_port;
3509 gboolean do_rtp, do_rtcp;
3510 const gchar *destination;
3515 /* get transport info */
3516 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
3517 &rtp_port, &rtcp_port, &ttl);
3519 /* see what we need to do */
3520 do_rtp = (rtp_port != -1);
3521 /* it's possible that the server does not want us to send RTCP in which case
3523 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
3525 /* we need a destination when we have RTP or RTCP ports */
3526 if (destination == NULL && (do_rtp || do_rtcp))
3527 goto no_destination;
3529 /* try to construct the fakesrc to the RTP port of the server to open up any
3532 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
3535 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
3536 stream->udpsink[0] =
3537 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3539 if (stream->udpsink[0] == NULL)
3540 goto no_sink_element;
3542 /* don't join multicast group, we will have the source socket do that */
3543 /* no sync or async state changes needed */
3544 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
3545 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3547 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3549 if (stream->udpsrc[0]) {
3550 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
3551 * so that NAT firewalls will open a hole for us */
3552 g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
3553 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
3554 /* configure socket and make sure udpsink does not close it when shutting
3555 * down, it belongs to udpsrc after all. */
3556 g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
3557 "close-socket", FALSE, NULL);
3558 g_object_unref (socket);
3561 /* the source for the dummy packets to open up NAT */
3562 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
3563 if (stream->fakesrc == NULL)
3564 goto no_fakesrc_element;
3566 /* random data in 5 buffers, a size of 200 bytes should be fine */
3567 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
3568 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
3570 /* we don't want to consider this a sink */
3571 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
3573 /* keep everything locked */
3574 gst_element_set_locked_state (stream->udpsink[0], TRUE);
3575 gst_element_set_locked_state (stream->fakesrc, TRUE);
3577 gst_object_ref (stream->udpsink[0]);
3578 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
3579 gst_object_ref (stream->fakesrc);
3580 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
3582 gst_element_link_pads_full (stream->fakesrc, "src", stream->udpsink[0],
3583 "sink", GST_PAD_LINK_CHECK_NOTHING);
3586 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
3589 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
3590 stream->udpsink[1] =
3591 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3593 if (stream->udpsink[1] == NULL)
3594 goto no_sink_element;
3596 /* don't join multicast group, we will have the source socket do that */
3597 /* no sync or async state changes needed */
3598 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
3599 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3601 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3603 if (stream->udpsrc[1]) {
3604 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
3605 * because some servers check the port number of where it sends RTCP to identify
3606 * the RTCP packets it receives */
3607 g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
3608 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
3609 /* configure socket and make sure udpsink does not close it when shutting
3610 * down, it belongs to udpsrc after all. */
3611 g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
3612 "close-socket", FALSE, NULL);
3613 g_object_unref (socket);
3616 /* we don't want to consider this a sink */
3617 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
3619 /* we keep this playing always */
3620 gst_element_set_locked_state (stream->udpsink[1], TRUE);
3621 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
3623 gst_object_ref (stream->udpsink[1]);
3624 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
3626 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
3628 /* get session RTCP pad */
3629 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3630 pad = gst_element_get_request_pad (src->manager, name);
3635 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3636 gst_object_unref (pad);
3645 GST_DEBUG_OBJECT (src, "no destination address specified");
3650 GST_DEBUG_OBJECT (src, "no UDP sink element found");
3655 GST_DEBUG_OBJECT (src, "no fakesrc element found");
3660 /* sets up all elements needed for streaming over the specified transport.
3661 * Does not yet expose the element pads, this will be done when there is actuall
3662 * dataflow detected, which might never happen when UDP is blocked in a
3663 * firewall, for example.
3666 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
3667 GstRTSPTransport * transport)
3670 GstPad *outpad = NULL;
3671 GstPadTemplate *template;
3673 const gchar *media_type;
3676 src = stream->parent;
3678 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
3680 /* get the proper media type for this stream now */
3681 if (gst_rtsp_transport_get_media_type (transport, &media_type) < 0)
3682 goto unknown_transport;
3684 goto unknown_transport;
3686 /* configure the final media type */
3687 GST_DEBUG_OBJECT (src, "setting media type to %s", media_type);
3689 len = stream->ptmap->len;
3690 for (i = 0; i < len; i++) {
3692 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
3694 if (item->caps == NULL)
3697 s = gst_caps_get_structure (item->caps, 0);
3698 gst_structure_set_name (s, media_type);
3699 /* set ssrc if known */
3700 if (transport->ssrc)
3701 gst_structure_set (s, "ssrc", G_TYPE_UINT, transport->ssrc, NULL);
3704 /* try to get and configure a manager, channelpad[0-1] will be configured with
3705 * the pads for the manager, or NULL when no manager is needed. */
3706 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
3709 switch (transport->lower_transport) {
3710 case GST_RTSP_LOWER_TRANS_TCP:
3711 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
3712 goto transport_failed;
3714 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3715 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
3716 goto transport_failed;
3717 /* fallthrough, the rest is the same for UDP and MCAST */
3718 case GST_RTSP_LOWER_TRANS_UDP:
3719 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
3720 goto transport_failed;
3721 /* configure udpsinks back to the server for RTCP messages and for the
3722 * dummy RTP messages to open NAT. */
3723 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
3724 goto transport_failed;
3727 goto unknown_transport;
3731 GST_DEBUG_OBJECT (src, "creating ghostpad");
3733 gst_pad_use_fixed_caps (outpad);
3735 /* create ghostpad, don't add just yet, this will be done when we activate
3737 name = g_strdup_printf ("stream_%u", stream->id);
3738 template = gst_static_pad_template_get (&rtptemplate);
3739 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
3740 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
3741 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
3742 gst_object_unref (template);
3745 gst_object_unref (outpad);
3747 /* mark pad as ok */
3748 stream->last_ret = GST_FLOW_OK;
3755 GST_DEBUG_OBJECT (src, "failed to configure transport");
3760 GST_DEBUG_OBJECT (src, "unknown transport");
3765 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3770 /* send a couple of dummy random packets on the receiver RTP port to the server,
3771 * this should make a firewall think we initiated the data transfer and
3772 * hopefully allow packets to go from the sender port to our RTP receiver port */
3774 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
3778 if (src->nat_method != GST_RTSP_NAT_DUMMY)
3781 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3782 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3784 if (stream->fakesrc && stream->udpsink[0]) {
3785 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
3786 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
3787 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
3788 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
3789 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
3795 /* Adds the source pads of all configured streams to the element.
3796 * This code is performed when we detected dataflow.
3798 * We detect dataflow from either the _loop function or with pad probes on the
3802 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
3806 GST_DEBUG_OBJECT (src, "activating streams");
3808 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3809 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3811 if (stream->udpsrc[0]) {
3812 /* remove timeout, we are streaming now and timeouts will be handled by
3813 * the session manager and jitter buffer */
3814 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
3816 if (stream->srcpad) {
3817 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
3818 gst_pad_set_active (stream->srcpad, TRUE);
3820 /* if we don't have a session manager, set the caps now. If we have a
3821 * session, we will get a notification of the pad and the caps. */
3822 if (!src->manager) {
3825 caps = stream_get_caps_for_pt (stream, stream->default_pt);
3826 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
3827 gst_pad_set_caps (stream->srcpad, caps);
3830 if (!stream->added) {
3831 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
3832 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
3833 stream->added = TRUE;
3838 /* unblock all pads */
3839 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3840 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3842 if (stream->blockid) {
3843 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
3844 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
3845 stream->blockid = 0;
3853 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
3854 gboolean reset_manager)
3857 guint64 start, stop;
3858 gdouble play_speed, play_scale;
3860 GST_DEBUG_OBJECT (src, "configuring stream caps");
3862 start = segment->position;
3863 stop = segment->duration;
3864 play_speed = segment->rate;
3865 play_scale = segment->applied_rate;
3867 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3868 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3874 len = stream->ptmap->len;
3875 for (j = 0; j < len; j++) {
3877 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, j);
3879 if (item->caps == NULL)
3882 caps = gst_caps_make_writable (item->caps);
3884 if (stream->timebase != -1)
3885 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
3886 (guint) stream->timebase, NULL);
3887 if (stream->seqbase != -1)
3888 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
3889 (guint) stream->seqbase, NULL);
3890 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
3892 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
3893 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
3894 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
3897 GST_DEBUG_OBJECT (src, "stream %p, pt %d, caps %" GST_PTR_FORMAT, stream,
3900 if (item->pt == stream->default_pt && stream->udpsrc[0]) {
3901 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
3905 if (reset_manager && src->manager) {
3906 GST_DEBUG_OBJECT (src, "clear session");
3907 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
3911 static GstFlowReturn
3912 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
3917 /* store the value */
3918 stream->last_ret = ret;
3920 /* if it's success we can return the value right away */
3921 if (ret == GST_FLOW_OK)
3924 /* any other error that is not-linked can be returned right
3926 if (ret != GST_FLOW_NOT_LINKED)
3929 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
3930 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3931 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3933 ret = ostream->last_ret;
3934 /* some other return value (must be SUCCESS but we can return
3935 * other values as well) */
3936 if (ret != GST_FLOW_NOT_LINKED)
3939 /* if we get here, all other pads were unlinked and we return
3940 * NOT_LINKED then */
3946 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
3949 gboolean res = TRUE;
3951 /* only streams that have a connection to the outside world */
3955 if (stream->udpsrc[0]) {
3956 gst_event_ref (event);
3957 res = gst_element_send_event (stream->udpsrc[0], event);
3958 } else if (stream->channelpad[0]) {
3959 gst_event_ref (event);
3960 if (GST_PAD_IS_SRC (stream->channelpad[0]))
3961 res = gst_pad_push_event (stream->channelpad[0], event);
3963 res = gst_pad_send_event (stream->channelpad[0], event);
3966 if (stream->udpsrc[1]) {
3967 gst_event_ref (event);
3968 res &= gst_element_send_event (stream->udpsrc[1], event);
3969 } else if (stream->channelpad[1]) {
3970 gst_event_ref (event);
3971 if (GST_PAD_IS_SRC (stream->channelpad[1]))
3972 res &= gst_pad_push_event (stream->channelpad[1], event);
3974 res &= gst_pad_send_event (stream->channelpad[1], event);
3978 gst_event_unref (event);
3984 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
3987 gboolean res = TRUE;
3989 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3990 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3992 gst_event_ref (event);
3993 res &= gst_rtspsrc_stream_push_event (src, ostream, event);
3995 gst_event_unref (event);
4000 static GstRTSPResult
4001 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
4006 if (info->connection == NULL) {
4007 if (info->url == NULL) {
4008 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
4009 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
4013 /* create connection */
4014 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
4015 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
4016 goto could_not_create;
4019 g_free (info->url_str);
4020 info->url_str = gst_rtsp_url_get_request_uri (info->url);
4022 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
4024 if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
4025 if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
4026 src->tls_validation_flags))
4027 GST_WARNING_OBJECT (src, "Unable to set TLS validation flags");
4029 if (src->tls_database)
4030 gst_rtsp_connection_set_tls_database (info->connection,
4034 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
4035 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
4037 if (src->proxy_host) {
4038 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
4040 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
4045 if (!info->connected) {
4048 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
4049 ("Connecting to %s", info->location));
4050 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
4052 gst_rtsp_connection_connect (info->connection,
4053 src->ptcp_timeout)) < 0)
4054 goto could_not_connect;
4056 info->connected = TRUE;
4063 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
4068 gchar *str = gst_rtsp_strresult (res);
4069 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
4075 gchar *str = gst_rtsp_strresult (res);
4076 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
4082 static GstRTSPResult
4083 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
4086 GST_RTSP_STATE_LOCK (src);
4087 if (info->connected) {
4088 GST_DEBUG_OBJECT (src, "closing connection...");
4089 gst_rtsp_connection_close (info->connection);
4090 info->connected = FALSE;
4092 if (free && info->connection) {
4093 /* free connection */
4094 GST_DEBUG_OBJECT (src, "freeing connection...");
4095 gst_rtsp_connection_free (info->connection);
4096 info->connection = NULL;
4098 GST_RTSP_STATE_UNLOCK (src);
4102 static GstRTSPResult
4103 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
4108 GST_DEBUG_OBJECT (src, "reconnecting connection...");
4109 gst_rtsp_conninfo_close (src, info, FALSE);
4110 res = gst_rtsp_conninfo_connect (src, info, async);
4116 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
4120 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
4121 GST_RTSP_STATE_LOCK (src);
4122 if (src->conninfo.connection && src->conninfo.flushing != flush) {
4123 GST_DEBUG_OBJECT (src, "connection flush");
4124 gst_rtsp_connection_flush (src->conninfo.connection, flush);
4125 src->conninfo.flushing = flush;
4127 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4128 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4129 if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
4130 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
4131 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
4132 stream->conninfo.flushing = flush;
4135 GST_RTSP_STATE_UNLOCK (src);
4138 /* FIXME, handle server request, reply with OK, for now */
4139 static GstRTSPResult
4140 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
4141 GstRTSPMessage * request)
4143 GstRTSPMessage response = { 0 };
4146 GST_DEBUG_OBJECT (src, "got server request message");
4149 gst_rtsp_message_dump (request);
4151 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
4153 if (res == GST_RTSP_ENOTIMPL) {
4154 /* default implementation, send OK */
4155 GST_DEBUG_OBJECT (src, "prepare OK reply");
4157 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
4162 /* let app parse and reply */
4163 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST],
4164 0, request, &response);
4167 gst_rtsp_message_dump (&response);
4169 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
4173 gst_rtsp_message_unset (&response);
4174 } else if (res == GST_RTSP_EEOF)
4182 gst_rtsp_message_unset (&response);
4187 /* send server keep-alive */
4188 static GstRTSPResult
4189 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
4191 GstRTSPMessage request = { 0 };
4193 GstRTSPMethod method;
4194 const gchar *control;
4196 if (src->do_rtsp_keep_alive == FALSE) {
4197 GST_DEBUG_OBJECT (src, "do-rtsp-keep-alive is FALSE, not sending.");
4198 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4202 GST_DEBUG_OBJECT (src, "creating server keep-alive");
4204 /* find a method to use for keep-alive */
4205 if (src->methods & GST_RTSP_GET_PARAMETER)
4206 method = GST_RTSP_GET_PARAMETER;
4208 method = GST_RTSP_OPTIONS;
4210 control = get_aggregate_control (src);
4211 if (control == NULL)
4214 res = gst_rtsp_message_init_request (&request, method, control);
4219 gst_rtsp_message_dump (&request);
4222 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
4227 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4228 gst_rtsp_message_unset (&request);
4235 GST_WARNING_OBJECT (src, "no control url to send keepalive");
4240 gchar *str = gst_rtsp_strresult (res);
4242 gst_rtsp_message_unset (&request);
4243 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
4244 ("Could not send keep-alive. (%s)", str));
4250 static GstFlowReturn
4251 gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
4253 GstFlowReturn ret = GST_FLOW_OK;
4255 GstRTSPStream *stream;
4256 GstPad *outpad = NULL;
4263 channel = message->type_data.data.channel;
4265 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
4267 goto unknown_stream;
4269 if (channel == stream->channel[0]) {
4270 outpad = stream->channelpad[0];
4272 } else if (channel == stream->channel[1]) {
4273 outpad = stream->channelpad[1];
4279 /* take a look at the body to figure out what we have */
4280 gst_rtsp_message_get_body (message, &data, &size);
4282 goto invalid_length;
4284 /* channels are not correct on some servers, do extra check */
4285 if (data[1] >= 200 && data[1] <= 204) {
4286 /* hmm RTCP message switch to the RTCP pad of the same stream. */
4287 outpad = stream->channelpad[1];
4291 /* we have no clue what this is, just ignore then. */
4293 goto unknown_stream;
4295 /* take the message body for further processing */
4296 gst_rtsp_message_steal_body (message, &data, &size);
4298 /* strip the trailing \0 */
4301 buf = gst_buffer_new ();
4302 gst_buffer_append_memory (buf,
4303 gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
4305 /* don't need message anymore */
4306 gst_rtsp_message_unset (message);
4308 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
4311 if (src->need_activate) {
4317 guint group_id = gst_util_group_id_next ();
4319 /* generate an SHA256 sum of the URI */
4320 cs = g_checksum_new (G_CHECKSUM_SHA256);
4321 uri = src->conninfo.location;
4322 g_checksum_update (cs, (const guchar *) uri, strlen (uri));
4324 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4325 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4328 g_strdup_printf ("%s/%d", g_checksum_get_string (cs), ostream->id);
4329 event = gst_event_new_stream_start (stream_id);
4330 gst_event_set_group_id (event, group_id);
4333 gst_rtspsrc_stream_push_event (src, ostream, event);
4335 g_checksum_free (cs);
4337 gst_rtspsrc_activate_streams (src);
4338 src->need_activate = FALSE;
4340 if ((event = src->start_segment) != NULL) {
4341 src->start_segment = NULL;
4342 gst_rtspsrc_push_event (src, event);
4345 if (src->base_time == -1) {
4346 /* Take current running_time. This timestamp will be put on
4347 * the first buffer of each stream because we are a live source and so we
4348 * timestamp with the running_time. When we are dealing with TCP, we also
4349 * only timestamp the first buffer (using the DISCONT flag) because a server
4350 * typically bursts data, for which we don't want to compensate by speeding
4351 * up the media. The other timestamps will be interpollated from this one
4352 * using the RTP timestamps. */
4353 GST_OBJECT_LOCK (src);
4354 if (GST_ELEMENT_CLOCK (src)) {
4356 GstClockTime base_time;
4358 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
4359 base_time = GST_ELEMENT_CAST (src)->base_time;
4361 src->base_time = now - base_time;
4363 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
4364 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
4366 GST_OBJECT_UNLOCK (src);
4369 if (stream->discont && !is_rtcp) {
4370 /* mark first RTP buffer as discont */
4371 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
4372 stream->discont = FALSE;
4373 /* first buffer gets the timestamp, other buffers are not timestamped and
4374 * their presentation time will be interpollated from the rtp timestamps. */
4375 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
4376 GST_TIME_ARGS (src->base_time));
4378 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
4381 /* chain to the peer pad */
4382 if (GST_PAD_IS_SINK (outpad))
4383 ret = gst_pad_chain (outpad, buf);
4385 ret = gst_pad_push (outpad, buf);
4388 /* combine all stream flows for the data transport */
4389 ret = gst_rtspsrc_combine_flows (src, stream, ret);
4396 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
4397 gst_rtsp_message_unset (message);
4402 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4403 ("Short message received, ignoring."));
4404 gst_rtsp_message_unset (message);
4409 static GstFlowReturn
4410 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
4412 GstRTSPMessage message = { 0 };
4414 GstFlowReturn ret = GST_FLOW_OK;
4415 GTimeVal tv_timeout;
4418 /* get the next timeout interval */
4419 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4421 /* see if the timeout period expired */
4422 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
4423 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
4424 /* send keep-alive, only act on interrupt, a warning will be posted for
4426 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4428 /* get new timeout */
4429 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4432 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
4433 tv_timeout.tv_sec, tv_timeout.tv_usec);
4435 /* protect the connection with the connection lock so that we can see when
4436 * we are finished doing server communication */
4438 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4439 &message, src->ptcp_timeout);
4443 GST_DEBUG_OBJECT (src, "we received a server message");
4445 case GST_RTSP_EINTR:
4446 /* we got interrupted this means we need to stop */
4448 case GST_RTSP_ETIMEOUT:
4449 /* no reply, send keep alive */
4450 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4451 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4455 /* go EOS when the server closed the connection */
4461 switch (message.type) {
4462 case GST_RTSP_MESSAGE_REQUEST:
4463 /* server sends us a request message, handle it */
4465 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4467 if (res == GST_RTSP_EEOF)
4470 goto handle_request_failed;
4472 case GST_RTSP_MESSAGE_RESPONSE:
4473 /* we ignore response messages */
4474 GST_DEBUG_OBJECT (src, "ignoring response message");
4476 gst_rtsp_message_dump (&message);
4478 case GST_RTSP_MESSAGE_DATA:
4479 GST_DEBUG_OBJECT (src, "got data message");
4480 ret = gst_rtspsrc_handle_data (src, &message);
4481 if (ret != GST_FLOW_OK)
4482 goto handle_data_failed;
4485 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4490 g_assert_not_reached ();
4495 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4496 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4497 ("The server closed the connection."));
4498 src->conninfo.connected = FALSE;
4499 gst_rtsp_message_unset (&message);
4500 return GST_FLOW_EOS;
4504 gst_rtsp_message_unset (&message);
4505 GST_DEBUG_OBJECT (src, "got interrupted");
4506 return GST_FLOW_FLUSHING;
4510 gchar *str = gst_rtsp_strresult (res);
4512 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4513 ("Could not receive message. (%s)", str));
4516 gst_rtsp_message_unset (&message);
4517 return GST_FLOW_ERROR;
4519 handle_request_failed:
4521 gchar *str = gst_rtsp_strresult (res);
4523 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4524 ("Could not handle server message. (%s)", str));
4526 gst_rtsp_message_unset (&message);
4527 return GST_FLOW_ERROR;
4531 GST_DEBUG_OBJECT (src, "could no handle data message");
4536 static GstFlowReturn
4537 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
4540 GstRTSPMessage message = { 0 };
4544 GTimeVal tv_timeout;
4546 /* get the next timeout interval */
4547 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4549 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
4550 (gint) tv_timeout.tv_sec);
4552 gst_rtsp_message_unset (&message);
4554 /* we should continue reading the TCP socket because the server might
4555 * send us requests. When the session timeout expires, we need to send a
4556 * keep-alive request to keep the session open. */
4557 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4558 &message, &tv_timeout);
4562 GST_DEBUG_OBJECT (src, "we received a server message");
4564 case GST_RTSP_EINTR:
4565 /* we got interrupted, see what we have to do */
4567 case GST_RTSP_ETIMEOUT:
4568 /* send keep-alive, ignore the result, a warning will be posted. */
4569 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4570 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4574 /* server closed the connection. not very fatal for UDP, reconnect and
4575 * see what happens. */
4576 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4577 ("The server closed the connection."));
4578 if (src->udp_reconnect) {
4580 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
4587 GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
4589 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4590 ("Unhandled return value %d.", res));
4594 switch (message.type) {
4595 case GST_RTSP_MESSAGE_REQUEST:
4596 /* server sends us a request message, handle it */
4598 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4600 if (res == GST_RTSP_EEOF)
4603 goto handle_request_failed;
4605 case GST_RTSP_MESSAGE_RESPONSE:
4606 /* we ignore response and data messages */
4607 GST_DEBUG_OBJECT (src, "ignoring response message");
4609 gst_rtsp_message_dump (&message);
4610 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
4611 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
4612 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
4613 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
4614 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4621 case GST_RTSP_MESSAGE_DATA:
4622 /* we ignore response and data messages */
4623 GST_DEBUG_OBJECT (src, "ignoring data message");
4626 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4631 g_assert_not_reached ();
4633 /* we get here when the connection got interrupted */
4636 gst_rtsp_message_unset (&message);
4637 GST_DEBUG_OBJECT (src, "got interrupted");
4638 return GST_FLOW_FLUSHING;
4642 gchar *str = gst_rtsp_strresult (res);
4645 src->conninfo.connected = FALSE;
4646 if (res != GST_RTSP_EINTR) {
4647 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
4648 ("Could not connect to server. (%s)", str));
4650 ret = GST_FLOW_ERROR;
4652 ret = GST_FLOW_FLUSHING;
4658 gchar *str = gst_rtsp_strresult (res);
4660 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4661 ("Could not receive message. (%s)", str));
4663 return GST_FLOW_ERROR;
4665 handle_request_failed:
4667 gchar *str = gst_rtsp_strresult (res);
4670 gst_rtsp_message_unset (&message);
4671 if (res != GST_RTSP_EINTR) {
4672 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4673 ("Could not handle server message. (%s)", str));
4675 ret = GST_FLOW_ERROR;
4677 ret = GST_FLOW_FLUSHING;
4683 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4684 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4685 ("The server closed the connection."));
4686 src->conninfo.connected = FALSE;
4687 gst_rtsp_message_unset (&message);
4688 return GST_FLOW_EOS;
4692 static GstRTSPResult
4693 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
4695 GstRTSPResult res = GST_RTSP_OK;
4698 GST_DEBUG_OBJECT (src, "doing reconnect");
4700 GST_OBJECT_LOCK (src);
4701 /* only restart when the pads were not yet activated, else we were
4702 * streaming over UDP */
4703 restart = src->need_activate;
4704 GST_OBJECT_UNLOCK (src);
4706 /* no need to restart, we're done */
4710 /* we can try only TCP now */
4711 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
4713 /* close and cleanup our state */
4714 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
4717 /* see if we have TCP left to try. Also don't try TCP when we were configured
4719 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
4722 /* We post a warning message now to inform the user
4723 * that nothing happened. It's most likely a firewall thing. */
4724 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4725 ("Could not receive any UDP packets for %.4f seconds, maybe your "
4726 "firewall is blocking it. Retrying using a TCP connection.",
4727 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
4729 /* open new connection using tcp */
4730 if (gst_rtspsrc_open (src, async) < 0)
4733 /* start playback */
4734 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
4743 src->cur_protocols = 0;
4744 /* no transport possible, post an error and stop */
4745 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4746 ("Could not receive any UDP packets for %.4f seconds, maybe your "
4747 "firewall is blocking it. No other protocols to try.",
4748 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
4749 return GST_RTSP_ERROR;
4753 GST_DEBUG_OBJECT (src, "open failed");
4758 GST_DEBUG_OBJECT (src, "play failed");
4764 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
4768 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
4771 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
4774 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
4777 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
4785 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
4789 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
4792 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
4795 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
4798 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
4806 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
4810 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
4813 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
4816 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
4819 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
4827 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
4831 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
4834 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
4837 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
4840 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
4848 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
4850 if (ret == GST_RTSP_OK)
4851 gst_rtspsrc_loop_complete_cmd (src, cmd);
4852 else if (ret == GST_RTSP_EINTR)
4853 gst_rtspsrc_loop_cancel_cmd (src, cmd);
4855 gst_rtspsrc_loop_error_cmd (src, cmd);
4859 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
4862 gboolean flushed = FALSE;
4864 /* start new request */
4865 gst_rtspsrc_loop_start_cmd (src, cmd);
4867 GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
4869 GST_OBJECT_LOCK (src);
4870 old = src->pending_cmd;
4871 if (old == CMD_RECONNECT) {
4872 GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
4873 cmd = CMD_RECONNECT;
4875 if (old != CMD_WAIT) {
4876 src->pending_cmd = CMD_WAIT;
4877 GST_OBJECT_UNLOCK (src);
4878 /* cancel previous request */
4879 GST_DEBUG_OBJECT (src, "cancel previous request %d", old);
4880 gst_rtspsrc_loop_cancel_cmd (src, old);
4881 GST_OBJECT_LOCK (src);
4883 src->pending_cmd = cmd;
4884 /* interrupt if allowed */
4885 if (src->busy_cmd & mask) {
4886 GST_DEBUG_OBJECT (src, "connection flush busy %d", src->busy_cmd);
4887 gst_rtspsrc_connection_flush (src, TRUE);
4890 GST_DEBUG_OBJECT (src, "not interrupting busy cmd %d", src->busy_cmd);
4893 gst_task_start (src->task);
4894 GST_OBJECT_UNLOCK (src);
4900 gst_rtspsrc_loop (GstRTSPSrc * src)
4904 if (!src->conninfo.connection || !src->conninfo.connected)
4907 if (src->interleaved)
4908 ret = gst_rtspsrc_loop_interleaved (src);
4910 ret = gst_rtspsrc_loop_udp (src);
4912 if (ret != GST_FLOW_OK)
4920 GST_WARNING_OBJECT (src, "we are not connected");
4921 ret = GST_FLOW_FLUSHING;
4926 const gchar *reason = gst_flow_get_name (ret);
4928 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
4929 src->running = FALSE;
4930 if (ret == GST_FLOW_EOS) {
4931 /* perform EOS logic */
4932 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
4933 gst_element_post_message (GST_ELEMENT_CAST (src),
4934 gst_message_new_segment_done (GST_OBJECT_CAST (src),
4935 src->segment.format, src->segment.position));
4936 gst_rtspsrc_push_event (src,
4937 gst_event_new_segment_done (src->segment.format,
4938 src->segment.position));
4940 gst_rtspsrc_push_event (src, gst_event_new_eos ());
4942 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
4943 /* for fatal errors we post an error message, post the error before the
4944 * EOS so the app knows about the error first. */
4945 GST_ELEMENT_ERROR (src, STREAM, FAILED,
4946 ("Internal data flow error."),
4947 ("streaming task paused, reason %s (%d)", reason, ret));
4948 gst_rtspsrc_push_event (src, gst_event_new_eos ());
4950 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_LOOP);
4955 #ifndef GST_DISABLE_GST_DEBUG
4956 static const gchar *
4957 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
4961 while (method != 0) {
4978 static const gchar *
4979 gst_rtspsrc_skip_lws (const gchar * s)
4981 while (g_ascii_isspace (*s))
4986 static const gchar *
4987 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
4989 while (s > start && g_ascii_isspace (*(s - 1)))
4994 static const gchar *
4995 gst_rtspsrc_skip_commas (const gchar * s)
4997 /* The grammar allows for multiple commas */
4998 while (g_ascii_isspace (*s) || *s == ',')
5003 static const gchar *
5004 gst_rtspsrc_skip_item (const gchar * s)
5006 gboolean quoted = FALSE;
5007 const gchar *start = s;
5009 /* A list item ends at the last non-whitespace character
5010 * before a comma which is not inside a quoted-string. Or at
5011 * the end of the string.
5017 if (*s == '\\' && *(s + 1))
5026 return gst_rtspsrc_unskip_lws (s, start);
5030 gst_rtsp_decode_quoted_string (gchar * quoted_string)
5034 src = quoted_string + 1;
5035 dst = quoted_string;
5036 while (*src && *src != '"') {
5037 if (*src == '\\' && *(src + 1))
5044 /* Extract the authentication tokens that the server provided for each method
5045 * into an array of structures and give those to the connection object.
5048 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
5049 const gchar * header, gboolean * stale)
5051 GSList *list = NULL, *iter;
5053 gchar *item, *eq, *name_end, *value;
5055 g_return_if_fail (stale != NULL);
5057 gst_rtsp_connection_clear_auth_params (conn);
5060 /* Parse a header whose content is described by RFC2616 as
5061 * "#something", where "something" does not itself contain commas,
5062 * except as part of quoted-strings, into a list of allocated strings.
5064 header = gst_rtspsrc_skip_commas (header);
5066 end = gst_rtspsrc_skip_item (header);
5067 list = g_slist_prepend (list, g_strndup (header, end - header));
5068 header = gst_rtspsrc_skip_commas (end);
5073 list = g_slist_reverse (list);
5074 for (iter = list; iter; iter = iter->next) {
5077 eq = strchr (item, '=');
5079 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
5080 if (name_end == item) {
5081 /* That's no good... */
5088 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
5090 gst_rtsp_decode_quoted_string (value);
5094 if (item && (strcmp (item, "stale") == 0) &&
5095 value && (strcmp (value, "TRUE") == 0))
5097 gst_rtsp_connection_set_auth_param (conn, item, value);
5101 g_slist_free (list);
5104 /* Parse a WWW-Authenticate Response header and determine the
5105 * available authentication methods
5107 * This code should also cope with the fact that each WWW-Authenticate
5108 * header can contain multiple challenge methods + tokens
5110 * At the moment, for Basic auth, we just do a minimal check and don't
5111 * even parse out the realm */
5113 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
5114 GstRTSPConnection * conn, gboolean * stale)
5118 g_return_if_fail (hdr != NULL);
5119 g_return_if_fail (methods != NULL);
5120 g_return_if_fail (stale != NULL);
5122 /* Skip whitespace at the start of the string */
5123 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
5125 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
5126 *methods |= GST_RTSP_AUTH_BASIC;
5127 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
5128 *methods |= GST_RTSP_AUTH_DIGEST;
5129 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
5134 * gst_rtspsrc_setup_auth:
5135 * @src: the rtsp source
5137 * Configure a username and password and auth method on the
5138 * connection object based on a response we received from the
5141 * Currently, this requires that a username and password were supplied
5142 * in the uri. In the future, they may be requested on demand by sending
5143 * a message up the bus.
5145 * Returns: TRUE if authentication information could be set up correctly.
5148 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
5152 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
5153 GstRTSPAuthMethod method;
5154 GstRTSPResult auth_result;
5156 GstRTSPConnection *conn;
5158 gboolean stale = FALSE;
5160 conn = src->conninfo.connection;
5162 /* Identify the available auth methods and see if any are supported */
5163 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
5164 &hdr, 0) == GST_RTSP_OK) {
5165 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
5168 if (avail_methods == GST_RTSP_AUTH_NONE)
5169 goto no_auth_available;
5171 /* For digest auth, if the response indicates that the session
5172 * data are stale, we just update them in the connection object and
5173 * return TRUE to retry the request */
5175 src->tried_url_auth = FALSE;
5177 url = gst_rtsp_connection_get_url (conn);
5179 /* Do we have username and password available? */
5180 if (url != NULL && !src->tried_url_auth && url->user != NULL
5181 && url->passwd != NULL) {
5184 src->tried_url_auth = TRUE;
5185 GST_DEBUG_OBJECT (src,
5186 "Attempting authentication using credentials from the URL");
5188 user = src->user_id;
5189 pass = src->user_pw;
5190 GST_DEBUG_OBJECT (src,
5191 "Attempting authentication using credentials from the properties");
5194 /* FIXME: If the url didn't contain username and password or we tried them
5195 * already, request a username and passwd from the application via some kind
5196 * of credentials request message */
5198 /* If we don't have a username and passwd at this point, bail out. */
5199 if (user == NULL || pass == NULL)
5202 /* Try to configure for each available authentication method, strongest to
5204 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
5205 /* Check if this method is available on the server */
5206 if ((method & avail_methods) == 0)
5209 /* Pass the credentials to the connection to try on the next request */
5210 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
5211 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
5212 * ignore it and end up retrying later */
5213 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
5214 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
5215 gst_rtsp_auth_method_to_string (method));
5220 if (method == GST_RTSP_AUTH_NONE)
5221 goto no_auth_available;
5227 /* Output an error indicating that we couldn't connect because there were
5228 * no supported authentication protocols */
5229 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5230 ("No supported authentication protocol was found"));
5235 /* We don't fire an error message, we just return FALSE and let the
5236 * normal NOT_AUTHORIZED error be propagated */
5241 static GstRTSPResult
5242 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5243 GstRTSPMessage * request, GstRTSPMessage * response,
5244 GstRTSPStatusCode * code)
5247 GstRTSPStatusCode thecode;
5248 gchar *content_base = NULL;
5252 if (!src->short_header)
5253 gst_rtsp_ext_list_before_send (src->extensions, request);
5255 GST_DEBUG_OBJECT (src, "sending message");
5258 gst_rtsp_message_dump (request);
5260 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
5264 gst_rtsp_connection_reset_timeout (conn);
5267 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
5272 gst_rtsp_message_dump (response);
5274 switch (response->type) {
5275 case GST_RTSP_MESSAGE_REQUEST:
5276 res = gst_rtspsrc_handle_request (src, conn, response);
5277 if (res == GST_RTSP_EEOF)
5280 goto handle_request_failed;
5282 case GST_RTSP_MESSAGE_RESPONSE:
5283 /* ok, a response is good */
5284 GST_DEBUG_OBJECT (src, "received response message");
5286 case GST_RTSP_MESSAGE_DATA:
5287 /* get next response */
5288 GST_DEBUG_OBJECT (src, "handle data response message");
5289 gst_rtspsrc_handle_data (src, response);
5292 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
5297 thecode = response->type_data.response.code;
5299 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
5301 /* if the caller wanted the result code, we store it. */
5305 /* If the request didn't succeed, bail out before doing any more */
5306 if (thecode != GST_RTSP_STS_OK)
5309 /* store new content base if any */
5310 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
5313 g_free (src->content_base);
5314 src->content_base = g_strdup (content_base);
5316 gst_rtsp_ext_list_after_send (src->extensions, request, response);
5323 gchar *str = gst_rtsp_strresult (res);
5325 if (res != GST_RTSP_EINTR) {
5326 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5327 ("Could not send message. (%s)", str));
5329 GST_WARNING_OBJECT (src, "send interrupted");
5338 GST_WARNING_OBJECT (src, "server closed connection");
5339 if ((try == 0) && !src->interleaved && src->udp_reconnect) {
5341 /* if reconnect succeeds, try again */
5343 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
5347 /* only try once after reconnect, then fallthrough and error out */
5350 gchar *str = gst_rtsp_strresult (res);
5352 if (res != GST_RTSP_EINTR) {
5353 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5354 ("Could not receive message. (%s)", str));
5356 GST_WARNING_OBJECT (src, "receive interrupted");
5364 handle_request_failed:
5366 /* ERROR was posted */
5367 gst_rtsp_message_unset (response);
5372 GST_DEBUG_OBJECT (src, "we got an eof from the server");
5373 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5374 ("The server closed the connection."));
5375 gst_rtsp_message_unset (response);
5382 * @src: the rtsp source
5383 * @conn: the connection to send on
5384 * @request: must point to a valid request
5385 * @response: must point to an empty #GstRTSPMessage
5386 * @code: an optional code result
5388 * send @request and retrieve the response in @response. optionally @code can be
5389 * non-NULL in which case it will contain the status code of the response.
5391 * If This function returns #GST_RTSP_OK, @response will contain a valid response
5392 * message that should be cleaned with gst_rtsp_message_unset() after usage.
5394 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
5395 * @response message) if the response code was not 200 (OK).
5397 * If the attempt results in an authentication failure, then this will attempt
5398 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
5401 * Returns: #GST_RTSP_OK if the processing was successful.
5403 static GstRTSPResult
5404 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5405 GstRTSPMessage * request, GstRTSPMessage * response,
5406 GstRTSPStatusCode * code)
5408 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
5409 GstRTSPResult res = GST_RTSP_ERROR;
5412 GstRTSPMethod method = GST_RTSP_INVALID;
5418 /* make sure we don't loop forever */
5422 /* save method so we can disable it when the server complains */
5423 method = request->type_data.request.method;
5426 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
5430 case GST_RTSP_STS_UNAUTHORIZED:
5431 if (gst_rtspsrc_setup_auth (src, response)) {
5432 /* Try the request/response again after configuring the auth info
5440 } while (retry == TRUE);
5442 /* If the user requested the code, let them handle errors, otherwise
5443 * post an error below */
5446 else if (int_code != GST_RTSP_STS_OK)
5447 goto error_response;
5454 GST_DEBUG_OBJECT (src, "got error %d", res);
5459 res = GST_RTSP_ERROR;
5461 switch (response->type_data.response.code) {
5462 case GST_RTSP_STS_NOT_FOUND:
5463 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
5464 response->type_data.response.reason));
5466 case GST_RTSP_STS_MOVED_PERMANENTLY:
5467 case GST_RTSP_STS_MOVE_TEMPORARILY:
5469 gchar *new_location;
5470 GstRTSPLowerTrans transports;
5472 GST_DEBUG_OBJECT (src, "got redirection");
5473 /* if we don't have a Location Header, we must error */
5474 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
5475 &new_location, 0) < 0)
5478 /* When we receive a redirect result, we go back to the INIT state after
5479 * parsing the new URI. The caller should do the needed steps to issue
5480 * a new setup when it detects this state change. */
5481 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
5483 /* save current transports */
5484 if (src->conninfo.url)
5485 transports = src->conninfo.url->transports;
5487 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
5489 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
5491 /* set old transports */
5492 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
5493 src->conninfo.url->transports = transports;
5495 src->need_redirect = TRUE;
5496 src->state = GST_RTSP_STATE_INIT;
5500 case GST_RTSP_STS_NOT_ACCEPTABLE:
5501 case GST_RTSP_STS_NOT_IMPLEMENTED:
5502 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
5503 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
5504 gst_rtsp_method_as_text (method));
5505 src->methods &= ~method;
5509 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5510 ("Got error response: %d (%s).", response->type_data.response.code,
5511 response->type_data.response.reason));
5514 /* if we return ERROR we should unset the response ourselves */
5515 if (res == GST_RTSP_ERROR)
5516 gst_rtsp_message_unset (response);
5522 static GstRTSPResult
5523 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
5524 GstRTSPMessage * response, GstRTSPSrc * src)
5526 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
5531 /* parse the response and collect all the supported methods. We need this
5532 * information so that we don't try to send an unsupported request to the
5536 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
5538 GstRTSPHeaderField field;
5542 /* reset supported methods */
5545 /* Try Allow Header first */
5546 field = GST_RTSP_HDR_ALLOW;
5549 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5550 if (indx == 0 && !respoptions) {
5551 /* if no Allow header was found then try the Public header... */
5552 field = GST_RTSP_HDR_PUBLIC;
5553 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5558 src->methods |= gst_rtsp_options_from_text (respoptions);
5563 if (src->methods == 0) {
5564 /* neither Allow nor Public are required, assume the server supports
5565 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
5567 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
5568 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
5570 /* always assume PLAY, FIXME, extensions should be able to override
5572 src->methods |= GST_RTSP_PLAY;
5573 /* also assume it will support Range */
5574 src->seekable = TRUE;
5576 /* we need describe and setup */
5577 if (!(src->methods & GST_RTSP_DESCRIBE))
5579 if (!(src->methods & GST_RTSP_SETUP))
5587 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5588 ("Server does not support DESCRIBE."));
5593 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5594 ("Server does not support SETUP."));
5599 /* masks to be kept in sync with the hardcoded protocol order of preference
5601 static guint protocol_masks[] = {
5602 GST_RTSP_LOWER_TRANS_UDP,
5603 GST_RTSP_LOWER_TRANS_UDP_MCAST,
5604 GST_RTSP_LOWER_TRANS_TCP,
5608 static GstRTSPResult
5609 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
5610 GstRTSPLowerTrans protocols, GstRTSPProfile profile, gchar ** transports)
5614 gboolean add_udp_str;
5619 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
5624 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
5626 /* extension listed transports, use those */
5627 if (*transports != NULL)
5630 /* it's the default */
5631 add_udp_str = FALSE;
5633 /* the default RTSP transports */
5634 result = g_string_new ("RTP");
5637 case GST_RTSP_PROFILE_AVP:
5638 g_string_append (result, "/AVP");
5640 case GST_RTSP_PROFILE_SAVP:
5641 g_string_append (result, "/SAVP");
5643 case GST_RTSP_PROFILE_AVPF:
5644 g_string_append (result, "/AVPF");
5646 case GST_RTSP_PROFILE_SAVPF:
5647 g_string_append (result, "/SAVPF");
5653 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
5654 GST_DEBUG_OBJECT (src, "adding UDP unicast");
5656 g_string_append (result, "/UDP");
5657 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
5658 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
5659 GST_DEBUG_OBJECT (src, "adding UDP multicast");
5660 /* we don't have to allocate any UDP ports yet, if the selected transport
5661 * turns out to be multicast we can create them and join the multicast
5662 * group indicated in the transport reply */
5664 g_string_append (result, "/UDP");
5665 g_string_append (result, ";multicast");
5666 if (src->next_port_num != 0) {
5667 if (src->client_port_range.max > 0 &&
5668 src->next_port_num >= src->client_port_range.max)
5671 g_string_append_printf (result, ";client_port=%d-%d",
5672 src->next_port_num, src->next_port_num + 1);
5674 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
5675 GST_DEBUG_OBJECT (src, "adding TCP");
5677 g_string_append (result, "/TCP;unicast;interleaved=%%i1-%%i2");
5679 *transports = g_string_free (result, FALSE);
5681 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
5688 GST_ERROR ("extension gave error %d", res);
5693 GST_ERROR ("no more ports available");
5694 return GST_RTSP_ERROR;
5698 static GstRTSPResult
5699 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
5700 gint orig_rtpport, gint orig_rtcpport)
5703 gint nr_udp, nr_int;
5705 gint rtpport = 0, rtcpport = 0;
5708 src = stream->parent;
5710 /* find number of placeholders first */
5711 if (strstr (*transports, "%%i2"))
5713 else if (strstr (*transports, "%%i1"))
5718 if (strstr (*transports, "%%u2"))
5720 else if (strstr (*transports, "%%u1"))
5725 if (nr_udp == 0 && nr_int == 0)
5729 if (!orig_rtpport || !orig_rtcpport) {
5730 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
5733 rtpport = orig_rtpport;
5734 rtcpport = orig_rtcpport;
5738 str = g_string_new ("");
5740 while ((next = strstr (p, "%%"))) {
5741 g_string_append_len (str, p, next - p);
5742 if (next[2] == 'u') {
5744 g_string_append_printf (str, "%d", rtpport);
5745 else if (next[3] == '2')
5746 g_string_append_printf (str, "%d", rtcpport);
5748 if (next[2] == 'i') {
5750 g_string_append_printf (str, "%d", src->free_channel);
5751 else if (next[3] == '2')
5752 g_string_append_printf (str, "%d", src->free_channel + 1);
5757 /* append final part */
5758 g_string_append (str, p);
5760 g_free (*transports);
5761 *transports = g_string_free (str, FALSE);
5769 GST_ERROR ("failed to allocate udp ports");
5770 return GST_RTSP_ERROR;
5774 /* Perform the SETUP request for all the streams.
5776 * We ask the server for a specific transport, which initially includes all the
5777 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
5778 * two local UDP ports that we send to the server.
5780 * Once the server replied with a transport, we configure the other streams
5781 * with the same transport.
5783 * This function will also configure the stream for the selected transport,
5784 * which basically means creating the pipeline.
5786 static GstRTSPResult
5787 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
5790 GstRTSPResult res = GST_RTSP_ERROR;
5791 GstRTSPMessage request = { 0 };
5792 GstRTSPMessage response = { 0 };
5793 GstRTSPStream *stream = NULL;
5794 GstRTSPLowerTrans protocols;
5795 GstRTSPStatusCode code;
5796 gboolean unsupported_real = FALSE;
5797 gint rtpport, rtcpport;
5801 if (src->conninfo.connection) {
5802 url = gst_rtsp_connection_get_url (src->conninfo.connection);
5803 /* we initially allow all configured lower transports. based on the URL
5804 * transports and the replies from the server we narrow them down. */
5805 protocols = url->transports & src->cur_protocols;
5808 protocols = src->cur_protocols;
5814 /* reset some state */
5815 src->free_channel = 0;
5816 src->interleaved = FALSE;
5817 src->need_activate = FALSE;
5818 /* keep track of next port number, 0 is random */
5819 src->next_port_num = src->client_port_range.min;
5820 rtpport = rtcpport = 0;
5822 if (G_UNLIKELY (src->streams == NULL))
5825 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5826 GstRTSPConnection *conn;
5833 stream = (GstRTSPStream *) walk->data;
5835 caps = stream_get_caps_for_pt (stream, stream->default_pt);
5837 GST_DEBUG_OBJECT (src, "skipping stream %p, no caps", stream);
5841 if (stream->skipped) {
5842 GST_DEBUG_OBJECT (src, "skipping stream %p", stream);
5846 /* see if we need to configure this stream */
5847 if (!gst_rtsp_ext_list_configure_stream (src->extensions, caps)) {
5848 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
5853 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_SELECT_STREAM], 0,
5854 stream->id, caps, &selected);
5856 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by signal", stream);
5860 /* merge/overwrite global caps */
5865 s = gst_caps_get_structure (caps, 0);
5867 num = gst_structure_n_fields (src->props);
5868 for (j = 0; j < num; j++) {
5872 name = gst_structure_nth_field_name (src->props, j);
5873 val = gst_structure_get_value (src->props, name);
5874 gst_structure_set_value (s, name, val);
5876 GST_DEBUG_OBJECT (src, "copied %s", name);
5880 /* skip setup if we have no URL for it */
5881 if (stream->conninfo.location == NULL) {
5882 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
5886 if (src->conninfo.connection == NULL) {
5887 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
5888 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
5891 conn = stream->conninfo.connection;
5893 conn = src->conninfo.connection;
5895 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
5896 stream->conninfo.location);
5898 /* if we have a multicast connection, only suggest multicast from now on */
5899 if (stream->is_multicast)
5900 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
5903 /* first selectable protocol */
5904 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5906 if (!protocol_masks[mask])
5910 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
5911 protocol_masks[mask]);
5912 /* create a string with first transport in line */
5914 res = gst_rtspsrc_create_transports_string (src,
5915 protocols & protocol_masks[mask], stream->profile, &transports);
5916 if (res < 0 || transports == NULL)
5917 goto setup_transport_failed;
5919 if (strlen (transports) == 0) {
5920 g_free (transports);
5921 GST_DEBUG_OBJECT (src, "no transports found");
5926 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
5928 /* replace placeholders with real values, this function will optionally
5929 * allocate UDP ports and other info needed to execute the setup request */
5930 res = gst_rtspsrc_prepare_transports (stream, &transports,
5931 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
5933 g_free (transports);
5934 goto setup_transport_failed;
5937 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
5939 /* create SETUP request */
5941 gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
5942 stream->conninfo.location);
5944 g_free (transports);
5945 goto create_request_failed;
5948 /* select transport */
5949 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
5951 /* if the user wants a non default RTP packet size we add the blocksize
5953 if (src->rtp_blocksize > 0) {
5954 hval = g_strdup_printf ("%d", src->rtp_blocksize);
5955 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
5959 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
5962 /* handle the code ourselves */
5963 if ((res = gst_rtspsrc_send (src, conn, &request, &response, &code) < 0))
5967 case GST_RTSP_STS_OK:
5969 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
5970 gst_rtsp_message_unset (&request);
5971 gst_rtsp_message_unset (&response);
5972 /* cleanup of leftover transport */
5973 gst_rtspsrc_stream_free_udp (stream);
5974 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
5975 * we might be in this case */
5976 if (stream->container && rtpport && rtcpport && !retry) {
5977 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
5982 /* this transport did not go down well, but we may have others to try
5983 * that we did not send yet, try those and only give up then
5984 * but not without checking for lost cause/extension so we can
5985 * post a nicer/more useful error message later */
5986 if (!unsupported_real)
5987 unsupported_real = stream->is_real;
5988 /* select next available protocol, give up on this stream if none */
5990 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5992 if (!protocol_masks[mask] || unsupported_real)
5997 /* cleanup of leftover transport and move to the next stream */
5998 gst_rtspsrc_stream_free_udp (stream);
5999 goto response_error;
6002 /* parse response transport */
6004 gchar *resptrans = NULL;
6005 GstRTSPTransport transport = { 0 };
6007 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
6010 gst_rtspsrc_stream_free_udp (stream);
6014 /* parse transport, go to next stream on parse error */
6015 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
6016 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
6020 /* update allowed transports for other streams. once the transport of
6021 * one stream has been determined, we make sure that all other streams
6022 * are configured in the same way */
6023 switch (transport.lower_transport) {
6024 case GST_RTSP_LOWER_TRANS_TCP:
6025 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
6026 protocols = GST_RTSP_LOWER_TRANS_TCP;
6027 src->interleaved = TRUE;
6028 /* update free channels */
6030 MAX (transport.interleaved.min, src->free_channel);
6032 MAX (transport.interleaved.max, src->free_channel);
6033 src->free_channel++;
6035 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
6036 /* only allow multicast for other streams */
6037 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
6038 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
6039 /* if the server selected our ports, increment our counters so that
6040 * we select a new port later */
6041 if (src->next_port_num == transport.port.min &&
6042 src->next_port_num + 1 == transport.port.max) {
6043 src->next_port_num += 2;
6046 case GST_RTSP_LOWER_TRANS_UDP:
6047 /* only allow unicast for other streams */
6048 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
6049 protocols = GST_RTSP_LOWER_TRANS_UDP;
6052 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
6053 transport.lower_transport);
6057 if (!stream->container || (!src->interleaved && !retry)) {
6058 /* now configure the stream with the selected transport */
6059 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
6060 GST_DEBUG_OBJECT (src,
6061 "could not configure stream %p transport, skipping stream",
6064 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
6065 /* retain the first allocated UDP port pair */
6066 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
6067 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
6070 /* we need to activate at least one streams when we detect activity */
6071 src->need_activate = TRUE;
6073 /* stream is setup now */
6074 stream->setup = TRUE;
6079 GstRTSPStream *sskip;
6081 skip = g_list_next (skip);
6085 sskip = (GstRTSPStream *) skip->data;
6087 /* skip all streams with the same control url */
6088 if (g_str_equal (stream->conninfo.location, sskip->conninfo.location)) {
6089 GST_DEBUG_OBJECT (src, "found stream %p with same control %s",
6090 sskip, sskip->conninfo.location);
6091 sskip->skipped = TRUE;
6096 /* clean up our transport struct */
6097 gst_rtsp_transport_init (&transport);
6098 /* clean up used RTSP messages */
6099 gst_rtsp_message_unset (&request);
6100 gst_rtsp_message_unset (&response);
6104 /* store the transport protocol that was configured */
6105 src->cur_protocols = protocols;
6107 gst_rtsp_ext_list_stream_select (src->extensions, url);
6109 /* if there is nothing to activate, error out */
6110 if (!src->need_activate)
6111 goto nothing_to_activate;
6118 /* no transport possible, post an error and stop */
6119 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
6120 ("Could not connect to server, no protocols left"));
6121 return GST_RTSP_ERROR;
6125 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6126 ("SDP contains no streams"));
6127 return GST_RTSP_ERROR;
6129 create_request_failed:
6131 gchar *str = gst_rtsp_strresult (res);
6133 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6134 ("Could not create request. (%s)", str));
6138 setup_transport_failed:
6140 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6141 ("Could not setup transport."));
6142 res = GST_RTSP_ERROR;
6147 const gchar *str = gst_rtsp_status_as_text (code);
6149 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6150 ("Error (%d): %s", code, GST_STR_NULL (str)));
6151 res = GST_RTSP_ERROR;
6156 gchar *str = gst_rtsp_strresult (res);
6158 if (res != GST_RTSP_EINTR) {
6159 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6160 ("Could not send message. (%s)", str));
6162 GST_WARNING_OBJECT (src, "send interrupted");
6169 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6170 ("Server did not select transport."));
6171 res = GST_RTSP_ERROR;
6174 nothing_to_activate:
6176 /* none of the available error codes is really right .. */
6177 if (unsupported_real) {
6178 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6179 (_("No supported stream was found. You might need to install a "
6180 "GStreamer RTSP extension plugin for Real media streams.")),
6183 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6184 (_("No supported stream was found. You might need to allow "
6185 "more transport protocols or may otherwise be missing "
6186 "the right GStreamer RTSP extension plugin.")), (NULL));
6188 return GST_RTSP_ERROR;
6192 gst_rtsp_message_unset (&request);
6193 gst_rtsp_message_unset (&response);
6199 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
6200 GstSegment * segment)
6203 GstRTSPTimeRange *therange;
6206 gst_rtsp_range_free (src->range);
6208 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
6209 GST_DEBUG_OBJECT (src, "parsed range %s", range);
6210 src->range = therange;
6212 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
6214 gst_segment_init (segment, GST_FORMAT_TIME);
6218 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
6219 therange->min.type, therange->min.seconds, therange->max.type,
6220 therange->max.seconds);
6222 if (therange->min.type == GST_RTSP_TIME_NOW)
6224 else if (therange->min.type == GST_RTSP_TIME_END)
6227 seconds = therange->min.seconds * GST_SECOND;
6229 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
6230 GST_TIME_ARGS (seconds));
6232 /* we need to start playback without clipping from the position reported by
6234 segment->start = seconds;
6235 segment->position = seconds;
6237 if (therange->max.type == GST_RTSP_TIME_NOW)
6239 else if (therange->max.type == GST_RTSP_TIME_END)
6242 seconds = therange->max.seconds * GST_SECOND;
6244 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
6245 GST_TIME_ARGS (seconds));
6247 /* live (WMS) server might send overflowed large max as its idea of infinity,
6248 * compensate to prevent problems later on */
6249 if (seconds != -1 && seconds < 0) {
6251 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
6254 /* live (WMS) might send min == max, which is not worth recording */
6255 if (segment->duration == -1 && seconds == segment->start)
6258 /* don't change duration with unknown value, we might have a valid value
6259 * there that we want to keep. */
6261 segment->duration = seconds;
6266 /* Parse clock profived by the server with following syntax:
6268 * "GstNetTimeProvider <wrapped-clock> <server-IP:port> <clock-time>"
6271 gst_rtspsrc_parse_gst_clock (GstRTSPSrc * src, const gchar * gstclock)
6273 gboolean res = FALSE;
6275 if (g_str_has_prefix (gstclock, "GstNetTimeProvider ")) {
6276 gchar **fields = NULL, **parts = NULL;
6277 gchar *remote_ip, *str;
6279 GstClockTime base_time;
6282 fields = g_strsplit (gstclock, " ", 0);
6284 /* wrapped clock, not very interesting for now */
6285 if (fields[1] == NULL)
6288 /* remote IP address and port */
6289 if ((str = fields[2]) == NULL)
6292 parts = g_strsplit (str, ":", 0);
6294 if ((remote_ip = parts[0]) == NULL)
6297 if ((str = parts[1]) == NULL)
6305 if ((str = fields[3]) == NULL)
6308 base_time = g_ascii_strtoull (str, NULL, 10);
6311 gst_net_client_clock_new ((gchar *) "GstRTSPClock", remote_ip, port,
6314 if (src->provided_clock)
6315 gst_object_unref (src->provided_clock);
6316 src->provided_clock = netclock;
6318 gst_element_post_message (GST_ELEMENT_CAST (src),
6319 gst_message_new_clock_provide (GST_OBJECT_CAST (src),
6320 src->provided_clock, TRUE));
6324 g_strfreev (fields);
6330 /* must be called with the RTSP state lock */
6331 static GstRTSPResult
6332 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
6338 /* prepare global stream caps properties */
6340 gst_structure_remove_all_fields (src->props);
6342 src->props = gst_structure_new_empty ("RTSPProperties");
6345 gst_sdp_message_dump (sdp);
6347 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
6349 /* let the app inspect and change the SDP */
6350 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_ON_SDP], 0, sdp);
6352 gst_segment_init (&src->segment, GST_FORMAT_TIME);
6354 /* parse range for duration reporting. */
6359 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
6363 /* keep track of the range and configure it in the segment */
6364 if (gst_rtspsrc_parse_range (src, range, &src->segment))
6368 /* parse clock information. This is GStreamer specific, a server can tell the
6369 * client what clock it is using and wrap that in a network clock. The
6370 * advantage of that is that we can slave to it. */
6372 const gchar *gstclock;
6375 gstclock = gst_sdp_message_get_attribute_val_n (sdp, "x-gst-clock", i);
6376 if (gstclock == NULL)
6379 /* parse the clock and expose it in the provide_clock method */
6380 if (gst_rtspsrc_parse_gst_clock (src, gstclock))
6384 /* try to find a global control attribute. Note that a '*' means that we should
6385 * do aggregate control with the current url (so we don't do anything and
6386 * leave the current connection as is) */
6388 const gchar *control;
6391 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
6392 if (control == NULL)
6395 /* only take fully qualified urls */
6396 if (g_str_has_prefix (control, "rtsp://"))
6400 g_free (src->conninfo.location);
6401 src->conninfo.location = g_strdup (control);
6402 /* make a connection for this, if there was a connection already, nothing
6404 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
6405 GST_ERROR_OBJECT (src, "could not connect");
6408 /* we need to keep the control url separate from the connection url because
6409 * the rules for constructing the media control url need it */
6410 g_free (src->control);
6411 src->control = g_strdup (control);
6414 /* create streams */
6415 n_streams = gst_sdp_message_medias_len (sdp);
6416 for (i = 0; i < n_streams; i++) {
6417 gst_rtspsrc_create_stream (src, sdp, i);
6420 src->state = GST_RTSP_STATE_INIT;
6423 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
6426 /* reset our state */
6427 src->need_range = TRUE;
6430 src->state = GST_RTSP_STATE_READY;
6437 GST_ERROR_OBJECT (src, "setup failed");
6438 gst_rtspsrc_cleanup (src);
6443 static GstRTSPResult
6444 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
6448 GstRTSPMessage request = { 0 };
6449 GstRTSPMessage response = { 0 };
6452 gchar *respcont = NULL;
6455 src->need_redirect = FALSE;
6457 /* can't continue without a valid url */
6458 if (G_UNLIKELY (src->conninfo.url == NULL)) {
6459 res = GST_RTSP_EINVAL;
6462 src->tried_url_auth = FALSE;
6464 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
6465 goto connect_failed;
6467 /* create OPTIONS */
6468 GST_DEBUG_OBJECT (src, "create options...");
6470 gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
6471 src->conninfo.url_str);
6473 goto create_request_failed;
6476 GST_DEBUG_OBJECT (src, "send options...");
6479 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
6482 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6487 if (!gst_rtspsrc_parse_methods (src, &response))
6490 /* create DESCRIBE */
6491 GST_DEBUG_OBJECT (src, "create describe...");
6493 gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
6494 src->conninfo.url_str);
6496 goto create_request_failed;
6498 /* we only accept SDP for now */
6499 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
6503 GST_DEBUG_OBJECT (src, "send describe...");
6506 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
6509 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6513 /* we only perform redirect for the describe, currently */
6514 if (src->need_redirect) {
6515 /* close connection, we don't have to send a TEARDOWN yet, ignore the
6517 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6519 gst_rtsp_message_unset (&request);
6520 gst_rtsp_message_unset (&response);
6526 /* it could be that the DESCRIBE method was not implemented */
6527 if (!src->methods & GST_RTSP_DESCRIBE)
6530 /* check if reply is SDP */
6531 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
6533 /* could not be set but since the request returned OK, we assume it
6534 * was SDP, else check it. */
6536 if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
6537 goto wrong_content_type;
6540 /* get message body and parse as SDP */
6541 gst_rtsp_message_get_body (&response, &data, &size);
6542 if (data == NULL || size == 0)
6545 GST_DEBUG_OBJECT (src, "parse SDP...");
6546 gst_sdp_message_new (sdp);
6547 gst_sdp_message_parse_buffer (data, size, *sdp);
6549 /* clean up any messages */
6550 gst_rtsp_message_unset (&request);
6551 gst_rtsp_message_unset (&response);
6558 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
6559 ("No valid RTSP URL was provided"));
6564 gchar *str = gst_rtsp_strresult (res);
6566 if (res != GST_RTSP_EINTR) {
6567 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
6568 ("Failed to connect. (%s)", str));
6570 GST_WARNING_OBJECT (src, "connect interrupted");
6575 create_request_failed:
6577 gchar *str = gst_rtsp_strresult (res);
6579 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6580 ("Could not create request. (%s)", str));
6586 /* Don't post a message - the rtsp_send method will have
6587 * taken care of it because we passed NULL for the response code */
6592 /* error was posted */
6593 res = GST_RTSP_ERROR;
6598 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6599 ("Server does not support SDP, got %s.", respcont));
6600 res = GST_RTSP_ERROR;
6605 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6606 ("Server can not provide an SDP."));
6607 res = GST_RTSP_ERROR;
6612 if (src->conninfo.connection) {
6613 GST_DEBUG_OBJECT (src, "free connection");
6614 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6616 gst_rtsp_message_unset (&request);
6617 gst_rtsp_message_unset (&response);
6622 static GstRTSPResult
6623 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
6628 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
6630 if (src->sdp == NULL) {
6631 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
6635 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
6640 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
6647 GST_WARNING_OBJECT (src, "can't get sdp");
6648 src->open_error = TRUE;
6653 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
6654 src->open_error = TRUE;
6659 static GstRTSPResult
6660 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
6662 GstRTSPMessage request = { 0 };
6663 GstRTSPMessage response = { 0 };
6664 GstRTSPResult res = GST_RTSP_OK;
6666 const gchar *control;
6668 GST_DEBUG_OBJECT (src, "TEARDOWN...");
6670 gst_rtspsrc_set_state (src, GST_STATE_READY);
6672 if (src->state < GST_RTSP_STATE_READY) {
6673 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
6680 /* construct a control url */
6681 control = get_aggregate_control (src);
6683 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
6686 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6687 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6688 const gchar *setup_url;
6689 GstRTSPConnInfo *info;
6691 /* try aggregate control first but do non-aggregate control otherwise */
6693 setup_url = control;
6694 else if ((setup_url = stream->conninfo.location) == NULL)
6697 if (src->conninfo.connection) {
6698 info = &src->conninfo;
6699 } else if (stream->conninfo.connection) {
6700 info = &stream->conninfo;
6704 if (!info->connected)
6709 gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
6711 goto create_request_failed;
6714 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
6717 gst_rtspsrc_send (src, info->connection, &request, &response,
6721 /* FIXME, parse result? */
6722 gst_rtsp_message_unset (&request);
6723 gst_rtsp_message_unset (&response);
6726 /* early exit when we did aggregate control */
6732 /* close connections */
6733 GST_DEBUG_OBJECT (src, "closing connection...");
6734 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6735 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6736 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6737 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
6741 gst_rtspsrc_cleanup (src);
6743 src->state = GST_RTSP_STATE_INVALID;
6746 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
6751 create_request_failed:
6753 gchar *str = gst_rtsp_strresult (res);
6755 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6756 ("Could not create request. (%s)", str));
6762 gchar *str = gst_rtsp_strresult (res);
6764 gst_rtsp_message_unset (&request);
6765 if (res != GST_RTSP_EINTR) {
6766 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6767 ("Could not send message. (%s)", str));
6769 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
6776 GST_DEBUG_OBJECT (src,
6777 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
6782 /* RTP-Info is of the format:
6784 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
6786 * rtptime corresponds to the timestamp for the NPT time given in the header
6787 * seqbase corresponds to the next sequence number we received. This number
6788 * indicates the first seqnum after the seek and should be used to discard
6789 * packets that are from before the seek.
6792 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
6797 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
6799 infos = g_strsplit (rtpinfo, ",", 0);
6800 for (i = 0; infos[i]; i++) {
6802 GstRTSPStream *stream;
6806 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
6808 /* init values, types of seqbase and timebase are bigger than needed so we
6809 * can store -1 as uninitialized values */
6814 /* parse url, find stream for url.
6815 * parse seq and rtptime. The seq number should be configured in the rtp
6816 * depayloader or session manager to detect gaps. Same for the rtptime, it
6817 * should be used to create an initial time newsegment. */
6818 fields = g_strsplit (infos[i], ";", 0);
6819 for (j = 0; fields[j]; j++) {
6820 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
6821 /* remove leading whitespace */
6822 fields[j] = g_strchug (fields[j]);
6823 if (g_str_has_prefix (fields[j], "url=")) {
6824 /* get the url and the stream */
6826 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
6827 } else if (g_str_has_prefix (fields[j], "seq=")) {
6828 seqbase = atoi (fields[j] + 4);
6829 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
6830 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
6833 g_strfreev (fields);
6834 /* now we need to store the values for the caps of the stream */
6835 if (stream != NULL) {
6836 GST_DEBUG_OBJECT (src,
6837 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
6838 stream, seqbase, timebase);
6840 /* we have a stream, configure detected params */
6841 stream->seqbase = seqbase;
6842 stream->timebase = timebase;
6851 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
6856 interval = strtoul (rtcp, NULL, 10);
6857 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
6862 interval *= GST_MSECOND;
6864 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6865 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6867 /* already (optionally) retrieved this when configuring manager */
6868 if (stream->session) {
6869 GObject *rtpsession = stream->session;
6871 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
6873 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
6877 /* now it happens that (Xenon) server sending this may also provide bogus
6878 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
6879 * and just use RTP-Info to sync */
6881 GObjectClass *klass;
6883 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
6884 if (g_object_class_find_property (klass, "rtcp-sync")) {
6885 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
6886 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
6892 gst_rtspsrc_get_float (const gchar * dstr)
6894 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
6896 /* canonicalise floating point string so we can handle float strings
6897 * in the form "24.930" or "24,930" irrespective of the current locale */
6898 g_strlcpy (s, dstr, sizeof (s));
6899 g_strdelimit (s, ",", '.');
6900 return g_ascii_strtod (s, NULL);
6904 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
6906 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
6908 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
6909 g_strlcpy (val_str, "now", sizeof (val_str));
6911 if (segment->position == 0) {
6912 g_strlcpy (val_str, "0", sizeof (val_str));
6914 g_ascii_dtostr (val_str, sizeof (val_str),
6915 ((gdouble) segment->position) / GST_SECOND);
6918 return g_strdup_printf ("npt=%s-", val_str);
6922 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
6926 stream->timebase = -1;
6927 stream->seqbase = -1;
6929 len = stream->ptmap->len;
6930 for (i = 0; i < len; i++) {
6931 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
6934 if (item->caps == NULL)
6937 item->caps = gst_caps_make_writable (item->caps);
6938 s = gst_caps_get_structure (item->caps, 0);
6939 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
6943 static GstRTSPResult
6944 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
6946 GstRTSPResult res = GST_RTSP_OK;
6948 if (src->state < GST_RTSP_STATE_READY) {
6949 res = GST_RTSP_ERROR;
6950 if (src->open_error) {
6951 GST_DEBUG_OBJECT (src, "the stream was in error");
6955 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
6957 if ((res = gst_rtspsrc_open (src, async)) < 0) {
6958 GST_DEBUG_OBJECT (src, "failed to open stream");
6967 static GstRTSPResult
6968 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
6970 GstRTSPMessage request = { 0 };
6971 GstRTSPMessage response = { 0 };
6972 GstRTSPResult res = GST_RTSP_OK;
6976 const gchar *control;
6978 GST_DEBUG_OBJECT (src, "PLAY...");
6980 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6983 if (!(src->methods & GST_RTSP_PLAY))
6986 if (src->state == GST_RTSP_STATE_PLAYING)
6989 if (!src->conninfo.connection || !src->conninfo.connected)
6992 /* send some dummy packets before we activate the receive in the
6994 gst_rtspsrc_send_dummy_packets (src);
6996 /* require new SR packets */
6998 g_signal_emit_by_name (src->manager, "reset-sync", NULL);
7000 gst_rtspsrc_set_state (src, GST_STATE_PLAYING);
7002 /* construct a control url */
7003 control = get_aggregate_control (src);
7005 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7006 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7007 const gchar *setup_url;
7008 GstRTSPConnection *conn;
7010 /* try aggregate control first but do non-aggregate control otherwise */
7012 setup_url = control;
7013 else if ((setup_url = stream->conninfo.location) == NULL)
7016 if (src->conninfo.connection) {
7017 conn = src->conninfo.connection;
7018 } else if (stream->conninfo.connection) {
7019 conn = stream->conninfo.connection;
7025 res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
7027 goto create_request_failed;
7029 if (src->need_range) {
7030 hval = gen_range_header (src, segment);
7032 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
7034 /* store the newsegment event so it can be sent from the streaming thread. */
7035 if (src->start_segment)
7036 gst_event_unref (src->start_segment);
7037 src->start_segment = gst_event_new_segment (&src->segment);
7040 if (segment->rate != 1.0) {
7041 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
7043 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
7045 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
7047 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
7051 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
7053 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7056 /* seek may have silently failed as it is not supported */
7057 if (!(src->methods & GST_RTSP_PLAY)) {
7058 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
7059 /* obviously it is supported as we made it here */
7060 src->methods |= GST_RTSP_PLAY;
7061 src->seekable = FALSE;
7062 /* but there is nothing to parse in the response,
7063 * so convey we have no idea and not to expect anything particular */
7064 clear_rtp_base (src, stream);
7068 /* need to do for all streams */
7069 for (run = src->streams; run; run = g_list_next (run))
7070 clear_rtp_base (src, (GstRTSPStream *) run->data);
7072 /* NOTE the above also disables npt based eos detection */
7073 /* and below forces position to 0,
7074 * which is visible feedback we lost the plot */
7075 segment->start = segment->position = src->last_pos;
7078 gst_rtsp_message_unset (&request);
7080 /* parse RTP npt field. This is the current position in the stream (Normal
7081 * Play Time) and should be put in the NEWSEGMENT position field. */
7082 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
7084 gst_rtspsrc_parse_range (src, hval, segment);
7086 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
7087 segment->rate = 1.0;
7089 /* parse Speed header. This is the intended playback rate of the stream
7090 * and should be put in the NEWSEGMENT rate field. */
7091 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
7092 0) == GST_RTSP_OK) {
7093 segment->rate = gst_rtspsrc_get_float (hval);
7094 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
7095 &hval, 0) == GST_RTSP_OK) {
7096 segment->rate = gst_rtspsrc_get_float (hval);
7099 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
7100 * for the RTP packets. If this is not present, we assume all starts from 0...
7101 * This is info for the RTP session manager that we pass to it in caps. */
7103 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
7104 &hval, hval_idx++) == GST_RTSP_OK)
7105 gst_rtspsrc_parse_rtpinfo (src, hval);
7107 /* some servers indicate RTCP parameters in PLAY response,
7108 * rather than properly in SDP */
7109 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
7110 &hval, 0) == GST_RTSP_OK)
7111 gst_rtspsrc_handle_rtcp_interval (src, hval);
7113 gst_rtsp_message_unset (&response);
7115 /* early exit when we did aggregate control */
7119 /* configure the caps of the streams after we parsed all headers. Only reset
7120 * the manager object when we set a new Range header (we did a seek) */
7121 gst_rtspsrc_configure_caps (src, segment, src->need_range);
7123 /* set again when needed */
7124 src->need_range = FALSE;
7126 src->running = TRUE;
7127 src->base_time = -1;
7128 src->state = GST_RTSP_STATE_PLAYING;
7131 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
7132 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7133 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7134 stream->discont = TRUE;
7139 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
7146 GST_DEBUG_OBJECT (src, "failed to open stream");
7151 GST_DEBUG_OBJECT (src, "PLAY is not supported");
7156 GST_DEBUG_OBJECT (src, "we were already PLAYING");
7159 create_request_failed:
7161 gchar *str = gst_rtsp_strresult (res);
7163 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7164 ("Could not create request. (%s)", str));
7170 gchar *str = gst_rtsp_strresult (res);
7172 gst_rtsp_message_unset (&request);
7173 if (res != GST_RTSP_EINTR) {
7174 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7175 ("Could not send message. (%s)", str));
7177 GST_WARNING_OBJECT (src, "PLAY interrupted");
7184 static GstRTSPResult
7185 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
7187 GstRTSPResult res = GST_RTSP_OK;
7188 GstRTSPMessage request = { 0 };
7189 GstRTSPMessage response = { 0 };
7191 const gchar *control;
7193 GST_DEBUG_OBJECT (src, "PAUSE...");
7195 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
7198 if (!(src->methods & GST_RTSP_PAUSE))
7201 if (src->state == GST_RTSP_STATE_READY)
7204 if (!src->conninfo.connection || !src->conninfo.connected)
7207 /* construct a control url */
7208 control = get_aggregate_control (src);
7210 /* loop over the streams. We might exit the loop early when we could do an
7211 * aggregate control */
7212 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7213 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7214 GstRTSPConnection *conn;
7215 const gchar *setup_url;
7217 /* try aggregate control first but do non-aggregate control otherwise */
7219 setup_url = control;
7220 else if ((setup_url = stream->conninfo.location) == NULL)
7223 if (src->conninfo.connection) {
7224 conn = src->conninfo.connection;
7225 } else if (stream->conninfo.connection) {
7226 conn = stream->conninfo.connection;
7232 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
7233 ("Sending PAUSE request"));
7236 gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
7238 goto create_request_failed;
7240 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7243 gst_rtsp_message_unset (&request);
7244 gst_rtsp_message_unset (&response);
7246 /* exit early when we did agregate control */
7251 /* change element states now */
7252 gst_rtspsrc_set_state (src, GST_STATE_PAUSED);
7255 src->state = GST_RTSP_STATE_READY;
7259 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
7266 GST_DEBUG_OBJECT (src, "failed to open stream");
7271 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
7276 GST_DEBUG_OBJECT (src, "we were already PAUSED");
7279 create_request_failed:
7281 gchar *str = gst_rtsp_strresult (res);
7283 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7284 ("Could not create request. (%s)", str));
7290 gchar *str = gst_rtsp_strresult (res);
7292 gst_rtsp_message_unset (&request);
7293 if (res != GST_RTSP_EINTR) {
7294 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7295 ("Could not send message. (%s)", str));
7297 GST_WARNING_OBJECT (src, "PAUSE interrupted");
7305 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
7307 GstRTSPSrc *rtspsrc;
7309 rtspsrc = GST_RTSPSRC (bin);
7311 switch (GST_MESSAGE_TYPE (message)) {
7312 case GST_MESSAGE_EOS:
7313 gst_message_unref (message);
7315 case GST_MESSAGE_ELEMENT:
7317 const GstStructure *s = gst_message_get_structure (message);
7319 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
7320 gboolean ignore_timeout;
7322 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
7324 GST_OBJECT_LOCK (rtspsrc);
7325 ignore_timeout = rtspsrc->ignore_timeout;
7326 rtspsrc->ignore_timeout = TRUE;
7327 GST_OBJECT_UNLOCK (rtspsrc);
7329 /* we only act on the first udp timeout message, others are irrelevant
7330 * and can be ignored. */
7331 if (!ignore_timeout)
7332 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
7334 gst_message_unref (message);
7337 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7340 case GST_MESSAGE_ERROR:
7343 GstRTSPStream *stream;
7346 udpsrc = GST_MESSAGE_SRC (message);
7348 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
7349 GST_ELEMENT_NAME (udpsrc));
7351 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
7355 /* we ignore the RTCP udpsrc */
7356 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
7359 /* if we get error messages from the udp sources, that's not a problem as
7360 * long as not all of them error out. We also don't really know what the
7361 * problem is, the message does not give enough detail... */
7362 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
7363 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
7364 if (ret != GST_FLOW_OK)
7368 gst_message_unref (message);
7372 /* fatal but not our message, forward */
7373 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7378 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7384 /* the thread where everything happens */
7386 gst_rtspsrc_thread (GstRTSPSrc * src)
7390 GST_OBJECT_LOCK (src);
7391 cmd = src->pending_cmd;
7392 if (cmd == CMD_RECONNECT || cmd == CMD_PLAY || cmd == CMD_PAUSE
7393 || cmd == CMD_LOOP || cmd == CMD_OPEN)
7394 src->pending_cmd = CMD_LOOP;
7396 src->pending_cmd = CMD_WAIT;
7397 GST_DEBUG_OBJECT (src, "got command %d", cmd);
7399 /* we got the message command, so ensure communication is possible again */
7400 gst_rtspsrc_connection_flush (src, FALSE);
7402 src->busy_cmd = cmd;
7403 GST_OBJECT_UNLOCK (src);
7407 gst_rtspsrc_open (src, TRUE);
7410 gst_rtspsrc_play (src, &src->segment, TRUE);
7413 gst_rtspsrc_pause (src, TRUE);
7416 gst_rtspsrc_close (src, TRUE, FALSE);
7419 gst_rtspsrc_loop (src);
7422 gst_rtspsrc_reconnect (src, FALSE);
7428 GST_OBJECT_LOCK (src);
7429 /* and go back to sleep */
7430 if (src->pending_cmd == CMD_WAIT) {
7432 gst_task_pause (src->task);
7435 src->busy_cmd = CMD_WAIT;
7436 GST_OBJECT_UNLOCK (src);
7440 gst_rtspsrc_start (GstRTSPSrc * src)
7442 GST_DEBUG_OBJECT (src, "starting");
7444 GST_OBJECT_LOCK (src);
7446 src->pending_cmd = CMD_WAIT;
7448 if (src->task == NULL) {
7449 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src, NULL);
7450 if (src->task == NULL)
7453 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
7455 GST_OBJECT_UNLOCK (src);
7462 GST_OBJECT_UNLOCK (src);
7463 GST_ERROR_OBJECT (src, "failed to create task");
7469 gst_rtspsrc_stop (GstRTSPSrc * src)
7473 GST_DEBUG_OBJECT (src, "stopping");
7475 /* also cancels pending task */
7476 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_ALL);
7478 GST_OBJECT_LOCK (src);
7479 if ((task = src->task)) {
7481 GST_OBJECT_UNLOCK (src);
7483 gst_task_stop (task);
7485 /* make sure it is not running */
7486 GST_RTSP_STREAM_LOCK (src);
7487 GST_RTSP_STREAM_UNLOCK (src);
7489 /* now wait for the task to finish */
7490 gst_task_join (task);
7492 /* and free the task */
7493 gst_object_unref (GST_OBJECT (task));
7495 GST_OBJECT_LOCK (src);
7497 GST_OBJECT_UNLOCK (src);
7499 /* ensure synchronously all is closed and clean */
7500 gst_rtspsrc_close (src, FALSE, TRUE);
7505 static GstStateChangeReturn
7506 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
7508 GstRTSPSrc *rtspsrc;
7509 GstStateChangeReturn ret;
7511 rtspsrc = GST_RTSPSRC (element);
7513 switch (transition) {
7514 case GST_STATE_CHANGE_NULL_TO_READY:
7515 if (!gst_rtspsrc_start (rtspsrc))
7518 case GST_STATE_CHANGE_READY_TO_PAUSED:
7519 /* init some state */
7520 rtspsrc->cur_protocols = rtspsrc->protocols;
7521 /* first attempt, don't ignore timeouts */
7522 rtspsrc->ignore_timeout = FALSE;
7523 rtspsrc->open_error = FALSE;
7524 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
7526 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
7527 set_manager_buffer_mode (rtspsrc);
7529 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
7530 /* unblock the tcp tasks and make the loop waiting */
7531 if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
7532 /* make sure it is waiting before we send PAUSE or PLAY below */
7533 GST_RTSP_STREAM_LOCK (rtspsrc);
7534 GST_RTSP_STREAM_UNLOCK (rtspsrc);
7537 case GST_STATE_CHANGE_PAUSED_TO_READY:
7543 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
7544 if (ret == GST_STATE_CHANGE_FAILURE)
7547 switch (transition) {
7548 case GST_STATE_CHANGE_NULL_TO_READY:
7549 ret = GST_STATE_CHANGE_SUCCESS;
7551 case GST_STATE_CHANGE_READY_TO_PAUSED:
7552 ret = GST_STATE_CHANGE_NO_PREROLL;
7554 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
7555 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
7556 ret = GST_STATE_CHANGE_SUCCESS;
7558 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
7559 /* send pause request and keep the idle task around */
7560 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
7561 ret = GST_STATE_CHANGE_NO_PREROLL;
7563 case GST_STATE_CHANGE_PAUSED_TO_READY:
7564 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
7565 ret = GST_STATE_CHANGE_SUCCESS;
7567 case GST_STATE_CHANGE_READY_TO_NULL:
7568 gst_rtspsrc_stop (rtspsrc);
7569 ret = GST_STATE_CHANGE_SUCCESS;
7580 GST_DEBUG_OBJECT (rtspsrc, "start failed");
7581 return GST_STATE_CHANGE_FAILURE;
7586 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
7589 GstRTSPSrc *rtspsrc;
7591 rtspsrc = GST_RTSPSRC (element);
7593 if (GST_EVENT_IS_DOWNSTREAM (event)) {
7594 res = gst_rtspsrc_push_event (rtspsrc, event);
7596 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
7603 /*** GSTURIHANDLER INTERFACE *************************************************/
7606 gst_rtspsrc_uri_get_type (GType type)
7611 static const gchar *const *
7612 gst_rtspsrc_uri_get_protocols (GType type)
7614 static const gchar *protocols[] =
7615 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
7616 "rtsps", "rtspsu", "rtspst", "rtspsh", NULL
7623 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
7625 GstRTSPSrc *src = GST_RTSPSRC (handler);
7627 /* FIXME: make thread-safe */
7628 return g_strdup (src->conninfo.location);
7632 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
7637 GstRTSPUrl *newurl = NULL;
7638 GstSDPMessage *sdp = NULL;
7640 src = GST_RTSPSRC (handler);
7642 /* same URI, we're fine */
7643 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
7646 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
7647 if ((res = gst_sdp_message_new (&sdp) < 0))
7650 GST_DEBUG_OBJECT (src, "parsing SDP message");
7651 if ((res = gst_sdp_message_parse_uri (uri, sdp) < 0))
7655 GST_DEBUG_OBJECT (src, "parsing URI");
7656 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
7660 /* if worked, free previous and store new url object along with the original
7662 GST_DEBUG_OBJECT (src, "configuring URI");
7663 g_free (src->conninfo.location);
7664 src->conninfo.location = g_strdup (uri);
7665 gst_rtsp_url_free (src->conninfo.url);
7666 src->conninfo.url = newurl;
7667 g_free (src->conninfo.url_str);
7669 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
7671 src->conninfo.url_str = NULL;
7674 gst_sdp_message_free (src->sdp);
7676 src->from_sdp = sdp != NULL;
7678 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
7679 GST_DEBUG_OBJECT (src, "request uri is: %s",
7680 GST_STR_NULL (src->conninfo.url_str));
7687 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
7692 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", res);
7693 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7694 "Could not create SDP");
7699 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", res,
7700 GST_STR_NULL (uri));
7701 gst_sdp_message_free (sdp);
7702 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7708 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
7709 GST_STR_NULL (uri), res);
7710 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7711 "Invalid RTSP URI");
7717 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
7719 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
7721 iface->get_type = gst_rtspsrc_uri_get_type;
7722 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
7723 iface->get_uri = gst_rtspsrc_uri_get_uri;
7724 iface->set_uri = gst_rtspsrc_uri_set_uri;