2 * Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
3 * <2006> Lutz Mueller <lutz at topfrose dot de>
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
18 * Boston, MA 02110-1301, USA.
21 * Unless otherwise indicated, Source Code is licensed under MIT license.
22 * See further explanation attached in License Statement (distributed in the file
25 * Permission is hereby granted, free of charge, to any person obtaining a copy of
26 * this software and associated documentation files (the "Software"), to deal in
27 * the Software without restriction, including without limitation the rights to
28 * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
29 * of the Software, and to permit persons to whom the Software is furnished to do
30 * so, subject to the following conditions:
32 * The above copyright notice and this permission notice shall be included in all
33 * copies or substantial portions of the Software.
35 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
36 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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39 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
40 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
44 * SECTION:element-rtspsrc
46 * Makes a connection to an RTSP server and read the data.
47 * rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
48 * RealMedia/Quicktime/Microsoft extensions.
50 * RTSP supports transport over TCP or UDP in unicast or multicast mode. By
51 * default rtspsrc will negotiate a connection in the following order:
52 * UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
53 * protocols can be controlled with the #GstRTSPSrc:protocols property.
55 * rtspsrc currently understands SDP as the format of the session description.
56 * For each stream listed in the SDP a new rtp_stream%d pad will be created
57 * with caps derived from the SDP media description. This is a caps of mime type
58 * "application/x-rtp" that can be connected to any available RTP depayloader
61 * rtspsrc will internally instantiate an RTP session manager element
62 * that will handle the RTCP messages to and from the server, jitter removal,
63 * packet reordering along with providing a clock for the pipeline.
64 * This feature is implemented using the gstrtpbin element.
66 * rtspsrc acts like a live source and will therefore only generate data in the
70 * <title>Example launch line</title>
72 * gst-launch-1.0 rtspsrc location=rtsp://some.server/url ! fakesink
73 * ]| Establish a connection to an RTSP server and send the raw RTP packets to a
77 * Last reviewed on 2006-08-18 (0.10.5)
86 #endif /* HAVE_UNISTD_H */
92 #include <gst/net/gstnet.h>
93 #include <gst/sdp/gstsdpmessage.h>
94 #include <gst/sdp/gstmikey.h>
95 #include <gst/rtp/gstrtppayloads.h>
97 #include "gst/gst-i18n-plugin.h"
99 #include "gstrtspsrc.h"
101 GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
102 #define GST_CAT_DEFAULT (rtspsrc_debug)
104 static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
107 GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
109 /* templates used internally */
110 static GstStaticPadTemplate anysrctemplate =
111 GST_STATIC_PAD_TEMPLATE ("internalsrc_%u",
114 GST_STATIC_CAPS_ANY);
116 static GstStaticPadTemplate anysinktemplate =
117 GST_STATIC_PAD_TEMPLATE ("internalsink_%u",
120 GST_STATIC_CAPS_ANY);
124 SIGNAL_HANDLE_REQUEST,
126 SIGNAL_SELECT_STREAM,
131 enum _GstRtspSrcRtcpSyncMode
138 enum _GstRtspSrcBufferMode
147 #define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
149 gst_rtsp_src_buffer_mode_get_type (void)
151 static GType buffer_mode_type = 0;
152 static const GEnumValue buffer_modes[] = {
153 {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
154 {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
155 {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
156 {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
157 {BUFFER_MODE_SYNCED, "Synchronized sender and receiver clocks", "synced"},
161 if (!buffer_mode_type) {
163 g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
165 return buffer_mode_type;
168 #define DEFAULT_LOCATION NULL
169 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
170 #define DEFAULT_DEBUG FALSE
171 #define DEFAULT_RETRY 20
172 #define DEFAULT_TIMEOUT 5000000
173 #define DEFAULT_UDP_BUFFER_SIZE 0x80000
174 #define DEFAULT_TCP_TIMEOUT 20000000
175 #define DEFAULT_LATENCY_MS 2000
176 #define DEFAULT_DROP_ON_LATENCY FALSE
177 #define DEFAULT_CONNECTION_SPEED 0
178 #define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
179 #define DEFAULT_DO_RTCP TRUE
180 #define DEFAULT_DO_RTSP_KEEP_ALIVE TRUE
181 #define DEFAULT_PROXY NULL
182 #define DEFAULT_RTP_BLOCKSIZE 0
183 #define DEFAULT_USER_ID NULL
184 #define DEFAULT_USER_PW NULL
185 #define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO
186 #define DEFAULT_PORT_RANGE NULL
187 #define DEFAULT_SHORT_HEADER FALSE
188 #define DEFAULT_PROBATION 2
189 #define DEFAULT_UDP_RECONNECT TRUE
190 #define DEFAULT_MULTICAST_IFACE NULL
191 #define DEFAULT_NTP_SYNC FALSE
192 #define DEFAULT_USE_PIPELINE_CLOCK FALSE
193 #define DEFAULT_TLS_VALIDATION_FLAGS G_TLS_CERTIFICATE_VALIDATE_ALL
194 #define DEFAULT_TLS_DATABASE NULL
206 PROP_DROP_ON_LATENCY,
207 PROP_CONNECTION_SPEED,
210 PROP_DO_RTSP_KEEP_ALIVE,
219 PROP_UDP_BUFFER_SIZE,
223 PROP_MULTICAST_IFACE,
225 PROP_USE_PIPELINE_CLOCK,
227 PROP_TLS_VALIDATION_FLAGS,
232 #define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
234 gst_rtsp_nat_method_get_type (void)
236 static GType rtsp_nat_method_type = 0;
237 static const GEnumValue rtsp_nat_method[] = {
238 {GST_RTSP_NAT_NONE, "None", "none"},
239 {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
243 if (!rtsp_nat_method_type) {
244 rtsp_nat_method_type =
245 g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
247 return rtsp_nat_method_type;
250 static void gst_rtspsrc_finalize (GObject * object);
252 static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
253 const GValue * value, GParamSpec * pspec);
254 static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
255 GValue * value, GParamSpec * pspec);
257 static GstClock *gst_rtspsrc_provide_clock (GstElement * element);
259 static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
260 gpointer iface_data);
262 static void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
265 static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
266 static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
268 static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
270 static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
271 GstStateChange transition);
272 static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
273 static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
275 static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
276 GstRTSPMessage * response);
278 static gboolean gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd,
280 static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
281 GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
283 static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
284 static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
286 static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
287 static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
288 gboolean only_close);
290 static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
291 const gchar * uri, GError ** error);
292 static gchar *gst_rtspsrc_uri_get_uri (GstURIHandler * handler);
294 static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
295 static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
296 static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
297 GstRTSPStream * stream, GstEvent * event);
298 static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
299 static void gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush);
307 /* commands we send to out loop to notify it of events */
308 #define CMD_OPEN (1 << 0)
309 #define CMD_PLAY (1 << 1)
310 #define CMD_PAUSE (1 << 2)
311 #define CMD_CLOSE (1 << 3)
312 #define CMD_WAIT (1 << 4)
313 #define CMD_RECONNECT (1 << 5)
314 #define CMD_LOOP (1 << 6)
316 /* mask for all commands */
317 #define CMD_ALL ((CMD_LOOP << 1) - 1)
319 #define GST_ELEMENT_PROGRESS(el, type, code, text) \
321 gchar *__txt = _gst_element_error_printf text; \
322 gst_element_post_message (GST_ELEMENT_CAST (el), \
323 gst_message_new_progress (GST_OBJECT_CAST (el), \
324 GST_PROGRESS_TYPE_ ##type, code, __txt)); \
328 static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 };
330 #define gst_rtspsrc_parent_class parent_class
331 G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
332 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
335 default_select_stream (GstRTSPSrc * src, guint id, GstCaps * caps)
337 GST_DEBUG_OBJECT (src, "default handler");
342 select_stream_accum (GSignalInvocationHint * ihint,
343 GValue * return_accu, const GValue * handler_return, gpointer data)
347 myboolean = g_value_get_boolean (handler_return);
348 GST_DEBUG ("accum %d", myboolean);
349 g_value_set_boolean (return_accu, myboolean);
351 /* stop emission if FALSE */
356 gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
358 GObjectClass *gobject_class;
359 GstElementClass *gstelement_class;
360 GstBinClass *gstbin_class;
362 gobject_class = (GObjectClass *) klass;
363 gstelement_class = (GstElementClass *) klass;
364 gstbin_class = (GstBinClass *) klass;
366 GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
368 gobject_class->set_property = gst_rtspsrc_set_property;
369 gobject_class->get_property = gst_rtspsrc_get_property;
371 gobject_class->finalize = gst_rtspsrc_finalize;
373 g_object_class_install_property (gobject_class, PROP_LOCATION,
374 g_param_spec_string ("location", "RTSP Location",
375 "Location of the RTSP url to read",
376 DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
378 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
379 g_param_spec_flags ("protocols", "Protocols",
380 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
381 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
383 g_object_class_install_property (gobject_class, PROP_DEBUG,
384 g_param_spec_boolean ("debug", "Debug",
385 "Dump request and response messages to stdout",
386 DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
388 g_object_class_install_property (gobject_class, PROP_RETRY,
389 g_param_spec_uint ("retry", "Retry",
390 "Max number of retries when allocating RTP ports.",
391 0, G_MAXUINT16, DEFAULT_RETRY,
392 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
394 g_object_class_install_property (gobject_class, PROP_TIMEOUT,
395 g_param_spec_uint64 ("timeout", "Timeout",
396 "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
397 0, G_MAXUINT64, DEFAULT_TIMEOUT,
398 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
400 g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
401 g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
402 "Fail after timeout microseconds on TCP connections (0 = disabled)",
403 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
404 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
406 g_object_class_install_property (gobject_class, PROP_LATENCY,
407 g_param_spec_uint ("latency", "Buffer latency in ms",
408 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
409 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
411 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
412 g_param_spec_boolean ("drop-on-latency",
413 "Drop buffers when maximum latency is reached",
414 "Tells the jitterbuffer to never exceed the given latency in size",
415 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
417 g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
418 g_param_spec_uint64 ("connection-speed", "Connection Speed",
419 "Network connection speed in kbps (0 = unknown)",
420 0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
421 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
423 g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
424 g_param_spec_enum ("nat-method", "NAT Method",
425 "Method to use for traversing firewalls and NAT",
426 GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
427 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
430 * GstRTSPSrc:do-rtcp:
432 * Enable RTCP support. Some old server don't like RTCP and then this property
433 * needs to be set to FALSE.
435 g_object_class_install_property (gobject_class, PROP_DO_RTCP,
436 g_param_spec_boolean ("do-rtcp", "Do RTCP",
437 "Send RTCP packets, disable for old incompatible server.",
438 DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
441 * GstRTSPSrc:do-rtsp-keep-alive:
443 * Enable RTSP keep alive support. Some old server don't like RTSP
444 * keep alive and then this property needs to be set to FALSE.
446 g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
447 g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
448 "Send RTSP keep alive packets, disable for old incompatible server.",
449 DEFAULT_DO_RTSP_KEEP_ALIVE,
450 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
455 * Set the proxy parameters. This has to be a string of the format
456 * [http://][user:passwd@]host[:port].
458 g_object_class_install_property (gobject_class, PROP_PROXY,
459 g_param_spec_string ("proxy", "Proxy",
460 "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
461 DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
463 * GstRTSPSrc:proxy-id:
465 * Sets the proxy URI user id for authentication. If the URI set via the
466 * "proxy" property contains a user-id already, that will take precedence.
470 g_object_class_install_property (gobject_class, PROP_PROXY_ID,
471 g_param_spec_string ("proxy-id", "proxy-id",
472 "HTTP proxy URI user id for authentication", "",
473 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
475 * GstRTSPSrc:proxy-pw:
477 * Sets the proxy URI password for authentication. If the URI set via the
478 * "proxy" property contains a password already, that will take precedence.
482 g_object_class_install_property (gobject_class, PROP_PROXY_PW,
483 g_param_spec_string ("proxy-pw", "proxy-pw",
484 "HTTP proxy URI user password for authentication", "",
485 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
488 * GstRTSPSrc:rtp-blocksize:
490 * RTP package size to suggest to server.
492 g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
493 g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
494 "RTP package size to suggest to server (0 = disabled)",
495 0, 65536, DEFAULT_RTP_BLOCKSIZE,
496 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
498 g_object_class_install_property (gobject_class,
500 g_param_spec_string ("user-id", "user-id",
501 "RTSP location URI user id for authentication", DEFAULT_USER_ID,
502 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
503 g_object_class_install_property (gobject_class, PROP_USER_PW,
504 g_param_spec_string ("user-pw", "user-pw",
505 "RTSP location URI user password for authentication", DEFAULT_USER_PW,
506 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
509 * GstRTSPSrc:buffer-mode:
511 * Control the buffering and timestamping mode used by the jitterbuffer.
513 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
514 g_param_spec_enum ("buffer-mode", "Buffer Mode",
515 "Control the buffering algorithm in use",
516 GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
517 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
520 * GstRTSPSrc:port-range:
522 * Configure the client port numbers that can be used to recieve RTP and
525 g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
526 g_param_spec_string ("port-range", "Port range",
527 "Client port range that can be used to receive RTP and RTCP data, "
528 "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
529 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
532 * GstRTSPSrc:udp-buffer-size:
534 * Size of the kernel UDP receive buffer in bytes.
536 g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
537 g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
538 "Size of the kernel UDP receive buffer in bytes, 0=default",
539 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
540 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
543 * GstRTSPSrc:short-header:
545 * Only send the basic RTSP headers for broken encoders.
547 g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
548 g_param_spec_boolean ("short-header", "Short Header",
549 "Only send the basic RTSP headers for broken encoders",
550 DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
552 g_object_class_install_property (gobject_class, PROP_PROBATION,
553 g_param_spec_uint ("probation", "Number of probations",
554 "Consecutive packet sequence numbers to accept the source",
555 0, G_MAXUINT, DEFAULT_PROBATION,
556 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
558 g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
559 g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
560 "Reconnect to the server if RTSP connection is closed when doing UDP",
561 DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
563 g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
564 g_param_spec_string ("multicast-iface", "Multicast Interface",
565 "The network interface on which to join the multicast group",
566 DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
568 g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
569 g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
570 "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
571 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
573 g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
574 g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
575 "Use the pipeline running-time to set the NTP time in the RTCP SR messages",
576 DEFAULT_USE_PIPELINE_CLOCK,
577 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
579 g_object_class_install_property (gobject_class, PROP_SDES,
580 g_param_spec_boxed ("sdes", "SDES",
581 "The SDES items of this session",
582 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
585 * GstRTSPSrc::tls-validation-flags:
587 * TLS certificate validation flags used to validate server
592 g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
593 g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
594 "TLS certificate validation flags used to validate the server certificate",
595 G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS,
596 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
599 * GstRTSPSrc::tls-database:
601 * TLS database with anchor certificate authorities used to validate
602 * the server certificate.
606 g_object_class_install_property (gobject_class, PROP_TLS_DATABASE,
607 g_param_spec_object ("tls-database", "TLS database",
608 "TLS database with anchor certificate authorities used to validate the server certificate",
609 G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
612 * GstRTSPSrc::handle-request:
613 * @rtspsrc: a #GstRTSPSrc
614 * @request: a #GstRTSPMessage
615 * @response: a #GstRTSPMessage
617 * Handle a server request in @request and prepare @response.
619 * This signal is called from the streaming thread, you should therefore not
620 * do any state changes on @rtspsrc because this might deadlock. If you want
621 * to modify the state as a result of this signal, post a
622 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
627 gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST] =
628 g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
629 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2,
630 G_TYPE_POINTER, G_TYPE_POINTER);
633 * GstRTSPSrc::on-sdp:
634 * @rtspsrc: a #GstRTSPSrc
635 * @sdp: a #GstSDPMessage
637 * Emited when the client has retrieved the SDP and before it configures the
638 * streams in the SDP. @sdp can be inspected and modified.
640 * This signal is called from the streaming thread, you should therefore not
641 * do any state changes on @rtspsrc because this might deadlock. If you want
642 * to modify the state as a result of this signal, post a
643 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
648 gst_rtspsrc_signals[SIGNAL_ON_SDP] =
649 g_signal_new ("on-sdp", G_TYPE_FROM_CLASS (klass), 0,
650 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
651 GST_TYPE_SDP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
654 * GstRTSPSrc::select-stream:
655 * @rtspsrc: a #GstRTSPSrc
656 * @num: the stream number
657 * @caps: the stream caps
659 * Emited before the client decides to configure the stream @num with
662 * Returns: %TRUE when the stream should be selected, %FALSE when the stream
667 gst_rtspsrc_signals[SIGNAL_SELECT_STREAM] =
668 g_signal_new_class_handler ("select-stream", G_TYPE_FROM_CLASS (klass),
669 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP,
670 (GCallback) default_select_stream, select_stream_accum, NULL,
671 g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2, G_TYPE_UINT,
674 * GstRTSPSrc::new-manager:
675 * @rtspsrc: a #GstRTSPSrc
676 * @manager: a #GstElement
678 * Emited after a new manager (like rtpbin) was created and the default
679 * properties were configured.
683 gst_rtspsrc_signals[SIGNAL_NEW_MANAGER] =
684 g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass),
685 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
686 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
688 gstelement_class->send_event = gst_rtspsrc_send_event;
689 gstelement_class->provide_clock = gst_rtspsrc_provide_clock;
690 gstelement_class->change_state = gst_rtspsrc_change_state;
692 gst_element_class_add_pad_template (gstelement_class,
693 gst_static_pad_template_get (&rtptemplate));
695 gst_element_class_set_static_metadata (gstelement_class,
696 "RTSP packet receiver", "Source/Network",
697 "Receive data over the network via RTSP (RFC 2326)",
698 "Wim Taymans <wim@fluendo.com>, "
699 "Thijs Vermeir <thijs.vermeir@barco.com>, "
700 "Lutz Mueller <lutz@topfrose.de>");
702 gstbin_class->handle_message = gst_rtspsrc_handle_message;
704 gst_rtsp_ext_list_init ();
708 gst_rtspsrc_init (GstRTSPSrc * src)
710 src->conninfo.location = g_strdup (DEFAULT_LOCATION);
711 src->protocols = DEFAULT_PROTOCOLS;
712 src->debug = DEFAULT_DEBUG;
713 src->retry = DEFAULT_RETRY;
714 src->udp_timeout = DEFAULT_TIMEOUT;
715 gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
716 src->latency = DEFAULT_LATENCY_MS;
717 src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
718 src->connection_speed = DEFAULT_CONNECTION_SPEED;
719 src->nat_method = DEFAULT_NAT_METHOD;
720 src->do_rtcp = DEFAULT_DO_RTCP;
721 src->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
722 gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
723 src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
724 src->user_id = g_strdup (DEFAULT_USER_ID);
725 src->user_pw = g_strdup (DEFAULT_USER_PW);
726 src->buffer_mode = DEFAULT_BUFFER_MODE;
727 src->client_port_range.min = 0;
728 src->client_port_range.max = 0;
729 src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
730 src->short_header = DEFAULT_SHORT_HEADER;
731 src->probation = DEFAULT_PROBATION;
732 src->udp_reconnect = DEFAULT_UDP_RECONNECT;
733 src->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
734 src->ntp_sync = DEFAULT_NTP_SYNC;
735 src->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
737 src->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
738 src->tls_database = DEFAULT_TLS_DATABASE;
740 /* get a list of all extensions */
741 src->extensions = gst_rtsp_ext_list_get ();
743 /* connect to send signal */
744 gst_rtsp_ext_list_connect (src->extensions, "send",
745 (GCallback) gst_rtspsrc_send_cb, src);
747 /* protects the streaming thread in interleaved mode or the polling
748 * thread in UDP mode. */
749 g_rec_mutex_init (&src->stream_rec_lock);
751 /* protects our state changes from multiple invocations */
752 g_rec_mutex_init (&src->state_rec_lock);
754 src->state = GST_RTSP_STATE_INVALID;
756 GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
760 gst_rtspsrc_finalize (GObject * object)
764 rtspsrc = GST_RTSPSRC (object);
766 gst_rtsp_ext_list_free (rtspsrc->extensions);
767 g_free (rtspsrc->conninfo.location);
768 gst_rtsp_url_free (rtspsrc->conninfo.url);
769 g_free (rtspsrc->conninfo.url_str);
770 g_free (rtspsrc->user_id);
771 g_free (rtspsrc->user_pw);
772 g_free (rtspsrc->multi_iface);
775 gst_sdp_message_free (rtspsrc->sdp);
778 if (rtspsrc->provided_clock)
779 gst_object_unref (rtspsrc->provided_clock);
782 gst_structure_free (rtspsrc->sdes);
784 if (rtspsrc->tls_database)
785 g_object_unref (rtspsrc->tls_database);
788 g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
789 g_rec_mutex_clear (&rtspsrc->state_rec_lock);
791 G_OBJECT_CLASS (parent_class)->finalize (object);
795 gst_rtspsrc_provide_clock (GstElement * element)
797 GstRTSPSrc *src = GST_RTSPSRC (element);
800 if ((clock = src->provided_clock) != NULL)
801 gst_object_ref (clock);
806 /* a proxy string of the format [user:passwd@]host[:port] */
808 gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
812 g_free (rtsp->proxy_user);
813 rtsp->proxy_user = NULL;
814 g_free (rtsp->proxy_passwd);
815 rtsp->proxy_passwd = NULL;
816 g_free (rtsp->proxy_host);
817 rtsp->proxy_host = NULL;
818 rtsp->proxy_port = 0;
825 /* we allow http:// in front but ignore it */
826 if (g_str_has_prefix (p, "http://"))
829 at = strchr (p, '@');
831 /* look for user:passwd */
832 col = strchr (proxy, ':');
833 if (col == NULL || col > at)
836 rtsp->proxy_user = g_strndup (p, col - p);
838 rtsp->proxy_passwd = g_strndup (col, at - col);
843 if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
844 rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
845 if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
846 rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
847 if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
848 GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
849 GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
852 col = strchr (p, ':');
855 /* everything before the colon is the hostname */
856 rtsp->proxy_host = g_strndup (p, col - p);
858 rtsp->proxy_port = strtoul (p, (char **) &p, 10);
860 rtsp->proxy_host = g_strdup (p);
861 rtsp->proxy_port = 8080;
867 gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
869 rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
870 rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
873 rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
875 rtspsrc->ptcp_timeout = NULL;
879 gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
884 rtspsrc = GST_RTSPSRC (object);
888 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
889 g_value_get_string (value), NULL);
892 rtspsrc->protocols = g_value_get_flags (value);
895 rtspsrc->debug = g_value_get_boolean (value);
898 rtspsrc->retry = g_value_get_uint (value);
901 rtspsrc->udp_timeout = g_value_get_uint64 (value);
903 case PROP_TCP_TIMEOUT:
904 gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
907 rtspsrc->latency = g_value_get_uint (value);
909 case PROP_DROP_ON_LATENCY:
910 rtspsrc->drop_on_latency = g_value_get_boolean (value);
912 case PROP_CONNECTION_SPEED:
913 rtspsrc->connection_speed = g_value_get_uint64 (value);
915 case PROP_NAT_METHOD:
916 rtspsrc->nat_method = g_value_get_enum (value);
919 rtspsrc->do_rtcp = g_value_get_boolean (value);
921 case PROP_DO_RTSP_KEEP_ALIVE:
922 rtspsrc->do_rtsp_keep_alive = g_value_get_boolean (value);
925 gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
928 if (rtspsrc->prop_proxy_id)
929 g_free (rtspsrc->prop_proxy_id);
930 rtspsrc->prop_proxy_id = g_value_dup_string (value);
933 if (rtspsrc->prop_proxy_pw)
934 g_free (rtspsrc->prop_proxy_pw);
935 rtspsrc->prop_proxy_pw = g_value_dup_string (value);
937 case PROP_RTP_BLOCKSIZE:
938 rtspsrc->rtp_blocksize = g_value_get_uint (value);
941 if (rtspsrc->user_id)
942 g_free (rtspsrc->user_id);
943 rtspsrc->user_id = g_value_dup_string (value);
946 if (rtspsrc->user_pw)
947 g_free (rtspsrc->user_pw);
948 rtspsrc->user_pw = g_value_dup_string (value);
950 case PROP_BUFFER_MODE:
951 rtspsrc->buffer_mode = g_value_get_enum (value);
953 case PROP_PORT_RANGE:
957 str = g_value_get_string (value);
959 sscanf (str, "%u-%u",
960 &rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
962 rtspsrc->client_port_range.min = 0;
963 rtspsrc->client_port_range.max = 0;
967 case PROP_UDP_BUFFER_SIZE:
968 rtspsrc->udp_buffer_size = g_value_get_int (value);
970 case PROP_SHORT_HEADER:
971 rtspsrc->short_header = g_value_get_boolean (value);
974 rtspsrc->probation = g_value_get_uint (value);
976 case PROP_UDP_RECONNECT:
977 rtspsrc->udp_reconnect = g_value_get_boolean (value);
979 case PROP_MULTICAST_IFACE:
980 g_free (rtspsrc->multi_iface);
982 if (g_value_get_string (value) == NULL)
983 rtspsrc->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
985 rtspsrc->multi_iface = g_value_dup_string (value);
988 rtspsrc->ntp_sync = g_value_get_boolean (value);
990 case PROP_USE_PIPELINE_CLOCK:
991 rtspsrc->use_pipeline_clock = g_value_get_boolean (value);
994 rtspsrc->sdes = g_value_dup_boxed (value);
996 case PROP_TLS_VALIDATION_FLAGS:
997 rtspsrc->tls_validation_flags = g_value_get_flags (value);
999 case PROP_TLS_DATABASE:
1000 g_clear_object (&rtspsrc->tls_database);
1001 rtspsrc->tls_database = g_value_dup_object (value);
1004 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1010 gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
1013 GstRTSPSrc *rtspsrc;
1015 rtspsrc = GST_RTSPSRC (object);
1019 g_value_set_string (value, rtspsrc->conninfo.location);
1021 case PROP_PROTOCOLS:
1022 g_value_set_flags (value, rtspsrc->protocols);
1025 g_value_set_boolean (value, rtspsrc->debug);
1028 g_value_set_uint (value, rtspsrc->retry);
1031 g_value_set_uint64 (value, rtspsrc->udp_timeout);
1033 case PROP_TCP_TIMEOUT:
1037 timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
1038 rtspsrc->tcp_timeout.tv_usec;
1039 g_value_set_uint64 (value, timeout);
1043 g_value_set_uint (value, rtspsrc->latency);
1045 case PROP_DROP_ON_LATENCY:
1046 g_value_set_boolean (value, rtspsrc->drop_on_latency);
1048 case PROP_CONNECTION_SPEED:
1049 g_value_set_uint64 (value, rtspsrc->connection_speed);
1051 case PROP_NAT_METHOD:
1052 g_value_set_enum (value, rtspsrc->nat_method);
1055 g_value_set_boolean (value, rtspsrc->do_rtcp);
1057 case PROP_DO_RTSP_KEEP_ALIVE:
1058 g_value_set_boolean (value, rtspsrc->do_rtsp_keep_alive);
1064 if (rtspsrc->proxy_host) {
1066 g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
1070 g_value_take_string (value, str);
1074 g_value_set_string (value, rtspsrc->prop_proxy_id);
1077 g_value_set_string (value, rtspsrc->prop_proxy_pw);
1079 case PROP_RTP_BLOCKSIZE:
1080 g_value_set_uint (value, rtspsrc->rtp_blocksize);
1083 g_value_set_string (value, rtspsrc->user_id);
1086 g_value_set_string (value, rtspsrc->user_pw);
1088 case PROP_BUFFER_MODE:
1089 g_value_set_enum (value, rtspsrc->buffer_mode);
1091 case PROP_PORT_RANGE:
1095 if (rtspsrc->client_port_range.min != 0) {
1096 str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
1097 rtspsrc->client_port_range.max);
1101 g_value_take_string (value, str);
1104 case PROP_UDP_BUFFER_SIZE:
1105 g_value_set_int (value, rtspsrc->udp_buffer_size);
1107 case PROP_SHORT_HEADER:
1108 g_value_set_boolean (value, rtspsrc->short_header);
1110 case PROP_PROBATION:
1111 g_value_set_uint (value, rtspsrc->probation);
1113 case PROP_UDP_RECONNECT:
1114 g_value_set_boolean (value, rtspsrc->udp_reconnect);
1116 case PROP_MULTICAST_IFACE:
1117 g_value_set_string (value, rtspsrc->multi_iface);
1120 g_value_set_boolean (value, rtspsrc->ntp_sync);
1122 case PROP_USE_PIPELINE_CLOCK:
1123 g_value_set_boolean (value, rtspsrc->use_pipeline_clock);
1126 g_value_set_boxed (value, rtspsrc->sdes);
1128 case PROP_TLS_VALIDATION_FLAGS:
1129 g_value_set_flags (value, rtspsrc->tls_validation_flags);
1131 case PROP_TLS_DATABASE:
1132 g_value_set_object (value, rtspsrc->tls_database);
1135 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1141 find_stream_by_id (GstRTSPStream * stream, gint * id)
1143 if (stream->id == *id)
1150 find_stream_by_channel (GstRTSPStream * stream, gint * channel)
1152 if (stream->channel[0] == *channel || stream->channel[1] == *channel)
1159 find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
1161 GstElement *src = (GstElement *) a;
1163 if (stream->udpsrc[0] == src)
1165 if (stream->udpsrc[1] == src)
1172 find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
1174 if (stream->conninfo.location) {
1175 /* check qualified setup_url */
1176 if (!strcmp (stream->conninfo.location, (gchar *) a))
1179 if (stream->control_url) {
1180 /* check original control_url */
1181 if (!strcmp (stream->control_url, (gchar *) a))
1184 /* check if qualified setup_url ends with string */
1185 if (g_str_has_suffix (stream->control_url, (gchar *) a))
1192 static GstRTSPStream *
1193 find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
1197 /* find and get stream */
1198 if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
1199 return (GstRTSPStream *) lstream->data;
1204 static const GstSDPBandwidth *
1205 gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1206 const GstSDPMedia * media, const gchar * type)
1210 /* first look in the media specific section */
1211 len = gst_sdp_media_bandwidths_len (media);
1212 for (i = 0; i < len; i++) {
1213 const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
1215 if (strcmp (bw->bwtype, type) == 0)
1218 /* then look in the message specific section */
1219 len = gst_sdp_message_bandwidths_len (sdp);
1220 for (i = 0; i < len; i++) {
1221 const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
1223 if (strcmp (bw->bwtype, type) == 0)
1230 gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1231 const GstSDPMedia * media, GstRTSPStream * stream)
1233 const GstSDPBandwidth *bw;
1235 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
1236 stream->as_bandwidth = bw->bandwidth;
1238 stream->as_bandwidth = -1;
1240 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
1241 stream->rr_bandwidth = bw->bandwidth;
1243 stream->rr_bandwidth = -1;
1245 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
1246 stream->rs_bandwidth = bw->bandwidth;
1248 stream->rs_bandwidth = -1;
1252 gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
1253 const GstSDPConnection * conn)
1255 if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
1258 if (conn->addrtype == NULL)
1261 /* check for IPV6 */
1262 if (strcmp (conn->addrtype, "IP4") == 0)
1263 stream->is_ipv6 = FALSE;
1264 else if (strcmp (conn->addrtype, "IP6") == 0)
1265 stream->is_ipv6 = TRUE;
1270 g_free (stream->destination);
1271 stream->destination = g_strdup (conn->address);
1273 /* check for multicast */
1274 stream->is_multicast =
1275 gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
1277 stream->ttl = conn->ttl;
1280 /* Go over the connections for a stream.
1281 * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
1283 * - If we are dealing with a localhost address, we disable multicast
1286 gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
1287 const GstSDPMedia * media, GstRTSPStream * stream)
1289 const GstSDPConnection *conn;
1292 /* first look in the media specific section */
1293 len = gst_sdp_media_connections_len (media);
1294 for (i = 0; i < len; i++) {
1295 conn = gst_sdp_media_get_connection (media, i);
1297 gst_rtspsrc_do_stream_connection (src, stream, conn);
1299 /* then look in the message specific section */
1300 if ((conn = gst_sdp_message_get_connection (sdp))) {
1301 gst_rtspsrc_do_stream_connection (src, stream, conn);
1305 /* m=<media> <UDP port> RTP/AVP <payload>
1308 gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
1309 const GstSDPMedia * media, GstRTSPStream * stream)
1315 proto = gst_sdp_media_get_proto (media);
1319 if (g_str_equal (proto, "RTP/AVP"))
1320 stream->profile = GST_RTSP_PROFILE_AVP;
1321 else if (g_str_equal (proto, "RTP/SAVP"))
1322 stream->profile = GST_RTSP_PROFILE_SAVP;
1323 else if (g_str_equal (proto, "RTP/AVPF"))
1324 stream->profile = GST_RTSP_PROFILE_AVPF;
1325 else if (g_str_equal (proto, "RTP/SAVPF"))
1326 stream->profile = GST_RTSP_PROFILE_SAVPF;
1330 len = gst_sdp_media_formats_len (media);
1331 for (i = 0; i < len; i++) {
1338 pt = atoi (gst_sdp_media_get_format (media, i));
1340 GST_DEBUG_OBJECT (src, " looking at %d pt: %d", i, pt);
1343 caps = gst_rtspsrc_media_to_caps (pt, media);
1345 GST_WARNING_OBJECT (src, " skipping pt %d without caps", pt);
1349 /* do some tweaks */
1350 s = gst_caps_get_structure (caps, 0);
1351 if ((enc = gst_structure_get_string (s, "encoding-name"))) {
1352 stream->is_real = (strstr (enc, "-REAL") != NULL);
1353 if (strcmp (enc, "X-ASF-PF") == 0)
1354 stream->container = TRUE;
1356 GST_DEBUG ("mapping sdp session level attributes to caps");
1357 gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, caps);
1358 GST_DEBUG ("mapping sdp media level attributes to caps");
1359 gst_rtspsrc_sdp_attributes_to_caps (media->attributes, caps);
1361 /* the first pt will be the default */
1362 if (stream->ptmap->len == 0)
1363 stream->default_pt = pt;
1367 g_array_append_val (stream->ptmap, item);
1373 GST_ERROR_OBJECT (src, "can't find proto in media");
1378 GST_ERROR_OBJECT (src, "unknown proto in media %s", proto);
1383 static const gchar *
1384 get_aggregate_control (GstRTSPSrc * src)
1389 base = src->control;
1390 else if (src->content_base)
1391 base = src->content_base;
1392 else if (src->conninfo.url_str)
1393 base = src->conninfo.url_str;
1401 clear_ptmap_item (PtMapItem * item)
1404 gst_caps_unref (item->caps);
1407 static GstRTSPStream *
1408 gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
1410 GstRTSPStream *stream;
1411 const gchar *control_url;
1412 const GstSDPMedia *media;
1414 /* get media, should not return NULL */
1415 media = gst_sdp_message_get_media (sdp, idx);
1419 stream = g_new0 (GstRTSPStream, 1);
1420 stream->parent = src;
1421 /* we mark the pad as not linked, we will mark it as OK when we add the pad to
1423 stream->last_ret = GST_FLOW_NOT_LINKED;
1424 stream->added = FALSE;
1425 stream->setup = FALSE;
1426 stream->skipped = FALSE;
1428 stream->eos = FALSE;
1429 stream->discont = TRUE;
1430 stream->seqbase = -1;
1431 stream->timebase = -1;
1432 stream->profile = GST_RTSP_PROFILE_AVP;
1433 stream->ptmap = g_array_new (FALSE, FALSE, sizeof (PtMapItem));
1434 g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);
1436 /* collect bandwidth information for this steam. FIXME, configure in the RTP
1437 * session manager to scale RTCP. */
1438 gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
1440 /* collect connection info */
1441 gst_rtspsrc_collect_connections (src, sdp, media, stream);
1443 /* make the payload type map */
1444 gst_rtspsrc_collect_payloads (src, sdp, media, stream);
1446 /* collect port number */
1447 stream->port = gst_sdp_media_get_port (media);
1449 /* get control url to construct the setup url. The setup url is used to
1450 * configure the transport of the stream and is used to identity the stream in
1451 * the RTP-Info header field returned from PLAY. */
1452 control_url = gst_sdp_media_get_attribute_val (media, "control");
1453 if (control_url == NULL)
1454 control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
1456 GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
1457 GST_DEBUG_OBJECT (src, " port: %d", stream->port);
1458 GST_DEBUG_OBJECT (src, " container: %d", stream->container);
1459 GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
1461 if (control_url != NULL) {
1462 stream->control_url = g_strdup (control_url);
1463 /* Build a fully qualified url using the content_base if any or by prefixing
1464 * the original request.
1465 * If the control_url starts with a '/' or a non rtsp: protocol we will most
1466 * likely build a URL that the server will fail to understand, this is ok,
1467 * we will fail then. */
1468 if (g_str_has_prefix (control_url, "rtsp://"))
1469 stream->conninfo.location = g_strdup (control_url);
1474 if (g_strcmp0 (control_url, "*") == 0)
1477 base = get_aggregate_control (src);
1479 /* check if the base ends or control starts with / */
1480 has_slash = g_str_has_prefix (control_url, "/");
1481 has_slash = has_slash || g_str_has_suffix (base, "/");
1483 /* concatenate the two strings, insert / when not present */
1484 stream->conninfo.location =
1485 g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
1488 GST_DEBUG_OBJECT (src, " setup: %s",
1489 GST_STR_NULL (stream->conninfo.location));
1491 /* we keep track of all streams */
1492 src->streams = g_list_append (src->streams, stream);
1500 gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
1504 GST_DEBUG_OBJECT (src, "free stream %p", stream);
1506 g_array_free (stream->ptmap, TRUE);
1508 g_free (stream->destination);
1509 g_free (stream->control_url);
1510 g_free (stream->conninfo.location);
1512 for (i = 0; i < 2; i++) {
1513 if (stream->udpsrc[i]) {
1514 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
1515 gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
1516 gst_object_unref (stream->udpsrc[i]);
1518 if (stream->channelpad[i])
1519 gst_object_unref (stream->channelpad[i]);
1521 if (stream->udpsink[i]) {
1522 gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
1523 gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
1524 gst_object_unref (stream->udpsink[i]);
1527 if (stream->fakesrc) {
1528 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
1529 gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
1530 gst_object_unref (stream->fakesrc);
1532 if (stream->srcpad) {
1533 gst_pad_set_active (stream->srcpad, FALSE);
1535 gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
1537 if (stream->srtpenc)
1538 gst_object_unref (stream->srtpenc);
1539 if (stream->srtpdec)
1540 gst_object_unref (stream->srtpdec);
1542 gst_buffer_unref (stream->key);
1543 if (stream->rtcppad)
1544 gst_object_unref (stream->rtcppad);
1545 if (stream->session)
1546 g_object_unref (stream->session);
1551 gst_rtspsrc_cleanup (GstRTSPSrc * src)
1555 GST_DEBUG_OBJECT (src, "cleanup");
1557 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1558 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1560 gst_rtspsrc_stream_free (src, stream);
1562 g_list_free (src->streams);
1563 src->streams = NULL;
1565 if (src->manager_sig_id) {
1566 g_signal_handler_disconnect (src->manager, src->manager_sig_id);
1567 src->manager_sig_id = 0;
1569 gst_element_set_state (src->manager, GST_STATE_NULL);
1570 gst_bin_remove (GST_BIN_CAST (src), src->manager);
1571 src->manager = NULL;
1574 gst_structure_free (src->props);
1577 g_free (src->content_base);
1578 src->content_base = NULL;
1580 g_free (src->control);
1581 src->control = NULL;
1584 gst_rtsp_range_free (src->range);
1587 /* don't clear the SDP when it was used in the url */
1588 if (src->sdp && !src->from_sdp) {
1589 gst_sdp_message_free (src->sdp);
1592 if (src->start_segment) {
1593 gst_event_unref (src->start_segment);
1594 src->start_segment = NULL;
1596 if (src->provided_clock) {
1597 gst_object_unref (src->provided_clock);
1598 src->provided_clock = NULL;
1602 #define PARSE_INT(p, del, res) \
1605 p = strstr (p, del); \
1615 #define PARSE_STRING(p, del, res) \
1618 p = strstr (p, del); \
1630 #define SKIP_SPACES(p) \
1631 while (*p && g_ascii_isspace (*p)) \
1636 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1639 gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
1640 gint * rate, gchar ** params)
1644 p = (gchar *) rtpmap;
1646 PARSE_INT (p, " ", *payload);
1654 PARSE_STRING (p, "/", *name);
1655 if (*name == NULL) {
1656 GST_DEBUG ("no rate, name %s", p);
1657 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
1658 * streams seem to omit the rate. */
1665 p = strstr (p, "/");
1683 parse_keymgmt (const gchar * keymgmt, GstCaps * caps)
1685 gboolean res = FALSE;
1689 GstMIKEYMessage *msg;
1690 const GstMIKEYPayload *payload;
1691 const gchar *srtp_cipher;
1692 const gchar *srtp_auth;
1694 p = (gchar *) keymgmt;
1700 PARSE_STRING (p, " ", kmpid);
1701 if (!g_str_equal (kmpid, "mikey"))
1704 data = g_base64_decode (p, &size);
1708 msg = gst_mikey_message_new_from_data (data, size);
1712 srtp_cipher = "aes-128-icm";
1713 srtp_auth = "hmac-sha1-80";
1715 /* check the Security policy if any */
1716 if ((payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, 0))) {
1717 GstMIKEYPayloadSP *p = (GstMIKEYPayloadSP *) payload;
1720 if (p->proto != GST_MIKEY_SEC_PROTO_SRTP)
1723 len = gst_mikey_payload_sp_get_n_params (payload);
1724 for (i = 0; i < len; i++) {
1725 const GstMIKEYPayloadSPParam *param =
1726 gst_mikey_payload_sp_get_param (payload, i);
1728 switch (param->type) {
1729 case GST_MIKEY_SP_SRTP_ENC_ALG:
1730 switch (param->val[0]) {
1732 srtp_cipher = "null";
1736 srtp_cipher = "aes-128-icm";
1742 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1743 switch (param->val[0]) {
1749 srtp_auth = "hmac-sha1-80";
1755 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1757 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1765 if (!(payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_KEMAC, 0)))
1768 GstMIKEYPayloadKEMAC *p = (GstMIKEYPayloadKEMAC *) payload;
1771 if (p->enc_alg != GST_MIKEY_ENC_NULL || p->mac_alg != GST_MIKEY_MAC_NULL)
1775 gst_buffer_new_wrapped (g_memdup (p->enc_data, p->enc_len), p->enc_len);
1776 gst_caps_set_simple (caps, "srtp-key", GST_TYPE_BUFFER, buf, NULL);
1779 gst_caps_set_simple (caps,
1780 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
1781 "srtp-auth", G_TYPE_STRING, srtp_auth,
1782 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
1783 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
1787 gst_mikey_message_free (msg);
1793 * Mapping SDP attributes to caps
1795 * prepend 'a-' to IANA registered sdp attributes names
1796 * (ie: not prefixed with 'x-') in order to avoid
1797 * collision with gstreamer standard caps properties names
1800 gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
1802 if (attributes->len > 0) {
1806 s = gst_caps_get_structure (caps, 0);
1808 for (i = 0; i < attributes->len; i++) {
1809 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
1810 gchar *tofree, *key;
1814 /* skip some of the attribute we already handle */
1815 if (!strcmp (key, "fmtp"))
1817 if (!strcmp (key, "rtpmap"))
1819 if (!strcmp (key, "control"))
1821 if (!strcmp (key, "range"))
1823 if (g_str_equal (key, "key-mgmt")) {
1824 parse_keymgmt (attr->value, caps);
1828 /* string must be valid UTF8 */
1829 if (!g_utf8_validate (attr->value, -1, NULL))
1832 if (!g_str_has_prefix (key, "x-"))
1833 tofree = key = g_strdup_printf ("a-%s", key);
1837 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
1838 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
1844 static const gchar *
1845 rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
1854 if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
1857 if (sscanf (attr, "%d ", &val) != 1)
1867 * Mapping of caps to and from SDP fields:
1869 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1870 * a=fmtp:<payload> <param>[=<value>];...
1873 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
1876 const gchar *rtpmap;
1880 gchar *params = NULL;
1886 /* get and parse rtpmap */
1887 rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
1890 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
1892 g_warning ("error parsing rtpmap, ignoring");
1896 /* dynamic payloads need rtpmap or we fail */
1897 if (rtpmap == NULL && pt >= 96)
1900 /* check if we have a rate, if not, we need to look up the rate from the
1901 * default rates based on the payload types. */
1903 const GstRTPPayloadInfo *info;
1905 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
1906 /* dynamic types, use media and encoding_name */
1907 tmp = g_ascii_strdown (media->media, -1);
1908 info = gst_rtp_payload_info_for_name (tmp, name);
1911 /* static types, use payload type */
1912 info = gst_rtp_payload_info_for_pt (pt);
1916 if ((rate = info->clock_rate) == 0)
1919 /* we fail if we cannot find one */
1924 tmp = g_ascii_strdown (media->media, -1);
1925 caps = gst_caps_new_simple ("application/x-unknown",
1926 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
1928 s = gst_caps_get_structure (caps, 0);
1930 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
1932 /* encoding name must be upper case */
1934 tmp = g_ascii_strup (name, -1);
1935 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
1939 /* params must be lower case */
1940 if (params != NULL) {
1941 tmp = g_ascii_strdown (params, -1);
1942 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
1946 /* parse optional fmtp: field */
1947 if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
1953 /* p is now of the format <payload> <param>[=<value>];... */
1954 PARSE_INT (p, " ", payload);
1955 if (payload != -1 && payload == pt) {
1959 /* <param>[=<value>] are separated with ';' */
1960 pairs = g_strsplit (p, ";", 0);
1961 for (i = 0; pairs[i]; i++) {
1963 const gchar *val, *key;
1965 /* the key may not have a '=', the value can have other '='s */
1966 valpos = strstr (pairs[i], "=");
1968 /* we have a '=' and thus a value, remove the '=' with \0 */
1970 /* value is everything between '=' and ';'. We split the pairs at ;
1971 * boundaries so we can take the remainder of the value. Some servers
1972 * put spaces around the value which we strip off here. Alternatively
1973 * we could strip those spaces in the depayloaders should these spaces
1974 * actually carry any meaning in the future. */
1975 val = g_strstrip (valpos + 1);
1977 /* simple <param>;.. is translated into <param>=1;... */
1980 /* strip the key of spaces, convert key to lowercase but not the value. */
1981 key = g_strstrip (pairs[i]);
1982 if (strlen (key) > 1) {
1983 tmp = g_ascii_strdown (key, -1);
1984 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
1996 g_warning ("rtpmap type not given for dynamic payload %d", pt);
2001 g_warning ("rate unknown for payload type %d", pt);
2007 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
2008 gint * rtpport, gint * rtcpport)
2011 GstStateChangeReturn ret;
2012 GstElement *udpsrc0, *udpsrc1;
2013 gint tmp_rtp, tmp_rtcp;
2017 src = stream->parent;
2023 /* Start at next port */
2024 tmp_rtp = src->next_port_num;
2026 if (stream->is_ipv6)
2027 host = "udp://[::0]";
2029 host = "udp://0.0.0.0";
2031 /* try to allocate 2 UDP ports, the RTP port should be an even
2032 * number and the RTCP port should be the next (uneven) port */
2035 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
2036 tmp_rtp >= src->client_port_range.max)
2039 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2040 if (udpsrc0 == NULL)
2041 goto no_udp_protocol;
2042 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
2044 if (src->udp_buffer_size != 0)
2045 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
2048 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
2049 if (ret == GST_STATE_CHANGE_FAILURE) {
2051 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
2054 if (++count > src->retry)
2057 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2058 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2059 gst_object_unref (udpsrc0);
2062 GST_DEBUG_OBJECT (src, "retry %d", count);
2065 goto no_udp_protocol;
2068 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
2069 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
2071 /* check if port is even */
2072 if ((tmp_rtp & 0x01) != 0) {
2073 /* port not even, close and allocate another */
2074 if (++count > src->retry)
2077 GST_DEBUG_OBJECT (src, "RTP port not even");
2079 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2080 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2081 gst_object_unref (udpsrc0);
2084 GST_DEBUG_OBJECT (src, "retry %d", count);
2089 /* allocate port+1 for RTCP now */
2090 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2091 if (udpsrc1 == NULL)
2092 goto no_udp_rtcp_protocol;
2095 tmp_rtcp = tmp_rtp + 1;
2096 if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
2099 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
2101 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
2102 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
2103 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
2104 if (ret == GST_STATE_CHANGE_FAILURE) {
2105 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
2107 if (++count > src->retry)
2110 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2111 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2112 gst_object_unref (udpsrc0);
2115 GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
2116 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2117 gst_object_unref (udpsrc1);
2121 GST_DEBUG_OBJECT (src, "retry %d", count);
2125 /* all fine, do port check */
2126 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
2127 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
2129 /* this should not happen... */
2130 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
2133 /* we keep these elements, we configure all in configure_transport when the
2134 * server told us to really use the UDP ports. */
2135 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
2136 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
2137 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
2138 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
2140 /* keep track of next available port number when we have a range
2142 if (src->next_port_num != 0)
2143 src->next_port_num = tmp_rtcp + 1;
2150 GST_DEBUG_OBJECT (src, "could not get UDP source");
2155 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
2159 no_udp_rtcp_protocol:
2161 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
2166 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
2167 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
2173 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2174 gst_object_unref (udpsrc0);
2177 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2178 gst_object_unref (udpsrc1);
2185 gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
2190 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
2192 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2193 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2196 for (i = 0; i < 2; i++) {
2197 if (stream->udpsrc[i])
2198 gst_element_set_state (stream->udpsrc[i], state);
2204 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
2211 event = gst_event_new_flush_start ();
2212 GST_DEBUG_OBJECT (src, "start flush");
2214 state = GST_STATE_PAUSED;
2216 event = gst_event_new_flush_stop (FALSE);
2217 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
2220 state = GST_STATE_PLAYING;
2222 state = GST_STATE_PAUSED;
2224 gst_rtspsrc_push_event (src, event);
2225 gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
2226 gst_rtspsrc_set_state (src, state);
2229 static GstRTSPResult
2230 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
2231 GstRTSPMessage * message, GTimeVal * timeout)
2236 ret = gst_rtsp_connection_send (conn, message, timeout);
2238 ret = GST_RTSP_ERROR;
2243 static GstRTSPResult
2244 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
2245 GstRTSPMessage * message, GTimeVal * timeout)
2250 ret = gst_rtsp_connection_receive (conn, message, timeout);
2252 ret = GST_RTSP_ERROR;
2258 gst_rtspsrc_get_position (GstRTSPSrc * src)
2263 query = gst_query_new_position (GST_FORMAT_TIME);
2264 /* should be known somewhere down the stream (e.g. jitterbuffer) */
2265 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2266 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2270 if (stream->srcpad) {
2271 if (gst_pad_query (stream->srcpad, query)) {
2272 gst_query_parse_position (query, &fmt, &pos);
2273 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
2274 GST_TIME_ARGS (pos));
2275 src->last_pos = pos;
2285 gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
2287 src->state = GST_RTSP_STATE_SEEKING;
2288 /* PLAY will add the range header now. */
2289 src->need_range = TRUE;
2295 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
2300 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
2302 gboolean flush, skip;
2305 GstSegment seeksegment = { 0, };
2309 GST_DEBUG_OBJECT (src, "doing seek with event");
2311 gst_event_parse_seek (event, &rate, &format, &flags,
2312 &cur_type, &cur, &stop_type, &stop);
2314 /* no negative rates yet */
2318 /* we need TIME format */
2319 if (format != src->segment.format)
2322 GST_DEBUG_OBJECT (src, "doing seek without event");
2324 cur_type = GST_SEEK_TYPE_SET;
2325 stop_type = GST_SEEK_TYPE_SET;
2328 /* get flush flag */
2329 flush = flags & GST_SEEK_FLAG_FLUSH;
2330 skip = flags & GST_SEEK_FLAG_SKIP;
2332 /* now we need to make sure the streaming thread is stopped. We do this by
2333 * either sending a FLUSH_START event downstream which will cause the
2334 * streaming thread to stop with a WRONG_STATE.
2335 * For a non-flushing seek we simply pause the task, which will happen as soon
2336 * as it completes one iteration (and thus might block when the sink is
2337 * blocking in preroll). */
2339 GST_DEBUG_OBJECT (src, "starting flush");
2340 gst_rtspsrc_flush (src, TRUE, FALSE);
2343 gst_task_pause (src->task);
2347 /* we should now be able to grab the streaming thread because we stopped it
2348 * with the above flush/pause code */
2349 GST_RTSP_STREAM_LOCK (src);
2351 GST_DEBUG_OBJECT (src, "stopped streaming");
2353 /* stop flushing the rtsp connection so we can send PAUSE/PLAY below */
2354 gst_rtspsrc_connection_flush (src, FALSE);
2356 /* copy segment, we need this because we still need the old
2357 * segment when we close the current segment. */
2358 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
2360 /* configure the seek parameters in the seeksegment. We will then have the
2361 * right values in the segment to perform the seek */
2363 GST_DEBUG_OBJECT (src, "configuring seek");
2364 gst_segment_do_seek (&seeksegment, rate, format, flags,
2365 cur_type, cur, stop_type, stop, &update);
2368 /* figure out the last position we need to play. If it's configured (stop !=
2369 * -1), use that, else we play until the total duration of the file */
2370 if ((stop = seeksegment.stop) == -1)
2371 stop = seeksegment.duration;
2373 playing = (src->state == GST_RTSP_STATE_PLAYING);
2375 /* if we were playing, pause first */
2377 /* obtain current position in case seek fails */
2378 gst_rtspsrc_get_position (src);
2379 gst_rtspsrc_pause (src, FALSE);
2383 gst_rtspsrc_do_seek (src, &seeksegment);
2385 /* and continue playing */
2387 gst_rtspsrc_play (src, &seeksegment, FALSE);
2389 /* prepare for streaming again */
2391 /* if we started flush, we stop now */
2392 GST_DEBUG_OBJECT (src, "stopping flush");
2393 gst_rtspsrc_flush (src, FALSE, playing);
2396 /* now we did the seek and can activate the new segment values */
2397 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
2399 /* if we're doing a segment seek, post a SEGMENT_START message */
2400 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2401 gst_element_post_message (GST_ELEMENT_CAST (src),
2402 gst_message_new_segment_start (GST_OBJECT_CAST (src),
2403 src->segment.format, src->segment.position));
2406 /* now create the newsegment */
2407 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
2408 " to %" G_GINT64_FORMAT, src->segment.position, stop);
2411 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
2412 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2413 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2414 stream->discont = TRUE;
2417 GST_RTSP_STREAM_UNLOCK (src);
2424 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
2429 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
2435 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
2439 gboolean res = TRUE;
2442 src = GST_RTSPSRC_CAST (parent);
2444 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
2445 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
2447 switch (GST_EVENT_TYPE (event)) {
2448 case GST_EVENT_SEEK:
2449 res = gst_rtspsrc_perform_seek (src, event);
2453 case GST_EVENT_NAVIGATION:
2454 case GST_EVENT_LATENCY:
2462 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
2463 res = gst_pad_send_event (target, event);
2464 gst_object_unref (target);
2466 gst_event_unref (event);
2469 gst_event_unref (event);
2475 /* this is the final event function we receive on the internal source pad when
2476 * we deal with TCP connections */
2478 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
2483 GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
2485 switch (GST_EVENT_TYPE (event)) {
2486 case GST_EVENT_SEEK:
2488 case GST_EVENT_NAVIGATION:
2489 case GST_EVENT_LATENCY:
2491 gst_event_unref (event);
2498 /* this is the final query function we receive on the internal source pad when
2499 * we deal with TCP connections */
2501 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
2505 gboolean res = TRUE;
2507 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
2509 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2510 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2512 switch (GST_QUERY_TYPE (query)) {
2513 case GST_QUERY_POSITION:
2518 case GST_QUERY_DURATION:
2522 gst_query_parse_duration (query, &format, NULL);
2525 case GST_FORMAT_TIME:
2526 gst_query_set_duration (query, format, src->segment.duration);
2534 case GST_QUERY_LATENCY:
2536 /* we are live with a min latency of 0 and unlimited max latency, this
2537 * result will be updated by the session manager if there is any. */
2538 gst_query_set_latency (query, TRUE, 0, -1);
2548 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2550 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2554 gboolean res = FALSE;
2556 src = GST_RTSPSRC_CAST (parent);
2558 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2559 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2561 switch (GST_QUERY_TYPE (query)) {
2562 case GST_QUERY_DURATION:
2566 gst_query_parse_duration (query, &format, NULL);
2569 case GST_FORMAT_TIME:
2570 gst_query_set_duration (query, format, src->segment.duration);
2578 case GST_QUERY_SEEKING:
2582 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2583 if (format == GST_FORMAT_TIME) {
2585 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2587 /* seeking without duration is unlikely */
2588 seekable = seekable && src->seekable && src->segment.duration &&
2589 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2591 /* FIXME ?? should we have 0 and segment.duration here; see demuxers */
2592 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
2593 src->segment.start, src->segment.stop);
2602 uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
2604 gst_query_set_uri (query, uri);
2612 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2614 /* forward the query to the proxy target pad */
2616 res = gst_pad_query (target, query);
2617 gst_object_unref (target);
2626 /* callback for RTCP messages to be sent to the server when operating in TCP
2628 static GstFlowReturn
2629 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2632 GstRTSPStream *stream;
2633 GstFlowReturn res = GST_FLOW_OK;
2638 GstRTSPMessage message = { 0 };
2639 GstRTSPConnection *conn;
2641 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2642 src = stream->parent;
2644 gst_buffer_map (buffer, &map, GST_MAP_READ);
2648 gst_rtsp_message_init_data (&message, stream->channel[1]);
2650 /* lend the body data to the message */
2651 gst_rtsp_message_take_body (&message, data, size);
2653 if (stream->conninfo.connection)
2654 conn = stream->conninfo.connection;
2656 conn = src->conninfo.connection;
2658 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2659 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2660 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2662 /* and steal it away again because we will free it when unreffing the
2664 gst_rtsp_message_steal_body (&message, &data, &size);
2665 gst_rtsp_message_unset (&message);
2667 gst_buffer_unmap (buffer, &map);
2668 gst_buffer_unref (buffer);
2673 static GstPadProbeReturn
2674 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2676 GstRTSPSrc *src = user_data;
2678 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2679 GST_DEBUG_PAD_NAME (pad));
2681 /* activate the streams */
2682 GST_OBJECT_LOCK (src);
2683 if (!src->need_activate)
2686 src->need_activate = FALSE;
2687 GST_OBJECT_UNLOCK (src);
2689 gst_rtspsrc_activate_streams (src);
2691 return GST_PAD_PROBE_OK;
2695 GST_OBJECT_UNLOCK (src);
2696 return GST_PAD_PROBE_OK;
2701 copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
2703 GstPad *gpad = GST_PAD_CAST (user_data);
2705 GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
2706 gst_pad_store_sticky_event (gpad, *event);
2711 /* this callback is called when the session manager generated a new src pad with
2712 * payloaded RTP packets. We simply ghost the pad here. */
2714 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2717 GstPadTemplate *template;
2720 GstRTSPStream *stream;
2723 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2725 GST_RTSP_STATE_LOCK (src);
2727 name = gst_object_get_name (GST_OBJECT_CAST (pad));
2728 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
2729 goto unknown_stream;
2731 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);
2733 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
2735 goto unknown_stream;
2738 stream->ssrc = ssrc;
2740 /* we'll add it later see below */
2741 stream->added = TRUE;
2743 /* check if we added all streams */
2745 for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
2746 GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;
2748 GST_DEBUG_OBJECT (src, "stream %p, container %d, added %d, setup %d",
2749 ostream, ostream->container, ostream->added, ostream->setup);
2751 /* if we find a stream for which we did a setup that is not added, we
2752 * need to wait some more */
2753 if (ostream->setup && !ostream->added) {
2758 GST_RTSP_STATE_UNLOCK (src);
2760 /* create a new pad we will use to stream to */
2761 template = gst_static_pad_template_get (&rtptemplate);
2762 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
2763 gst_object_unref (template);
2766 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
2767 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
2768 gst_pad_set_active (stream->srcpad, TRUE);
2769 gst_pad_sticky_events_foreach (pad, copy_sticky_events, stream->srcpad);
2770 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
2773 GST_DEBUG_OBJECT (src, "We added all streams");
2774 /* when we get here, all stream are added and we can fire the no-more-pads
2776 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
2784 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
2785 GST_RTSP_STATE_UNLOCK (src);
2792 stream_get_caps_for_pt (GstRTSPStream * stream, guint pt)
2796 len = stream->ptmap->len;
2797 for (i = 0; i < len; i++) {
2798 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
2806 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
2808 GstRTSPStream *stream;
2811 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
2813 GST_RTSP_STATE_LOCK (src);
2814 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2816 goto unknown_stream;
2818 if ((caps = stream_get_caps_for_pt (stream, pt)))
2819 gst_caps_ref (caps);
2820 GST_RTSP_STATE_UNLOCK (src);
2826 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
2827 GST_RTSP_STATE_UNLOCK (src);
2833 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
2835 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
2841 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
2847 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
2853 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2855 GstRTSPSrc *src = stream->parent;
2858 g_object_get (source, "ssrc", &ssrc, NULL);
2860 GST_DEBUG_OBJECT (src, "source %08x, stream %08x, session %u received BYE",
2861 ssrc, stream->ssrc, stream->id);
2863 if (ssrc == stream->ssrc)
2864 gst_rtspsrc_do_stream_eos (src, stream);
2868 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2870 GstRTSPSrc *src = stream->parent;
2873 g_object_get (source, "ssrc", &ssrc, NULL);
2875 GST_WARNING_OBJECT (src, "source %08x, stream %08x in session %u timed out",
2876 ssrc, stream->ssrc, stream->id);
2878 if (ssrc == stream->ssrc)
2879 gst_rtspsrc_do_stream_eos (src, stream);
2883 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
2885 GstRTSPStream *stream;
2887 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
2889 /* get stream for session */
2890 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2892 gst_rtspsrc_do_stream_eos (src, stream);
2897 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
2899 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
2904 set_manager_buffer_mode (GstRTSPSrc * src)
2906 GObjectClass *klass;
2908 if (src->manager == NULL)
2911 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
2913 if (!g_object_class_find_property (klass, "buffer-mode"))
2916 if (src->buffer_mode != BUFFER_MODE_AUTO) {
2917 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
2922 GST_DEBUG_OBJECT (src,
2923 "auto buffering mode, have clock %" GST_PTR_FORMAT, src->provided_clock);
2925 if (src->provided_clock) {
2926 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
2928 if (clock == src->provided_clock) {
2929 GST_DEBUG_OBJECT (src, "selected synced");
2930 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SYNCED, NULL);
2933 gst_object_unref (clock);
2938 /* Otherwise fall-through and use another buffer mode */
2940 gst_object_unref (clock);
2943 GST_DEBUG_OBJECT (src, "auto buffering mode");
2944 if (src->use_buffering) {
2945 GST_DEBUG_OBJECT (src, "selected buffer");
2946 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER, NULL);
2948 GST_DEBUG_OBJECT (src, "selected slave");
2949 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
2954 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
2956 GST_DEBUG ("request key %u", ssrc);
2957 return gst_caps_ref (stream_get_caps_for_pt (stream, stream->default_pt));
2961 request_rtp_decoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
2963 GST_DEBUG ("decoder session %u, stream %p, %d", session, stream, stream->id);
2964 if (stream->id != session)
2967 if (stream->profile != GST_RTSP_PROFILE_SAVP &&
2968 stream->profile != GST_RTSP_PROFILE_SAVPF)
2971 if (stream->srtpdec == NULL) {
2974 name = g_strdup_printf ("srtpdec_%u", session);
2975 stream->srtpdec = gst_element_factory_make ("srtpdec", name);
2978 g_signal_connect (stream->srtpdec, "request-key",
2979 (GCallback) request_key, stream);
2981 return gst_object_ref (stream->srtpdec);
2985 request_rtcp_encoder (GstElement * rtpbin, guint session,
2986 GstRTSPStream * stream)
2991 GST_DEBUG ("decoder session %u, stream %p, %d", session, stream, stream->id);
2992 if (stream->id != session)
2995 if (stream->profile != GST_RTSP_PROFILE_SAVP &&
2996 stream->profile != GST_RTSP_PROFILE_SAVPF)
2999 if (stream->srtpenc == NULL) {
3000 name = g_strdup_printf ("srtpenc_%u", session);
3001 stream->srtpenc = gst_element_factory_make ("srtpenc", name);
3004 /* key has been made before */
3005 g_object_set (stream->srtpenc, "key", stream->key, NULL);
3007 name = g_strdup_printf ("rtcp_sink_%d", session);
3008 pad = gst_element_get_request_pad (stream->srtpenc, name);
3010 gst_object_unref (pad);
3012 return gst_object_ref (stream->srtpenc);
3016 /* try to get and configure a manager */
3018 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
3019 GstRTSPTransport * transport)
3021 const gchar *manager;
3023 GstStateChangeReturn ret;
3025 /* find a manager */
3026 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
3030 GST_DEBUG_OBJECT (src, "using manager %s", manager);
3032 /* configure the manager */
3033 if (src->manager == NULL) {
3034 GObjectClass *klass;
3036 if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
3038 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
3042 goto use_no_manager;
3044 if (!(src->manager = gst_element_factory_make (manager, "manager")))
3045 goto manager_failed;
3048 /* we manage this element */
3049 gst_element_set_locked_state (src->manager, TRUE);
3050 gst_bin_add (GST_BIN_CAST (src), src->manager);
3052 ret = gst_element_set_state (src->manager, GST_STATE_PAUSED);
3053 if (ret == GST_STATE_CHANGE_FAILURE)
3054 goto start_manager_failure;
3056 g_object_set (src->manager, "latency", src->latency, NULL);
3058 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
3060 if (g_object_class_find_property (klass, "ntp-sync")) {
3061 g_object_set (src->manager, "ntp-sync", src->ntp_sync, NULL);
3064 if (g_object_class_find_property (klass, "use-pipeline-clock")) {
3065 g_object_set (src->manager, "use-pipeline-clock",
3066 src->use_pipeline_clock, NULL);
3069 if (src->sdes && g_object_class_find_property (klass, "sdes")) {
3070 g_object_set (src->manager, "sdes", src->sdes, NULL);
3073 if (g_object_class_find_property (klass, "drop-on-latency")) {
3074 g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
3078 /* buffer mode pauses are handled by adding offsets to buffer times,
3079 * but some depayloaders may have a hard time syncing output times
3080 * with such input times, e.g. container ones, most notably ASF */
3081 /* TODO alternatives are having an event that indicates these shifts,
3082 * or having rtsp extensions provide suggestion on buffer mode */
3083 /* valid duration implies not likely live pipeline,
3084 * so slaving in jitterbuffer does not make much sense
3085 * (and might mess things up due to bursts) */
3086 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
3087 src->segment.duration && !stream->container) {
3088 src->use_buffering = TRUE;
3090 src->use_buffering = FALSE;
3093 set_manager_buffer_mode (src);
3095 /* connect to signals */
3096 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
3098 src->manager_sig_id =
3099 g_signal_connect (src->manager, "pad-added",
3100 (GCallback) new_manager_pad, src);
3101 src->manager_ptmap_id =
3102 g_signal_connect (src->manager, "request-pt-map",
3103 (GCallback) request_pt_map, src);
3105 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
3108 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_NEW_MANAGER], 0,
3111 g_signal_connect (src->manager, "request-rtp-decoder",
3112 (GCallback) request_rtp_decoder, stream);
3113 g_signal_connect (src->manager, "request-rtcp-decoder",
3114 (GCallback) request_rtp_decoder, stream);
3115 g_signal_connect (src->manager, "request-rtcp-encoder",
3116 (GCallback) request_rtcp_encoder, stream);
3118 /* we stream directly to the manager, get some pads. Each RTSP stream goes
3119 * into a separate RTP session. */
3120 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
3121 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
3123 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
3124 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
3127 /* now configure the bandwidth in the manager */
3128 if (g_signal_lookup ("get-internal-session",
3129 G_OBJECT_TYPE (src->manager)) != 0) {
3130 GObject *rtpsession;
3132 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
3135 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
3137 stream->session = rtpsession;
3139 if (stream->as_bandwidth != -1) {
3140 GST_INFO_OBJECT (src, "setting AS: %f",
3141 (gdouble) (stream->as_bandwidth * 1000));
3142 g_object_set (rtpsession, "bandwidth",
3143 (gdouble) (stream->as_bandwidth * 1000), NULL);
3145 if (stream->rr_bandwidth != -1) {
3146 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
3147 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
3150 if (stream->rs_bandwidth != -1) {
3151 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
3152 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
3156 g_object_set (rtpsession, "probation", src->probation, NULL);
3158 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
3160 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
3162 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
3164 g_signal_connect (rtpsession, "on-ssrc-active",
3165 (GCallback) on_ssrc_active, stream);
3176 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3181 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
3184 start_manager_failure:
3186 GST_DEBUG_OBJECT (src, "could not start session manager");
3191 /* free the UDP sources allocated when negotiating a transport.
3192 * This function is called when the server negotiated to a transport where the
3193 * UDP sources are not needed anymore, such as TCP or multicast. */
3195 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
3199 for (i = 0; i < 2; i++) {
3200 if (stream->udpsrc[i]) {
3201 GST_DEBUG ("free UDP source %d for stream %p", i, stream);
3202 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
3203 gst_object_unref (stream->udpsrc[i]);
3204 stream->udpsrc[i] = NULL;
3209 /* for TCP, create pads to send and receive data to and from the manager and to
3210 * intercept various events and queries
3213 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
3214 GstRTSPTransport * transport, GstPad ** outpad)
3217 GstPadTemplate *template;
3218 GstPad *pad0, *pad1;
3220 /* configure for interleaved delivery, nothing needs to be done
3221 * here, the loop function will call the chain functions of the
3222 * session manager. */
3223 stream->channel[0] = transport->interleaved.min;
3224 stream->channel[1] = transport->interleaved.max;
3225 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
3226 stream->channel[0], stream->channel[1]);
3228 /* we can remove the allocated UDP ports now */
3229 gst_rtspsrc_stream_free_udp (stream);
3231 /* no session manager, send data to srcpad directly */
3232 if (!stream->channelpad[0]) {
3233 GST_DEBUG_OBJECT (src, "no manager, creating pad");
3235 /* create a new pad we will use to stream to */
3236 name = g_strdup_printf ("stream_%u", stream->id);
3237 template = gst_static_pad_template_get (&rtptemplate);
3238 stream->channelpad[0] = gst_pad_new_from_template (template, name);
3239 gst_object_unref (template);
3242 /* set caps and activate */
3243 gst_pad_use_fixed_caps (stream->channelpad[0]);
3244 gst_pad_set_active (stream->channelpad[0], TRUE);
3246 *outpad = gst_object_ref (stream->channelpad[0]);
3248 GST_DEBUG_OBJECT (src, "using manager source pad");
3250 template = gst_static_pad_template_get (&anysrctemplate);
3252 /* allocate pads for sending the channel data into the manager */
3253 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
3254 gst_pad_link_full (pad0, stream->channelpad[0], GST_PAD_LINK_CHECK_NOTHING);
3255 gst_object_unref (stream->channelpad[0]);
3256 stream->channelpad[0] = pad0;
3257 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
3258 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
3259 gst_pad_set_element_private (pad0, src);
3260 gst_pad_set_active (pad0, TRUE);
3262 if (stream->channelpad[1]) {
3263 /* if we have a sinkpad for the other channel, create a pad and link to the
3265 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
3266 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
3267 gst_pad_link_full (pad1, stream->channelpad[1],
3268 GST_PAD_LINK_CHECK_NOTHING);
3269 gst_object_unref (stream->channelpad[1]);
3270 stream->channelpad[1] = pad1;
3271 gst_pad_set_active (pad1, TRUE);
3273 gst_object_unref (template);
3275 /* setup RTCP transport back to the server if we have to. */
3276 if (src->manager && src->do_rtcp) {
3279 template = gst_static_pad_template_get (&anysinktemplate);
3281 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
3282 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
3283 gst_pad_set_element_private (stream->rtcppad, stream);
3284 gst_pad_set_active (stream->rtcppad, TRUE);
3286 /* get session RTCP pad */
3287 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3288 pad = gst_element_get_request_pad (src->manager, name);
3293 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3294 gst_object_unref (pad);
3297 gst_object_unref (template);
3303 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
3304 GstRTSPTransport * transport, const gchar ** destination, gint * min,
3305 gint * max, guint * ttl)
3307 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3309 if (!(*destination = transport->destination))
3310 *destination = stream->destination;
3313 /* transport first */
3314 *min = transport->port.min;
3315 *max = transport->port.max;
3316 if (*min == -1 && *max == -1) {
3317 /* then try from SDP */
3318 if (stream->port != 0) {
3319 *min = stream->port;
3320 *max = stream->port + 1;
3326 if (!(*ttl = transport->ttl))
3331 /* first take the source, then the endpoint to figure out where to send
3333 if (!(*destination = transport->source)) {
3334 if (src->conninfo.connection)
3335 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
3336 else if (stream->conninfo.connection)
3338 gst_rtsp_connection_get_ip (stream->conninfo.connection);
3342 /* for unicast we only expect the ports here */
3343 *min = transport->server_port.min;
3344 *max = transport->server_port.max;
3349 /* For multicast create UDP sources and join the multicast group. */
3351 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
3352 GstRTSPTransport * transport, GstPad ** outpad)
3355 const gchar *destination;
3358 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
3360 /* we can remove the allocated UDP ports now */
3361 gst_rtspsrc_stream_free_udp (stream);
3363 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
3366 /* we need a destination now */
3367 if (destination == NULL)
3368 goto no_destination;
3370 /* we really need ports now or we won't be able to receive anything at all */
3371 if (min == -1 && max == -1)
3374 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
3375 destination, min, max);
3377 /* creating UDP source for RTP */
3379 uri = g_strdup_printf ("udp://%s:%d", destination, min);
3381 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3383 if (stream->udpsrc[0] == NULL)
3386 /* take ownership */
3387 gst_object_ref_sink (stream->udpsrc[0]);
3389 if (src->udp_buffer_size != 0)
3390 g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
3391 src->udp_buffer_size, NULL);
3393 if (src->multi_iface != NULL)
3394 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3395 src->multi_iface, NULL);
3398 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3399 gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
3402 /* creating another UDP source for RTCP */
3406 uri = g_strdup_printf ("udp://%s:%d", destination, max);
3408 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3410 if (stream->udpsrc[1] == NULL)
3413 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3414 stream->profile == GST_RTSP_PROFILE_SAVPF)
3415 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3417 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3418 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3419 gst_caps_unref (caps);
3421 /* take ownership */
3422 gst_object_ref_sink (stream->udpsrc[1]);
3424 if (src->multi_iface != NULL)
3425 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3426 src->multi_iface, NULL);
3428 gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
3435 GST_DEBUG_OBJECT (src, "no UDP source element found");
3440 GST_DEBUG_OBJECT (src, "no destination found");
3445 GST_DEBUG_OBJECT (src, "no ports found");
3450 /* configure the remainder of the UDP ports */
3452 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
3453 GstRTSPTransport * transport, GstPad ** outpad)
3455 /* we manage the UDP elements now. For unicast, the UDP sources where
3456 * allocated in the stream when we suggested a transport. */
3457 if (stream->udpsrc[0]) {
3460 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3461 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
3463 GST_DEBUG_OBJECT (src, "setting up UDP source");
3465 /* configure a timeout on the UDP port. When the timeout message is
3466 * posted, we assume UDP transport is not possible. We reconnect using TCP
3468 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
3469 src->udp_timeout * 1000, NULL);
3471 if ((caps = stream_get_caps_for_pt (stream, stream->default_pt)))
3472 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
3474 /* get output pad of the UDP source. */
3475 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
3477 /* save it so we can unblock */
3478 stream->blockedpad = *outpad;
3480 /* configure pad block on the pad. As soon as there is dataflow on the
3481 * UDP source, we know that UDP is not blocked by a firewall and we can
3482 * configure all the streams to let the application autoplug decoders. */
3484 gst_pad_add_probe (stream->blockedpad,
3485 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3486 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocked, src, NULL);
3488 if (stream->channelpad[0]) {
3489 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
3490 /* configure for UDP delivery, we need to connect the UDP pads to
3491 * the session plugin. */
3492 gst_pad_link_full (*outpad, stream->channelpad[0],
3493 GST_PAD_LINK_CHECK_NOTHING);
3494 gst_object_unref (*outpad);
3496 /* we connected to pad-added signal to get pads from the manager */
3498 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
3503 if (stream->udpsrc[1]) {
3506 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
3507 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
3509 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3510 stream->profile == GST_RTSP_PROFILE_SAVPF)
3511 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3513 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3514 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3515 gst_caps_unref (caps);
3517 if (stream->channelpad[1]) {
3520 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
3522 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
3523 gst_pad_link_full (pad, stream->channelpad[1],
3524 GST_PAD_LINK_CHECK_NOTHING);
3525 gst_object_unref (pad);
3527 /* leave unlinked */
3533 /* configure the UDP sink back to the server for status reports */
3535 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
3536 GstRTSPStream * stream, GstRTSPTransport * transport)
3539 gint rtp_port, rtcp_port;
3540 gboolean do_rtp, do_rtcp;
3541 const gchar *destination;
3546 /* get transport info */
3547 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
3548 &rtp_port, &rtcp_port, &ttl);
3550 /* see what we need to do */
3551 do_rtp = (rtp_port != -1);
3552 /* it's possible that the server does not want us to send RTCP in which case
3554 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
3556 /* we need a destination when we have RTP or RTCP ports */
3557 if (destination == NULL && (do_rtp || do_rtcp))
3558 goto no_destination;
3560 /* try to construct the fakesrc to the RTP port of the server to open up any
3563 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
3566 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
3567 stream->udpsink[0] =
3568 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3570 if (stream->udpsink[0] == NULL)
3571 goto no_sink_element;
3573 /* don't join multicast group, we will have the source socket do that */
3574 /* no sync or async state changes needed */
3575 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
3576 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3578 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3580 if (stream->udpsrc[0]) {
3581 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
3582 * so that NAT firewalls will open a hole for us */
3583 g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
3584 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
3585 /* configure socket and make sure udpsink does not close it when shutting
3586 * down, it belongs to udpsrc after all. */
3587 g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
3588 "close-socket", FALSE, NULL);
3589 g_object_unref (socket);
3592 /* the source for the dummy packets to open up NAT */
3593 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
3594 if (stream->fakesrc == NULL)
3595 goto no_fakesrc_element;
3597 /* random data in 5 buffers, a size of 200 bytes should be fine */
3598 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
3599 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
3601 /* we don't want to consider this a sink */
3602 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
3604 /* keep everything locked */
3605 gst_element_set_locked_state (stream->udpsink[0], TRUE);
3606 gst_element_set_locked_state (stream->fakesrc, TRUE);
3608 gst_object_ref (stream->udpsink[0]);
3609 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
3610 gst_object_ref (stream->fakesrc);
3611 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
3613 gst_element_link_pads_full (stream->fakesrc, "src", stream->udpsink[0],
3614 "sink", GST_PAD_LINK_CHECK_NOTHING);
3617 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
3620 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
3621 stream->udpsink[1] =
3622 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3624 if (stream->udpsink[1] == NULL)
3625 goto no_sink_element;
3627 /* don't join multicast group, we will have the source socket do that */
3628 /* no sync or async state changes needed */
3629 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
3630 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3632 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3634 if (stream->udpsrc[1]) {
3635 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
3636 * because some servers check the port number of where it sends RTCP to identify
3637 * the RTCP packets it receives */
3638 g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
3639 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
3640 /* configure socket and make sure udpsink does not close it when shutting
3641 * down, it belongs to udpsrc after all. */
3642 g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
3643 "close-socket", FALSE, NULL);
3644 g_object_unref (socket);
3647 /* we don't want to consider this a sink */
3648 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
3650 /* we keep this playing always */
3651 gst_element_set_locked_state (stream->udpsink[1], TRUE);
3652 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
3654 gst_object_ref (stream->udpsink[1]);
3655 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
3657 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
3659 /* get session RTCP pad */
3660 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3661 pad = gst_element_get_request_pad (src->manager, name);
3666 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3667 gst_object_unref (pad);
3676 GST_DEBUG_OBJECT (src, "no destination address specified");
3681 GST_DEBUG_OBJECT (src, "no UDP sink element found");
3686 GST_DEBUG_OBJECT (src, "no fakesrc element found");
3691 /* sets up all elements needed for streaming over the specified transport.
3692 * Does not yet expose the element pads, this will be done when there is actuall
3693 * dataflow detected, which might never happen when UDP is blocked in a
3694 * firewall, for example.
3697 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
3698 GstRTSPTransport * transport)
3701 GstPad *outpad = NULL;
3702 GstPadTemplate *template;
3704 const gchar *media_type;
3707 src = stream->parent;
3709 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
3711 /* get the proper media type for this stream now */
3712 if (gst_rtsp_transport_get_media_type (transport, &media_type) < 0)
3713 goto unknown_transport;
3715 goto unknown_transport;
3717 /* configure the final media type */
3718 GST_DEBUG_OBJECT (src, "setting media type to %s", media_type);
3720 len = stream->ptmap->len;
3721 for (i = 0; i < len; i++) {
3723 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
3725 if (item->caps == NULL)
3728 s = gst_caps_get_structure (item->caps, 0);
3729 gst_structure_set_name (s, media_type);
3730 /* set ssrc if known */
3731 if (transport->ssrc)
3732 gst_structure_set (s, "ssrc", G_TYPE_UINT, transport->ssrc, NULL);
3735 /* try to get and configure a manager, channelpad[0-1] will be configured with
3736 * the pads for the manager, or NULL when no manager is needed. */
3737 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
3740 switch (transport->lower_transport) {
3741 case GST_RTSP_LOWER_TRANS_TCP:
3742 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
3743 goto transport_failed;
3745 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3746 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
3747 goto transport_failed;
3748 /* fallthrough, the rest is the same for UDP and MCAST */
3749 case GST_RTSP_LOWER_TRANS_UDP:
3750 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
3751 goto transport_failed;
3752 /* configure udpsinks back to the server for RTCP messages and for the
3753 * dummy RTP messages to open NAT. */
3754 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
3755 goto transport_failed;
3758 goto unknown_transport;
3762 GST_DEBUG_OBJECT (src, "creating ghostpad");
3764 gst_pad_use_fixed_caps (outpad);
3766 /* create ghostpad, don't add just yet, this will be done when we activate
3768 name = g_strdup_printf ("stream_%u", stream->id);
3769 template = gst_static_pad_template_get (&rtptemplate);
3770 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
3771 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
3772 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
3773 gst_object_unref (template);
3776 gst_object_unref (outpad);
3778 /* mark pad as ok */
3779 stream->last_ret = GST_FLOW_OK;
3786 GST_DEBUG_OBJECT (src, "failed to configure transport");
3791 GST_DEBUG_OBJECT (src, "unknown transport");
3796 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3801 /* send a couple of dummy random packets on the receiver RTP port to the server,
3802 * this should make a firewall think we initiated the data transfer and
3803 * hopefully allow packets to go from the sender port to our RTP receiver port */
3805 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
3809 if (src->nat_method != GST_RTSP_NAT_DUMMY)
3812 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3813 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3815 if (stream->fakesrc && stream->udpsink[0]) {
3816 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
3817 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
3818 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
3819 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
3820 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
3826 /* Adds the source pads of all configured streams to the element.
3827 * This code is performed when we detected dataflow.
3829 * We detect dataflow from either the _loop function or with pad probes on the
3833 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
3837 GST_DEBUG_OBJECT (src, "activating streams");
3839 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3840 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3842 if (stream->udpsrc[0]) {
3843 /* remove timeout, we are streaming now and timeouts will be handled by
3844 * the session manager and jitter buffer */
3845 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
3847 if (stream->srcpad) {
3848 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
3849 gst_pad_set_active (stream->srcpad, TRUE);
3851 /* if we don't have a session manager, set the caps now. If we have a
3852 * session, we will get a notification of the pad and the caps. */
3853 if (!src->manager) {
3856 caps = stream_get_caps_for_pt (stream, stream->default_pt);
3857 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
3858 gst_pad_set_caps (stream->srcpad, caps);
3861 if (!stream->added) {
3862 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
3863 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
3864 stream->added = TRUE;
3869 /* unblock all pads */
3870 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3871 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3873 if (stream->blockid) {
3874 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
3875 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
3876 stream->blockid = 0;
3884 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
3885 gboolean reset_manager)
3888 guint64 start, stop;
3889 gdouble play_speed, play_scale;
3891 GST_DEBUG_OBJECT (src, "configuring stream caps");
3893 start = segment->position;
3894 stop = segment->duration;
3895 play_speed = segment->rate;
3896 play_scale = segment->applied_rate;
3898 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3899 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3905 len = stream->ptmap->len;
3906 for (j = 0; j < len; j++) {
3908 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, j);
3910 if (item->caps == NULL)
3913 caps = gst_caps_make_writable (item->caps);
3915 if (stream->timebase != -1)
3916 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
3917 (guint) stream->timebase, NULL);
3918 if (stream->seqbase != -1)
3919 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
3920 (guint) stream->seqbase, NULL);
3921 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
3923 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
3924 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
3925 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
3928 GST_DEBUG_OBJECT (src, "stream %p, pt %d, caps %" GST_PTR_FORMAT, stream,
3931 if (item->pt == stream->default_pt && stream->udpsrc[0]) {
3932 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
3936 if (reset_manager && src->manager) {
3937 GST_DEBUG_OBJECT (src, "clear session");
3938 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
3942 static GstFlowReturn
3943 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
3948 /* store the value */
3949 stream->last_ret = ret;
3951 /* if it's success we can return the value right away */
3952 if (ret == GST_FLOW_OK)
3955 /* any other error that is not-linked can be returned right
3957 if (ret != GST_FLOW_NOT_LINKED)
3960 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
3961 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3962 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3964 ret = ostream->last_ret;
3965 /* some other return value (must be SUCCESS but we can return
3966 * other values as well) */
3967 if (ret != GST_FLOW_NOT_LINKED)
3970 /* if we get here, all other pads were unlinked and we return
3971 * NOT_LINKED then */
3977 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
3980 gboolean res = TRUE;
3982 /* only streams that have a connection to the outside world */
3986 if (stream->udpsrc[0]) {
3987 gst_event_ref (event);
3988 res = gst_element_send_event (stream->udpsrc[0], event);
3989 } else if (stream->channelpad[0]) {
3990 gst_event_ref (event);
3991 if (GST_PAD_IS_SRC (stream->channelpad[0]))
3992 res = gst_pad_push_event (stream->channelpad[0], event);
3994 res = gst_pad_send_event (stream->channelpad[0], event);
3997 if (stream->udpsrc[1]) {
3998 gst_event_ref (event);
3999 res &= gst_element_send_event (stream->udpsrc[1], event);
4000 } else if (stream->channelpad[1]) {
4001 gst_event_ref (event);
4002 if (GST_PAD_IS_SRC (stream->channelpad[1]))
4003 res &= gst_pad_push_event (stream->channelpad[1], event);
4005 res &= gst_pad_send_event (stream->channelpad[1], event);
4009 gst_event_unref (event);
4015 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
4018 gboolean res = TRUE;
4020 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4021 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4023 gst_event_ref (event);
4024 res &= gst_rtspsrc_stream_push_event (src, ostream, event);
4026 gst_event_unref (event);
4031 static GstRTSPResult
4032 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
4037 if (info->connection == NULL) {
4038 if (info->url == NULL) {
4039 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
4040 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
4044 /* create connection */
4045 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
4046 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
4047 goto could_not_create;
4050 g_free (info->url_str);
4051 info->url_str = gst_rtsp_url_get_request_uri (info->url);
4053 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
4055 if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
4056 if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
4057 src->tls_validation_flags))
4058 GST_WARNING_OBJECT (src, "Unable to set TLS validation flags");
4060 if (src->tls_database)
4061 gst_rtsp_connection_set_tls_database (info->connection,
4065 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
4066 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
4068 if (src->proxy_host) {
4069 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
4071 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
4076 if (!info->connected) {
4079 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
4080 ("Connecting to %s", info->location));
4081 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
4083 gst_rtsp_connection_connect (info->connection,
4084 src->ptcp_timeout)) < 0)
4085 goto could_not_connect;
4087 info->connected = TRUE;
4094 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
4099 gchar *str = gst_rtsp_strresult (res);
4100 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
4106 gchar *str = gst_rtsp_strresult (res);
4107 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
4113 static GstRTSPResult
4114 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
4117 GST_RTSP_STATE_LOCK (src);
4118 if (info->connected) {
4119 GST_DEBUG_OBJECT (src, "closing connection...");
4120 gst_rtsp_connection_close (info->connection);
4121 info->connected = FALSE;
4123 if (free && info->connection) {
4124 /* free connection */
4125 GST_DEBUG_OBJECT (src, "freeing connection...");
4126 gst_rtsp_connection_free (info->connection);
4127 info->connection = NULL;
4129 GST_RTSP_STATE_UNLOCK (src);
4133 static GstRTSPResult
4134 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
4139 GST_DEBUG_OBJECT (src, "reconnecting connection...");
4140 gst_rtsp_conninfo_close (src, info, FALSE);
4141 res = gst_rtsp_conninfo_connect (src, info, async);
4147 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
4151 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
4152 GST_RTSP_STATE_LOCK (src);
4153 if (src->conninfo.connection && src->conninfo.flushing != flush) {
4154 GST_DEBUG_OBJECT (src, "connection flush");
4155 gst_rtsp_connection_flush (src->conninfo.connection, flush);
4156 src->conninfo.flushing = flush;
4158 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4159 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4160 if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
4161 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
4162 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
4163 stream->conninfo.flushing = flush;
4166 GST_RTSP_STATE_UNLOCK (src);
4169 /* FIXME, handle server request, reply with OK, for now */
4170 static GstRTSPResult
4171 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
4172 GstRTSPMessage * request)
4174 GstRTSPMessage response = { 0 };
4177 GST_DEBUG_OBJECT (src, "got server request message");
4180 gst_rtsp_message_dump (request);
4182 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
4184 if (res == GST_RTSP_ENOTIMPL) {
4185 /* default implementation, send OK */
4186 GST_DEBUG_OBJECT (src, "prepare OK reply");
4188 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
4193 /* let app parse and reply */
4194 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST],
4195 0, request, &response);
4198 gst_rtsp_message_dump (&response);
4200 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
4204 gst_rtsp_message_unset (&response);
4205 } else if (res == GST_RTSP_EEOF)
4213 gst_rtsp_message_unset (&response);
4218 /* send server keep-alive */
4219 static GstRTSPResult
4220 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
4222 GstRTSPMessage request = { 0 };
4224 GstRTSPMethod method;
4225 const gchar *control;
4227 if (src->do_rtsp_keep_alive == FALSE) {
4228 GST_DEBUG_OBJECT (src, "do-rtsp-keep-alive is FALSE, not sending.");
4229 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4233 GST_DEBUG_OBJECT (src, "creating server keep-alive");
4235 /* find a method to use for keep-alive */
4236 if (src->methods & GST_RTSP_GET_PARAMETER)
4237 method = GST_RTSP_GET_PARAMETER;
4239 method = GST_RTSP_OPTIONS;
4241 control = get_aggregate_control (src);
4242 if (control == NULL)
4245 res = gst_rtsp_message_init_request (&request, method, control);
4250 gst_rtsp_message_dump (&request);
4253 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
4258 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4259 gst_rtsp_message_unset (&request);
4266 GST_WARNING_OBJECT (src, "no control url to send keepalive");
4271 gchar *str = gst_rtsp_strresult (res);
4273 gst_rtsp_message_unset (&request);
4274 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
4275 ("Could not send keep-alive. (%s)", str));
4281 static GstFlowReturn
4282 gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
4284 GstFlowReturn ret = GST_FLOW_OK;
4286 GstRTSPStream *stream;
4287 GstPad *outpad = NULL;
4294 channel = message->type_data.data.channel;
4296 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
4298 goto unknown_stream;
4300 if (channel == stream->channel[0]) {
4301 outpad = stream->channelpad[0];
4303 } else if (channel == stream->channel[1]) {
4304 outpad = stream->channelpad[1];
4310 /* take a look at the body to figure out what we have */
4311 gst_rtsp_message_get_body (message, &data, &size);
4313 goto invalid_length;
4315 /* channels are not correct on some servers, do extra check */
4316 if (data[1] >= 200 && data[1] <= 204) {
4317 /* hmm RTCP message switch to the RTCP pad of the same stream. */
4318 outpad = stream->channelpad[1];
4322 /* we have no clue what this is, just ignore then. */
4324 goto unknown_stream;
4326 /* take the message body for further processing */
4327 gst_rtsp_message_steal_body (message, &data, &size);
4329 /* strip the trailing \0 */
4332 buf = gst_buffer_new ();
4333 gst_buffer_append_memory (buf,
4334 gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
4336 /* don't need message anymore */
4337 gst_rtsp_message_unset (message);
4339 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
4342 if (src->need_activate) {
4348 guint group_id = gst_util_group_id_next ();
4350 /* generate an SHA256 sum of the URI */
4351 cs = g_checksum_new (G_CHECKSUM_SHA256);
4352 uri = src->conninfo.location;
4353 g_checksum_update (cs, (const guchar *) uri, strlen (uri));
4355 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4356 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4359 g_strdup_printf ("%s/%d", g_checksum_get_string (cs), ostream->id);
4360 event = gst_event_new_stream_start (stream_id);
4361 gst_event_set_group_id (event, group_id);
4364 gst_rtspsrc_stream_push_event (src, ostream, event);
4366 g_checksum_free (cs);
4368 gst_rtspsrc_activate_streams (src);
4369 src->need_activate = FALSE;
4371 if ((event = src->start_segment) != NULL) {
4372 src->start_segment = NULL;
4373 gst_rtspsrc_push_event (src, event);
4376 if (src->base_time == -1) {
4377 /* Take current running_time. This timestamp will be put on
4378 * the first buffer of each stream because we are a live source and so we
4379 * timestamp with the running_time. When we are dealing with TCP, we also
4380 * only timestamp the first buffer (using the DISCONT flag) because a server
4381 * typically bursts data, for which we don't want to compensate by speeding
4382 * up the media. The other timestamps will be interpollated from this one
4383 * using the RTP timestamps. */
4384 GST_OBJECT_LOCK (src);
4385 if (GST_ELEMENT_CLOCK (src)) {
4387 GstClockTime base_time;
4389 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
4390 base_time = GST_ELEMENT_CAST (src)->base_time;
4392 src->base_time = now - base_time;
4394 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
4395 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
4397 GST_OBJECT_UNLOCK (src);
4400 if (stream->discont && !is_rtcp) {
4401 /* mark first RTP buffer as discont */
4402 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
4403 stream->discont = FALSE;
4404 /* first buffer gets the timestamp, other buffers are not timestamped and
4405 * their presentation time will be interpollated from the rtp timestamps. */
4406 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
4407 GST_TIME_ARGS (src->base_time));
4409 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
4412 /* chain to the peer pad */
4413 if (GST_PAD_IS_SINK (outpad))
4414 ret = gst_pad_chain (outpad, buf);
4416 ret = gst_pad_push (outpad, buf);
4419 /* combine all stream flows for the data transport */
4420 ret = gst_rtspsrc_combine_flows (src, stream, ret);
4427 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
4428 gst_rtsp_message_unset (message);
4433 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4434 ("Short message received, ignoring."));
4435 gst_rtsp_message_unset (message);
4440 static GstFlowReturn
4441 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
4443 GstRTSPMessage message = { 0 };
4445 GstFlowReturn ret = GST_FLOW_OK;
4446 GTimeVal tv_timeout;
4449 /* get the next timeout interval */
4450 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4452 /* see if the timeout period expired */
4453 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
4454 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
4455 /* send keep-alive, only act on interrupt, a warning will be posted for
4457 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4459 /* get new timeout */
4460 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4463 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
4464 tv_timeout.tv_sec, tv_timeout.tv_usec);
4466 /* protect the connection with the connection lock so that we can see when
4467 * we are finished doing server communication */
4469 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4470 &message, src->ptcp_timeout);
4474 GST_DEBUG_OBJECT (src, "we received a server message");
4476 case GST_RTSP_EINTR:
4477 /* we got interrupted this means we need to stop */
4479 case GST_RTSP_ETIMEOUT:
4480 /* no reply, send keep alive */
4481 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4482 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4486 /* go EOS when the server closed the connection */
4492 switch (message.type) {
4493 case GST_RTSP_MESSAGE_REQUEST:
4494 /* server sends us a request message, handle it */
4496 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4498 if (res == GST_RTSP_EEOF)
4501 goto handle_request_failed;
4503 case GST_RTSP_MESSAGE_RESPONSE:
4504 /* we ignore response messages */
4505 GST_DEBUG_OBJECT (src, "ignoring response message");
4507 gst_rtsp_message_dump (&message);
4509 case GST_RTSP_MESSAGE_DATA:
4510 GST_DEBUG_OBJECT (src, "got data message");
4511 ret = gst_rtspsrc_handle_data (src, &message);
4512 if (ret != GST_FLOW_OK)
4513 goto handle_data_failed;
4516 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4521 g_assert_not_reached ();
4526 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4527 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4528 ("The server closed the connection."));
4529 src->conninfo.connected = FALSE;
4530 gst_rtsp_message_unset (&message);
4531 return GST_FLOW_EOS;
4535 gst_rtsp_message_unset (&message);
4536 GST_DEBUG_OBJECT (src, "got interrupted");
4537 return GST_FLOW_FLUSHING;
4541 gchar *str = gst_rtsp_strresult (res);
4543 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4544 ("Could not receive message. (%s)", str));
4547 gst_rtsp_message_unset (&message);
4548 return GST_FLOW_ERROR;
4550 handle_request_failed:
4552 gchar *str = gst_rtsp_strresult (res);
4554 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4555 ("Could not handle server message. (%s)", str));
4557 gst_rtsp_message_unset (&message);
4558 return GST_FLOW_ERROR;
4562 GST_DEBUG_OBJECT (src, "could no handle data message");
4567 static GstFlowReturn
4568 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
4571 GstRTSPMessage message = { 0 };
4575 GTimeVal tv_timeout;
4577 /* get the next timeout interval */
4578 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4580 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
4581 (gint) tv_timeout.tv_sec);
4583 gst_rtsp_message_unset (&message);
4585 /* we should continue reading the TCP socket because the server might
4586 * send us requests. When the session timeout expires, we need to send a
4587 * keep-alive request to keep the session open. */
4588 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4589 &message, &tv_timeout);
4593 GST_DEBUG_OBJECT (src, "we received a server message");
4595 case GST_RTSP_EINTR:
4596 /* we got interrupted, see what we have to do */
4598 case GST_RTSP_ETIMEOUT:
4599 /* send keep-alive, ignore the result, a warning will be posted. */
4600 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4601 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4605 /* server closed the connection. not very fatal for UDP, reconnect and
4606 * see what happens. */
4607 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4608 ("The server closed the connection."));
4609 if (src->udp_reconnect) {
4611 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
4618 GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
4620 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4621 ("Unhandled return value %d.", res));
4625 switch (message.type) {
4626 case GST_RTSP_MESSAGE_REQUEST:
4627 /* server sends us a request message, handle it */
4629 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4631 if (res == GST_RTSP_EEOF)
4634 goto handle_request_failed;
4636 case GST_RTSP_MESSAGE_RESPONSE:
4637 /* we ignore response and data messages */
4638 GST_DEBUG_OBJECT (src, "ignoring response message");
4640 gst_rtsp_message_dump (&message);
4641 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
4642 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
4643 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
4644 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
4645 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4652 case GST_RTSP_MESSAGE_DATA:
4653 /* we ignore response and data messages */
4654 GST_DEBUG_OBJECT (src, "ignoring data message");
4657 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4662 g_assert_not_reached ();
4664 /* we get here when the connection got interrupted */
4667 gst_rtsp_message_unset (&message);
4668 GST_DEBUG_OBJECT (src, "got interrupted");
4669 return GST_FLOW_FLUSHING;
4673 gchar *str = gst_rtsp_strresult (res);
4676 src->conninfo.connected = FALSE;
4677 if (res != GST_RTSP_EINTR) {
4678 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
4679 ("Could not connect to server. (%s)", str));
4681 ret = GST_FLOW_ERROR;
4683 ret = GST_FLOW_FLUSHING;
4689 gchar *str = gst_rtsp_strresult (res);
4691 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4692 ("Could not receive message. (%s)", str));
4694 return GST_FLOW_ERROR;
4696 handle_request_failed:
4698 gchar *str = gst_rtsp_strresult (res);
4701 gst_rtsp_message_unset (&message);
4702 if (res != GST_RTSP_EINTR) {
4703 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4704 ("Could not handle server message. (%s)", str));
4706 ret = GST_FLOW_ERROR;
4708 ret = GST_FLOW_FLUSHING;
4714 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4715 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4716 ("The server closed the connection."));
4717 src->conninfo.connected = FALSE;
4718 gst_rtsp_message_unset (&message);
4719 return GST_FLOW_EOS;
4723 static GstRTSPResult
4724 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
4726 GstRTSPResult res = GST_RTSP_OK;
4729 GST_DEBUG_OBJECT (src, "doing reconnect");
4731 GST_OBJECT_LOCK (src);
4732 /* only restart when the pads were not yet activated, else we were
4733 * streaming over UDP */
4734 restart = src->need_activate;
4735 GST_OBJECT_UNLOCK (src);
4737 /* no need to restart, we're done */
4741 /* we can try only TCP now */
4742 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
4744 /* close and cleanup our state */
4745 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
4748 /* see if we have TCP left to try. Also don't try TCP when we were configured
4750 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
4753 /* We post a warning message now to inform the user
4754 * that nothing happened. It's most likely a firewall thing. */
4755 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4756 ("Could not receive any UDP packets for %.4f seconds, maybe your "
4757 "firewall is blocking it. Retrying using a TCP connection.",
4758 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
4760 /* open new connection using tcp */
4761 if (gst_rtspsrc_open (src, async) < 0)
4764 /* start playback */
4765 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
4774 src->cur_protocols = 0;
4775 /* no transport possible, post an error and stop */
4776 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4777 ("Could not receive any UDP packets for %.4f seconds, maybe your "
4778 "firewall is blocking it. No other protocols to try.",
4779 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
4780 return GST_RTSP_ERROR;
4784 GST_DEBUG_OBJECT (src, "open failed");
4789 GST_DEBUG_OBJECT (src, "play failed");
4795 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
4799 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
4802 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
4805 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
4808 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
4816 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
4820 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
4823 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
4826 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
4829 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
4837 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
4841 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
4844 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
4847 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
4850 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
4858 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
4862 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
4865 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
4868 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
4871 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
4879 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
4881 if (ret == GST_RTSP_OK)
4882 gst_rtspsrc_loop_complete_cmd (src, cmd);
4883 else if (ret == GST_RTSP_EINTR)
4884 gst_rtspsrc_loop_cancel_cmd (src, cmd);
4886 gst_rtspsrc_loop_error_cmd (src, cmd);
4890 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
4893 gboolean flushed = FALSE;
4895 /* start new request */
4896 gst_rtspsrc_loop_start_cmd (src, cmd);
4898 GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
4900 GST_OBJECT_LOCK (src);
4901 old = src->pending_cmd;
4902 if (old == CMD_RECONNECT) {
4903 GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
4904 cmd = CMD_RECONNECT;
4906 if (old != CMD_WAIT) {
4907 src->pending_cmd = CMD_WAIT;
4908 GST_OBJECT_UNLOCK (src);
4909 /* cancel previous request */
4910 GST_DEBUG_OBJECT (src, "cancel previous request %d", old);
4911 gst_rtspsrc_loop_cancel_cmd (src, old);
4912 GST_OBJECT_LOCK (src);
4914 src->pending_cmd = cmd;
4915 /* interrupt if allowed */
4916 if (src->busy_cmd & mask) {
4917 GST_DEBUG_OBJECT (src, "connection flush busy %d", src->busy_cmd);
4918 gst_rtspsrc_connection_flush (src, TRUE);
4921 GST_DEBUG_OBJECT (src, "not interrupting busy cmd %d", src->busy_cmd);
4924 gst_task_start (src->task);
4925 GST_OBJECT_UNLOCK (src);
4931 gst_rtspsrc_loop (GstRTSPSrc * src)
4935 if (!src->conninfo.connection || !src->conninfo.connected)
4938 if (src->interleaved)
4939 ret = gst_rtspsrc_loop_interleaved (src);
4941 ret = gst_rtspsrc_loop_udp (src);
4943 if (ret != GST_FLOW_OK)
4951 GST_WARNING_OBJECT (src, "we are not connected");
4952 ret = GST_FLOW_FLUSHING;
4957 const gchar *reason = gst_flow_get_name (ret);
4959 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
4960 src->running = FALSE;
4961 if (ret == GST_FLOW_EOS) {
4962 /* perform EOS logic */
4963 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
4964 gst_element_post_message (GST_ELEMENT_CAST (src),
4965 gst_message_new_segment_done (GST_OBJECT_CAST (src),
4966 src->segment.format, src->segment.position));
4967 gst_rtspsrc_push_event (src,
4968 gst_event_new_segment_done (src->segment.format,
4969 src->segment.position));
4971 gst_rtspsrc_push_event (src, gst_event_new_eos ());
4973 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
4974 /* for fatal errors we post an error message, post the error before the
4975 * EOS so the app knows about the error first. */
4976 GST_ELEMENT_ERROR (src, STREAM, FAILED,
4977 ("Internal data flow error."),
4978 ("streaming task paused, reason %s (%d)", reason, ret));
4979 gst_rtspsrc_push_event (src, gst_event_new_eos ());
4981 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_LOOP);
4986 #ifndef GST_DISABLE_GST_DEBUG
4987 static const gchar *
4988 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
4992 while (method != 0) {
5009 static const gchar *
5010 gst_rtspsrc_skip_lws (const gchar * s)
5012 while (g_ascii_isspace (*s))
5017 static const gchar *
5018 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
5020 while (s > start && g_ascii_isspace (*(s - 1)))
5025 static const gchar *
5026 gst_rtspsrc_skip_commas (const gchar * s)
5028 /* The grammar allows for multiple commas */
5029 while (g_ascii_isspace (*s) || *s == ',')
5034 static const gchar *
5035 gst_rtspsrc_skip_item (const gchar * s)
5037 gboolean quoted = FALSE;
5038 const gchar *start = s;
5040 /* A list item ends at the last non-whitespace character
5041 * before a comma which is not inside a quoted-string. Or at
5042 * the end of the string.
5048 if (*s == '\\' && *(s + 1))
5057 return gst_rtspsrc_unskip_lws (s, start);
5061 gst_rtsp_decode_quoted_string (gchar * quoted_string)
5065 src = quoted_string + 1;
5066 dst = quoted_string;
5067 while (*src && *src != '"') {
5068 if (*src == '\\' && *(src + 1))
5075 /* Extract the authentication tokens that the server provided for each method
5076 * into an array of structures and give those to the connection object.
5079 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
5080 const gchar * header, gboolean * stale)
5082 GSList *list = NULL, *iter;
5084 gchar *item, *eq, *name_end, *value;
5086 g_return_if_fail (stale != NULL);
5088 gst_rtsp_connection_clear_auth_params (conn);
5091 /* Parse a header whose content is described by RFC2616 as
5092 * "#something", where "something" does not itself contain commas,
5093 * except as part of quoted-strings, into a list of allocated strings.
5095 header = gst_rtspsrc_skip_commas (header);
5097 end = gst_rtspsrc_skip_item (header);
5098 list = g_slist_prepend (list, g_strndup (header, end - header));
5099 header = gst_rtspsrc_skip_commas (end);
5104 list = g_slist_reverse (list);
5105 for (iter = list; iter; iter = iter->next) {
5108 eq = strchr (item, '=');
5110 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
5111 if (name_end == item) {
5112 /* That's no good... */
5119 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
5121 gst_rtsp_decode_quoted_string (value);
5125 if (item && (strcmp (item, "stale") == 0) &&
5126 value && (strcmp (value, "TRUE") == 0))
5128 gst_rtsp_connection_set_auth_param (conn, item, value);
5132 g_slist_free (list);
5135 /* Parse a WWW-Authenticate Response header and determine the
5136 * available authentication methods
5138 * This code should also cope with the fact that each WWW-Authenticate
5139 * header can contain multiple challenge methods + tokens
5141 * At the moment, for Basic auth, we just do a minimal check and don't
5142 * even parse out the realm */
5144 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
5145 GstRTSPConnection * conn, gboolean * stale)
5149 g_return_if_fail (hdr != NULL);
5150 g_return_if_fail (methods != NULL);
5151 g_return_if_fail (stale != NULL);
5153 /* Skip whitespace at the start of the string */
5154 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
5156 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
5157 *methods |= GST_RTSP_AUTH_BASIC;
5158 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
5159 *methods |= GST_RTSP_AUTH_DIGEST;
5160 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
5165 * gst_rtspsrc_setup_auth:
5166 * @src: the rtsp source
5168 * Configure a username and password and auth method on the
5169 * connection object based on a response we received from the
5172 * Currently, this requires that a username and password were supplied
5173 * in the uri. In the future, they may be requested on demand by sending
5174 * a message up the bus.
5176 * Returns: TRUE if authentication information could be set up correctly.
5179 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
5183 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
5184 GstRTSPAuthMethod method;
5185 GstRTSPResult auth_result;
5187 GstRTSPConnection *conn;
5189 gboolean stale = FALSE;
5191 conn = src->conninfo.connection;
5193 /* Identify the available auth methods and see if any are supported */
5194 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
5195 &hdr, 0) == GST_RTSP_OK) {
5196 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
5199 if (avail_methods == GST_RTSP_AUTH_NONE)
5200 goto no_auth_available;
5202 /* For digest auth, if the response indicates that the session
5203 * data are stale, we just update them in the connection object and
5204 * return TRUE to retry the request */
5206 src->tried_url_auth = FALSE;
5208 url = gst_rtsp_connection_get_url (conn);
5210 /* Do we have username and password available? */
5211 if (url != NULL && !src->tried_url_auth && url->user != NULL
5212 && url->passwd != NULL) {
5215 src->tried_url_auth = TRUE;
5216 GST_DEBUG_OBJECT (src,
5217 "Attempting authentication using credentials from the URL");
5219 user = src->user_id;
5220 pass = src->user_pw;
5221 GST_DEBUG_OBJECT (src,
5222 "Attempting authentication using credentials from the properties");
5225 /* FIXME: If the url didn't contain username and password or we tried them
5226 * already, request a username and passwd from the application via some kind
5227 * of credentials request message */
5229 /* If we don't have a username and passwd at this point, bail out. */
5230 if (user == NULL || pass == NULL)
5233 /* Try to configure for each available authentication method, strongest to
5235 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
5236 /* Check if this method is available on the server */
5237 if ((method & avail_methods) == 0)
5240 /* Pass the credentials to the connection to try on the next request */
5241 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
5242 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
5243 * ignore it and end up retrying later */
5244 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
5245 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
5246 gst_rtsp_auth_method_to_string (method));
5251 if (method == GST_RTSP_AUTH_NONE)
5252 goto no_auth_available;
5258 /* Output an error indicating that we couldn't connect because there were
5259 * no supported authentication protocols */
5260 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5261 ("No supported authentication protocol was found"));
5266 /* We don't fire an error message, we just return FALSE and let the
5267 * normal NOT_AUTHORIZED error be propagated */
5272 static GstRTSPResult
5273 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5274 GstRTSPMessage * request, GstRTSPMessage * response,
5275 GstRTSPStatusCode * code)
5278 GstRTSPStatusCode thecode;
5279 gchar *content_base = NULL;
5283 if (!src->short_header)
5284 gst_rtsp_ext_list_before_send (src->extensions, request);
5286 GST_DEBUG_OBJECT (src, "sending message");
5289 gst_rtsp_message_dump (request);
5291 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
5295 gst_rtsp_connection_reset_timeout (conn);
5298 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
5303 gst_rtsp_message_dump (response);
5305 switch (response->type) {
5306 case GST_RTSP_MESSAGE_REQUEST:
5307 res = gst_rtspsrc_handle_request (src, conn, response);
5308 if (res == GST_RTSP_EEOF)
5311 goto handle_request_failed;
5313 case GST_RTSP_MESSAGE_RESPONSE:
5314 /* ok, a response is good */
5315 GST_DEBUG_OBJECT (src, "received response message");
5317 case GST_RTSP_MESSAGE_DATA:
5318 /* get next response */
5319 GST_DEBUG_OBJECT (src, "handle data response message");
5320 gst_rtspsrc_handle_data (src, response);
5323 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
5328 thecode = response->type_data.response.code;
5330 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
5332 /* if the caller wanted the result code, we store it. */
5336 /* If the request didn't succeed, bail out before doing any more */
5337 if (thecode != GST_RTSP_STS_OK)
5340 /* store new content base if any */
5341 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
5344 g_free (src->content_base);
5345 src->content_base = g_strdup (content_base);
5347 gst_rtsp_ext_list_after_send (src->extensions, request, response);
5354 gchar *str = gst_rtsp_strresult (res);
5356 if (res != GST_RTSP_EINTR) {
5357 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5358 ("Could not send message. (%s)", str));
5360 GST_WARNING_OBJECT (src, "send interrupted");
5369 GST_WARNING_OBJECT (src, "server closed connection");
5370 if ((try == 0) && !src->interleaved && src->udp_reconnect) {
5372 /* if reconnect succeeds, try again */
5374 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
5378 /* only try once after reconnect, then fallthrough and error out */
5381 gchar *str = gst_rtsp_strresult (res);
5383 if (res != GST_RTSP_EINTR) {
5384 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5385 ("Could not receive message. (%s)", str));
5387 GST_WARNING_OBJECT (src, "receive interrupted");
5395 handle_request_failed:
5397 /* ERROR was posted */
5398 gst_rtsp_message_unset (response);
5403 GST_DEBUG_OBJECT (src, "we got an eof from the server");
5404 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5405 ("The server closed the connection."));
5406 gst_rtsp_message_unset (response);
5413 * @src: the rtsp source
5414 * @conn: the connection to send on
5415 * @request: must point to a valid request
5416 * @response: must point to an empty #GstRTSPMessage
5417 * @code: an optional code result
5419 * send @request and retrieve the response in @response. optionally @code can be
5420 * non-NULL in which case it will contain the status code of the response.
5422 * If This function returns #GST_RTSP_OK, @response will contain a valid response
5423 * message that should be cleaned with gst_rtsp_message_unset() after usage.
5425 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
5426 * @response message) if the response code was not 200 (OK).
5428 * If the attempt results in an authentication failure, then this will attempt
5429 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
5432 * Returns: #GST_RTSP_OK if the processing was successful.
5434 static GstRTSPResult
5435 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5436 GstRTSPMessage * request, GstRTSPMessage * response,
5437 GstRTSPStatusCode * code)
5439 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
5440 GstRTSPResult res = GST_RTSP_ERROR;
5443 GstRTSPMethod method = GST_RTSP_INVALID;
5449 /* make sure we don't loop forever */
5453 /* save method so we can disable it when the server complains */
5454 method = request->type_data.request.method;
5457 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
5461 case GST_RTSP_STS_UNAUTHORIZED:
5462 if (gst_rtspsrc_setup_auth (src, response)) {
5463 /* Try the request/response again after configuring the auth info
5471 } while (retry == TRUE);
5473 /* If the user requested the code, let them handle errors, otherwise
5474 * post an error below */
5477 else if (int_code != GST_RTSP_STS_OK)
5478 goto error_response;
5485 GST_DEBUG_OBJECT (src, "got error %d", res);
5490 res = GST_RTSP_ERROR;
5492 switch (response->type_data.response.code) {
5493 case GST_RTSP_STS_NOT_FOUND:
5494 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
5495 response->type_data.response.reason));
5497 case GST_RTSP_STS_MOVED_PERMANENTLY:
5498 case GST_RTSP_STS_MOVE_TEMPORARILY:
5500 gchar *new_location;
5501 GstRTSPLowerTrans transports;
5503 GST_DEBUG_OBJECT (src, "got redirection");
5504 /* if we don't have a Location Header, we must error */
5505 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
5506 &new_location, 0) < 0)
5509 /* When we receive a redirect result, we go back to the INIT state after
5510 * parsing the new URI. The caller should do the needed steps to issue
5511 * a new setup when it detects this state change. */
5512 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
5514 /* save current transports */
5515 if (src->conninfo.url)
5516 transports = src->conninfo.url->transports;
5518 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
5520 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
5522 /* set old transports */
5523 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
5524 src->conninfo.url->transports = transports;
5526 src->need_redirect = TRUE;
5527 src->state = GST_RTSP_STATE_INIT;
5531 case GST_RTSP_STS_NOT_ACCEPTABLE:
5532 case GST_RTSP_STS_NOT_IMPLEMENTED:
5533 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
5534 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
5535 gst_rtsp_method_as_text (method));
5536 src->methods &= ~method;
5540 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5541 ("Got error response: %d (%s).", response->type_data.response.code,
5542 response->type_data.response.reason));
5545 /* if we return ERROR we should unset the response ourselves */
5546 if (res == GST_RTSP_ERROR)
5547 gst_rtsp_message_unset (response);
5553 static GstRTSPResult
5554 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
5555 GstRTSPMessage * response, GstRTSPSrc * src)
5557 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
5562 /* parse the response and collect all the supported methods. We need this
5563 * information so that we don't try to send an unsupported request to the
5567 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
5569 GstRTSPHeaderField field;
5573 /* reset supported methods */
5576 /* Try Allow Header first */
5577 field = GST_RTSP_HDR_ALLOW;
5580 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5581 if (indx == 0 && !respoptions) {
5582 /* if no Allow header was found then try the Public header... */
5583 field = GST_RTSP_HDR_PUBLIC;
5584 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5589 src->methods |= gst_rtsp_options_from_text (respoptions);
5594 if (src->methods == 0) {
5595 /* neither Allow nor Public are required, assume the server supports
5596 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
5598 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
5599 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
5601 /* always assume PLAY, FIXME, extensions should be able to override
5603 src->methods |= GST_RTSP_PLAY;
5604 /* also assume it will support Range */
5605 src->seekable = TRUE;
5607 /* we need describe and setup */
5608 if (!(src->methods & GST_RTSP_DESCRIBE))
5610 if (!(src->methods & GST_RTSP_SETUP))
5618 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5619 ("Server does not support DESCRIBE."));
5624 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5625 ("Server does not support SETUP."));
5630 /* masks to be kept in sync with the hardcoded protocol order of preference
5632 static guint protocol_masks[] = {
5633 GST_RTSP_LOWER_TRANS_UDP,
5634 GST_RTSP_LOWER_TRANS_UDP_MCAST,
5635 GST_RTSP_LOWER_TRANS_TCP,
5639 static GstRTSPResult
5640 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
5641 GstRTSPLowerTrans protocols, GstRTSPProfile profile, gchar ** transports)
5645 gboolean add_udp_str;
5650 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
5655 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
5657 /* extension listed transports, use those */
5658 if (*transports != NULL)
5661 /* it's the default */
5662 add_udp_str = FALSE;
5664 /* the default RTSP transports */
5665 result = g_string_new ("RTP");
5668 case GST_RTSP_PROFILE_AVP:
5669 g_string_append (result, "/AVP");
5671 case GST_RTSP_PROFILE_SAVP:
5672 g_string_append (result, "/SAVP");
5674 case GST_RTSP_PROFILE_AVPF:
5675 g_string_append (result, "/AVPF");
5677 case GST_RTSP_PROFILE_SAVPF:
5678 g_string_append (result, "/SAVPF");
5684 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
5685 GST_DEBUG_OBJECT (src, "adding UDP unicast");
5687 g_string_append (result, "/UDP");
5688 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
5689 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
5690 GST_DEBUG_OBJECT (src, "adding UDP multicast");
5691 /* we don't have to allocate any UDP ports yet, if the selected transport
5692 * turns out to be multicast we can create them and join the multicast
5693 * group indicated in the transport reply */
5695 g_string_append (result, "/UDP");
5696 g_string_append (result, ";multicast");
5697 if (src->next_port_num != 0) {
5698 if (src->client_port_range.max > 0 &&
5699 src->next_port_num >= src->client_port_range.max)
5702 g_string_append_printf (result, ";client_port=%d-%d",
5703 src->next_port_num, src->next_port_num + 1);
5705 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
5706 GST_DEBUG_OBJECT (src, "adding TCP");
5708 g_string_append (result, "/TCP;unicast;interleaved=%%i1-%%i2");
5710 *transports = g_string_free (result, FALSE);
5712 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
5719 GST_ERROR ("extension gave error %d", res);
5724 GST_ERROR ("no more ports available");
5725 return GST_RTSP_ERROR;
5729 static GstRTSPResult
5730 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
5731 gint orig_rtpport, gint orig_rtcpport)
5734 gint nr_udp, nr_int;
5736 gint rtpport = 0, rtcpport = 0;
5739 src = stream->parent;
5741 /* find number of placeholders first */
5742 if (strstr (*transports, "%%i2"))
5744 else if (strstr (*transports, "%%i1"))
5749 if (strstr (*transports, "%%u2"))
5751 else if (strstr (*transports, "%%u1"))
5756 if (nr_udp == 0 && nr_int == 0)
5760 if (!orig_rtpport || !orig_rtcpport) {
5761 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
5764 rtpport = orig_rtpport;
5765 rtcpport = orig_rtcpport;
5769 str = g_string_new ("");
5771 while ((next = strstr (p, "%%"))) {
5772 g_string_append_len (str, p, next - p);
5773 if (next[2] == 'u') {
5775 g_string_append_printf (str, "%d", rtpport);
5776 else if (next[3] == '2')
5777 g_string_append_printf (str, "%d", rtcpport);
5779 if (next[2] == 'i') {
5781 g_string_append_printf (str, "%d", src->free_channel);
5782 else if (next[3] == '2')
5783 g_string_append_printf (str, "%d", src->free_channel + 1);
5788 /* append final part */
5789 g_string_append (str, p);
5791 g_free (*transports);
5792 *transports = g_string_free (str, FALSE);
5800 GST_ERROR ("failed to allocate udp ports");
5801 return GST_RTSP_ERROR;
5806 gst_rtspsrc_stream_make_keymgmt (GstRTSPSrc * src, GstRTSPStream * stream)
5809 gchar *result, *base64;
5814 GstMIKEYMessage *msg;
5815 GstMIKEYPayload *payload;
5819 key_data = g_malloc (KEY_SIZE);
5820 for (i = 0; i < KEY_SIZE; i += 4)
5821 GST_WRITE_UINT32_BE (key_data + i, g_random_int ());
5824 gst_buffer_unref (stream->key);
5825 stream->key = gst_buffer_new_wrapped (key_data, KEY_SIZE);
5827 msg = gst_mikey_message_new ();
5828 /* unencrypted MIKEY message, we send this over TLS so this is allowed */
5829 gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT,
5830 FALSE, GST_MIKEY_PRF_MIKEY_1, g_random_int (), GST_MIKEY_MAP_TYPE_SRTP);
5831 /* add policy '0' for our SSRC */
5832 gst_mikey_message_add_cs_srtp (msg, 0, stream->ssrc, 0);
5833 /* timestamp is now */
5834 gst_mikey_message_add_t_now_ntp_utc (msg);
5835 /* add some random data */
5836 gst_mikey_message_add_rand_len (msg, 16);
5838 /* the policy '0' is SRTP */
5839 payload = gst_mikey_payload_new (GST_MIKEY_PT_SP);
5840 gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP);
5842 /* only AES-CM is supported */
5844 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1, &byte);
5845 /* only HMAC-SHA1 */
5846 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1,
5848 /* we enable encryption on RTP and RTCP */
5849 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1,
5851 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1,
5853 /* we enable authentication on RTP and RTCP */
5854 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1,
5856 gst_mikey_message_add_payload (msg, payload);
5858 /* add the key in KEMAC */
5859 gst_mikey_message_add_kemac (msg, GST_MIKEY_ENC_NULL, KEY_SIZE, key_data,
5860 GST_MIKEY_MAC_NULL, NULL);
5862 /* now serialize this to bytes */
5863 bytes = gst_mikey_message_to_bytes (msg);
5864 gst_mikey_message_free (msg);
5865 /* and make it into base64 */
5866 data = g_bytes_get_data (bytes, &size);
5867 base64 = g_base64_encode (data, size);
5868 g_bytes_unref (bytes);
5870 result = g_strdup_printf ("prot=mikey;uri=\"%s\";data=\"%s\"",
5871 stream->conninfo.location, base64);
5878 /* Perform the SETUP request for all the streams.
5880 * We ask the server for a specific transport, which initially includes all the
5881 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
5882 * two local UDP ports that we send to the server.
5884 * Once the server replied with a transport, we configure the other streams
5885 * with the same transport.
5887 * This function will also configure the stream for the selected transport,
5888 * which basically means creating the pipeline.
5890 static GstRTSPResult
5891 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
5894 GstRTSPResult res = GST_RTSP_ERROR;
5895 GstRTSPMessage request = { 0 };
5896 GstRTSPMessage response = { 0 };
5897 GstRTSPStream *stream = NULL;
5898 GstRTSPLowerTrans protocols;
5899 GstRTSPStatusCode code;
5900 gboolean unsupported_real = FALSE;
5901 gint rtpport, rtcpport;
5905 if (src->conninfo.connection) {
5906 url = gst_rtsp_connection_get_url (src->conninfo.connection);
5907 /* we initially allow all configured lower transports. based on the URL
5908 * transports and the replies from the server we narrow them down. */
5909 protocols = url->transports & src->cur_protocols;
5912 protocols = src->cur_protocols;
5918 /* reset some state */
5919 src->free_channel = 0;
5920 src->interleaved = FALSE;
5921 src->need_activate = FALSE;
5922 /* keep track of next port number, 0 is random */
5923 src->next_port_num = src->client_port_range.min;
5924 rtpport = rtcpport = 0;
5926 if (G_UNLIKELY (src->streams == NULL))
5929 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5930 GstRTSPConnection *conn;
5937 stream = (GstRTSPStream *) walk->data;
5939 caps = stream_get_caps_for_pt (stream, stream->default_pt);
5941 GST_DEBUG_OBJECT (src, "skipping stream %p, no caps", stream);
5945 if (stream->skipped) {
5946 GST_DEBUG_OBJECT (src, "skipping stream %p", stream);
5950 /* see if we need to configure this stream */
5951 if (!gst_rtsp_ext_list_configure_stream (src->extensions, caps)) {
5952 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
5957 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_SELECT_STREAM], 0,
5958 stream->id, caps, &selected);
5960 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by signal", stream);
5964 /* merge/overwrite global caps */
5969 s = gst_caps_get_structure (caps, 0);
5971 num = gst_structure_n_fields (src->props);
5972 for (j = 0; j < num; j++) {
5976 name = gst_structure_nth_field_name (src->props, j);
5977 val = gst_structure_get_value (src->props, name);
5978 gst_structure_set_value (s, name, val);
5980 GST_DEBUG_OBJECT (src, "copied %s", name);
5984 /* skip setup if we have no URL for it */
5985 if (stream->conninfo.location == NULL) {
5986 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
5990 if (src->conninfo.connection == NULL) {
5991 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
5992 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
5995 conn = stream->conninfo.connection;
5997 conn = src->conninfo.connection;
5999 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
6000 stream->conninfo.location);
6002 /* if we have a multicast connection, only suggest multicast from now on */
6003 if (stream->is_multicast)
6004 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
6007 /* first selectable protocol */
6008 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
6010 if (!protocol_masks[mask])
6014 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
6015 protocol_masks[mask]);
6016 /* create a string with first transport in line */
6018 res = gst_rtspsrc_create_transports_string (src,
6019 protocols & protocol_masks[mask], stream->profile, &transports);
6020 if (res < 0 || transports == NULL)
6021 goto setup_transport_failed;
6023 if (strlen (transports) == 0) {
6024 g_free (transports);
6025 GST_DEBUG_OBJECT (src, "no transports found");
6030 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
6032 /* replace placeholders with real values, this function will optionally
6033 * allocate UDP ports and other info needed to execute the setup request */
6034 res = gst_rtspsrc_prepare_transports (stream, &transports,
6035 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
6037 g_free (transports);
6038 goto setup_transport_failed;
6041 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
6043 /* create SETUP request */
6045 gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
6046 stream->conninfo.location);
6048 g_free (transports);
6049 goto create_request_failed;
6052 /* select transport */
6053 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
6056 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
6057 stream->profile == GST_RTSP_PROFILE_SAVPF) {
6058 hval = gst_rtspsrc_stream_make_keymgmt (src, stream);
6059 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_KEYMGMT, hval);
6062 /* if the user wants a non default RTP packet size we add the blocksize
6064 if (src->rtp_blocksize > 0) {
6065 hval = g_strdup_printf ("%d", src->rtp_blocksize);
6066 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
6070 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
6073 /* handle the code ourselves */
6074 if ((res = gst_rtspsrc_send (src, conn, &request, &response, &code) < 0))
6078 case GST_RTSP_STS_OK:
6080 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
6081 gst_rtsp_message_unset (&request);
6082 gst_rtsp_message_unset (&response);
6083 /* cleanup of leftover transport */
6084 gst_rtspsrc_stream_free_udp (stream);
6085 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
6086 * we might be in this case */
6087 if (stream->container && rtpport && rtcpport && !retry) {
6088 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
6093 /* this transport did not go down well, but we may have others to try
6094 * that we did not send yet, try those and only give up then
6095 * but not without checking for lost cause/extension so we can
6096 * post a nicer/more useful error message later */
6097 if (!unsupported_real)
6098 unsupported_real = stream->is_real;
6099 /* select next available protocol, give up on this stream if none */
6101 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
6103 if (!protocol_masks[mask] || unsupported_real)
6108 /* cleanup of leftover transport and move to the next stream */
6109 gst_rtspsrc_stream_free_udp (stream);
6110 goto response_error;
6113 /* parse response transport */
6115 gchar *resptrans = NULL;
6116 GstRTSPTransport transport = { 0 };
6118 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
6121 gst_rtspsrc_stream_free_udp (stream);
6125 /* parse transport, go to next stream on parse error */
6126 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
6127 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
6131 /* update allowed transports for other streams. once the transport of
6132 * one stream has been determined, we make sure that all other streams
6133 * are configured in the same way */
6134 switch (transport.lower_transport) {
6135 case GST_RTSP_LOWER_TRANS_TCP:
6136 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
6137 protocols = GST_RTSP_LOWER_TRANS_TCP;
6138 src->interleaved = TRUE;
6139 /* update free channels */
6141 MAX (transport.interleaved.min, src->free_channel);
6143 MAX (transport.interleaved.max, src->free_channel);
6144 src->free_channel++;
6146 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
6147 /* only allow multicast for other streams */
6148 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
6149 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
6150 /* if the server selected our ports, increment our counters so that
6151 * we select a new port later */
6152 if (src->next_port_num == transport.port.min &&
6153 src->next_port_num + 1 == transport.port.max) {
6154 src->next_port_num += 2;
6157 case GST_RTSP_LOWER_TRANS_UDP:
6158 /* only allow unicast for other streams */
6159 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
6160 protocols = GST_RTSP_LOWER_TRANS_UDP;
6163 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
6164 transport.lower_transport);
6168 if (!stream->container || (!src->interleaved && !retry)) {
6169 /* now configure the stream with the selected transport */
6170 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
6171 GST_DEBUG_OBJECT (src,
6172 "could not configure stream %p transport, skipping stream",
6175 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
6176 /* retain the first allocated UDP port pair */
6177 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
6178 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
6181 /* we need to activate at least one streams when we detect activity */
6182 src->need_activate = TRUE;
6184 /* stream is setup now */
6185 stream->setup = TRUE;
6190 GstRTSPStream *sskip;
6192 skip = g_list_next (skip);
6196 sskip = (GstRTSPStream *) skip->data;
6198 /* skip all streams with the same control url */
6199 if (g_str_equal (stream->conninfo.location, sskip->conninfo.location)) {
6200 GST_DEBUG_OBJECT (src, "found stream %p with same control %s",
6201 sskip, sskip->conninfo.location);
6202 sskip->skipped = TRUE;
6207 /* clean up our transport struct */
6208 gst_rtsp_transport_init (&transport);
6209 /* clean up used RTSP messages */
6210 gst_rtsp_message_unset (&request);
6211 gst_rtsp_message_unset (&response);
6215 /* store the transport protocol that was configured */
6216 src->cur_protocols = protocols;
6218 gst_rtsp_ext_list_stream_select (src->extensions, url);
6220 /* if there is nothing to activate, error out */
6221 if (!src->need_activate)
6222 goto nothing_to_activate;
6229 /* no transport possible, post an error and stop */
6230 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
6231 ("Could not connect to server, no protocols left"));
6232 return GST_RTSP_ERROR;
6236 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6237 ("SDP contains no streams"));
6238 return GST_RTSP_ERROR;
6240 create_request_failed:
6242 gchar *str = gst_rtsp_strresult (res);
6244 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6245 ("Could not create request. (%s)", str));
6249 setup_transport_failed:
6251 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6252 ("Could not setup transport."));
6253 res = GST_RTSP_ERROR;
6258 const gchar *str = gst_rtsp_status_as_text (code);
6260 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6261 ("Error (%d): %s", code, GST_STR_NULL (str)));
6262 res = GST_RTSP_ERROR;
6267 gchar *str = gst_rtsp_strresult (res);
6269 if (res != GST_RTSP_EINTR) {
6270 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6271 ("Could not send message. (%s)", str));
6273 GST_WARNING_OBJECT (src, "send interrupted");
6280 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6281 ("Server did not select transport."));
6282 res = GST_RTSP_ERROR;
6285 nothing_to_activate:
6287 /* none of the available error codes is really right .. */
6288 if (unsupported_real) {
6289 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6290 (_("No supported stream was found. You might need to install a "
6291 "GStreamer RTSP extension plugin for Real media streams.")),
6294 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6295 (_("No supported stream was found. You might need to allow "
6296 "more transport protocols or may otherwise be missing "
6297 "the right GStreamer RTSP extension plugin.")), (NULL));
6299 return GST_RTSP_ERROR;
6303 gst_rtsp_message_unset (&request);
6304 gst_rtsp_message_unset (&response);
6310 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
6311 GstSegment * segment)
6314 GstRTSPTimeRange *therange;
6317 gst_rtsp_range_free (src->range);
6319 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
6320 GST_DEBUG_OBJECT (src, "parsed range %s", range);
6321 src->range = therange;
6323 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
6325 gst_segment_init (segment, GST_FORMAT_TIME);
6329 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
6330 therange->min.type, therange->min.seconds, therange->max.type,
6331 therange->max.seconds);
6333 if (therange->min.type == GST_RTSP_TIME_NOW)
6335 else if (therange->min.type == GST_RTSP_TIME_END)
6338 seconds = therange->min.seconds * GST_SECOND;
6340 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
6341 GST_TIME_ARGS (seconds));
6343 /* we need to start playback without clipping from the position reported by
6345 segment->start = seconds;
6346 segment->position = seconds;
6348 if (therange->max.type == GST_RTSP_TIME_NOW)
6350 else if (therange->max.type == GST_RTSP_TIME_END)
6353 seconds = therange->max.seconds * GST_SECOND;
6355 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
6356 GST_TIME_ARGS (seconds));
6358 /* live (WMS) server might send overflowed large max as its idea of infinity,
6359 * compensate to prevent problems later on */
6360 if (seconds != -1 && seconds < 0) {
6362 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
6365 /* live (WMS) might send min == max, which is not worth recording */
6366 if (segment->duration == -1 && seconds == segment->start)
6369 /* don't change duration with unknown value, we might have a valid value
6370 * there that we want to keep. */
6372 segment->duration = seconds;
6377 /* Parse clock profived by the server with following syntax:
6379 * "GstNetTimeProvider <wrapped-clock> <server-IP:port> <clock-time>"
6382 gst_rtspsrc_parse_gst_clock (GstRTSPSrc * src, const gchar * gstclock)
6384 gboolean res = FALSE;
6386 if (g_str_has_prefix (gstclock, "GstNetTimeProvider ")) {
6387 gchar **fields = NULL, **parts = NULL;
6388 gchar *remote_ip, *str;
6390 GstClockTime base_time;
6393 fields = g_strsplit (gstclock, " ", 0);
6395 /* wrapped clock, not very interesting for now */
6396 if (fields[1] == NULL)
6399 /* remote IP address and port */
6400 if ((str = fields[2]) == NULL)
6403 parts = g_strsplit (str, ":", 0);
6405 if ((remote_ip = parts[0]) == NULL)
6408 if ((str = parts[1]) == NULL)
6416 if ((str = fields[3]) == NULL)
6419 base_time = g_ascii_strtoull (str, NULL, 10);
6422 gst_net_client_clock_new ((gchar *) "GstRTSPClock", remote_ip, port,
6425 if (src->provided_clock)
6426 gst_object_unref (src->provided_clock);
6427 src->provided_clock = netclock;
6429 gst_element_post_message (GST_ELEMENT_CAST (src),
6430 gst_message_new_clock_provide (GST_OBJECT_CAST (src),
6431 src->provided_clock, TRUE));
6435 g_strfreev (fields);
6441 /* must be called with the RTSP state lock */
6442 static GstRTSPResult
6443 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
6449 /* prepare global stream caps properties */
6451 gst_structure_remove_all_fields (src->props);
6453 src->props = gst_structure_new_empty ("RTSPProperties");
6456 gst_sdp_message_dump (sdp);
6458 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
6460 /* let the app inspect and change the SDP */
6461 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_ON_SDP], 0, sdp);
6463 gst_segment_init (&src->segment, GST_FORMAT_TIME);
6465 /* parse range for duration reporting. */
6470 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
6474 /* keep track of the range and configure it in the segment */
6475 if (gst_rtspsrc_parse_range (src, range, &src->segment))
6479 /* parse clock information. This is GStreamer specific, a server can tell the
6480 * client what clock it is using and wrap that in a network clock. The
6481 * advantage of that is that we can slave to it. */
6483 const gchar *gstclock;
6486 gstclock = gst_sdp_message_get_attribute_val_n (sdp, "x-gst-clock", i);
6487 if (gstclock == NULL)
6490 /* parse the clock and expose it in the provide_clock method */
6491 if (gst_rtspsrc_parse_gst_clock (src, gstclock))
6495 /* try to find a global control attribute. Note that a '*' means that we should
6496 * do aggregate control with the current url (so we don't do anything and
6497 * leave the current connection as is) */
6499 const gchar *control;
6502 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
6503 if (control == NULL)
6506 /* only take fully qualified urls */
6507 if (g_str_has_prefix (control, "rtsp://"))
6511 g_free (src->conninfo.location);
6512 src->conninfo.location = g_strdup (control);
6513 /* make a connection for this, if there was a connection already, nothing
6515 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
6516 GST_ERROR_OBJECT (src, "could not connect");
6519 /* we need to keep the control url separate from the connection url because
6520 * the rules for constructing the media control url need it */
6521 g_free (src->control);
6522 src->control = g_strdup (control);
6525 /* create streams */
6526 n_streams = gst_sdp_message_medias_len (sdp);
6527 for (i = 0; i < n_streams; i++) {
6528 gst_rtspsrc_create_stream (src, sdp, i);
6531 src->state = GST_RTSP_STATE_INIT;
6534 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
6537 /* reset our state */
6538 src->need_range = TRUE;
6541 src->state = GST_RTSP_STATE_READY;
6548 GST_ERROR_OBJECT (src, "setup failed");
6549 gst_rtspsrc_cleanup (src);
6554 static GstRTSPResult
6555 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
6559 GstRTSPMessage request = { 0 };
6560 GstRTSPMessage response = { 0 };
6563 gchar *respcont = NULL;
6566 src->need_redirect = FALSE;
6568 /* can't continue without a valid url */
6569 if (G_UNLIKELY (src->conninfo.url == NULL)) {
6570 res = GST_RTSP_EINVAL;
6573 src->tried_url_auth = FALSE;
6575 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
6576 goto connect_failed;
6578 /* create OPTIONS */
6579 GST_DEBUG_OBJECT (src, "create options...");
6581 gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
6582 src->conninfo.url_str);
6584 goto create_request_failed;
6587 GST_DEBUG_OBJECT (src, "send options...");
6590 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
6593 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6598 if (!gst_rtspsrc_parse_methods (src, &response))
6601 /* create DESCRIBE */
6602 GST_DEBUG_OBJECT (src, "create describe...");
6604 gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
6605 src->conninfo.url_str);
6607 goto create_request_failed;
6609 /* we only accept SDP for now */
6610 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
6614 GST_DEBUG_OBJECT (src, "send describe...");
6617 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
6620 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6624 /* we only perform redirect for the describe, currently */
6625 if (src->need_redirect) {
6626 /* close connection, we don't have to send a TEARDOWN yet, ignore the
6628 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6630 gst_rtsp_message_unset (&request);
6631 gst_rtsp_message_unset (&response);
6637 /* it could be that the DESCRIBE method was not implemented */
6638 if (!src->methods & GST_RTSP_DESCRIBE)
6641 /* check if reply is SDP */
6642 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
6644 /* could not be set but since the request returned OK, we assume it
6645 * was SDP, else check it. */
6647 if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
6648 goto wrong_content_type;
6651 /* get message body and parse as SDP */
6652 gst_rtsp_message_get_body (&response, &data, &size);
6653 if (data == NULL || size == 0)
6656 GST_DEBUG_OBJECT (src, "parse SDP...");
6657 gst_sdp_message_new (sdp);
6658 gst_sdp_message_parse_buffer (data, size, *sdp);
6660 /* clean up any messages */
6661 gst_rtsp_message_unset (&request);
6662 gst_rtsp_message_unset (&response);
6669 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
6670 ("No valid RTSP URL was provided"));
6675 gchar *str = gst_rtsp_strresult (res);
6677 if (res != GST_RTSP_EINTR) {
6678 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
6679 ("Failed to connect. (%s)", str));
6681 GST_WARNING_OBJECT (src, "connect interrupted");
6686 create_request_failed:
6688 gchar *str = gst_rtsp_strresult (res);
6690 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6691 ("Could not create request. (%s)", str));
6697 /* Don't post a message - the rtsp_send method will have
6698 * taken care of it because we passed NULL for the response code */
6703 /* error was posted */
6704 res = GST_RTSP_ERROR;
6709 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6710 ("Server does not support SDP, got %s.", respcont));
6711 res = GST_RTSP_ERROR;
6716 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6717 ("Server can not provide an SDP."));
6718 res = GST_RTSP_ERROR;
6723 if (src->conninfo.connection) {
6724 GST_DEBUG_OBJECT (src, "free connection");
6725 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6727 gst_rtsp_message_unset (&request);
6728 gst_rtsp_message_unset (&response);
6733 static GstRTSPResult
6734 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
6739 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
6741 if (src->sdp == NULL) {
6742 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
6746 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
6751 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
6758 GST_WARNING_OBJECT (src, "can't get sdp");
6759 src->open_error = TRUE;
6764 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
6765 src->open_error = TRUE;
6770 static GstRTSPResult
6771 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
6773 GstRTSPMessage request = { 0 };
6774 GstRTSPMessage response = { 0 };
6775 GstRTSPResult res = GST_RTSP_OK;
6777 const gchar *control;
6779 GST_DEBUG_OBJECT (src, "TEARDOWN...");
6781 gst_rtspsrc_set_state (src, GST_STATE_READY);
6783 if (src->state < GST_RTSP_STATE_READY) {
6784 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
6791 /* construct a control url */
6792 control = get_aggregate_control (src);
6794 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
6797 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6798 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6799 const gchar *setup_url;
6800 GstRTSPConnInfo *info;
6802 /* try aggregate control first but do non-aggregate control otherwise */
6804 setup_url = control;
6805 else if ((setup_url = stream->conninfo.location) == NULL)
6808 if (src->conninfo.connection) {
6809 info = &src->conninfo;
6810 } else if (stream->conninfo.connection) {
6811 info = &stream->conninfo;
6815 if (!info->connected)
6820 gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
6822 goto create_request_failed;
6825 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
6828 gst_rtspsrc_send (src, info->connection, &request, &response,
6832 /* FIXME, parse result? */
6833 gst_rtsp_message_unset (&request);
6834 gst_rtsp_message_unset (&response);
6837 /* early exit when we did aggregate control */
6843 /* close connections */
6844 GST_DEBUG_OBJECT (src, "closing connection...");
6845 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6846 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6847 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6848 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
6852 gst_rtspsrc_cleanup (src);
6854 src->state = GST_RTSP_STATE_INVALID;
6857 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
6862 create_request_failed:
6864 gchar *str = gst_rtsp_strresult (res);
6866 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6867 ("Could not create request. (%s)", str));
6873 gchar *str = gst_rtsp_strresult (res);
6875 gst_rtsp_message_unset (&request);
6876 if (res != GST_RTSP_EINTR) {
6877 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6878 ("Could not send message. (%s)", str));
6880 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
6887 GST_DEBUG_OBJECT (src,
6888 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
6893 /* RTP-Info is of the format:
6895 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
6897 * rtptime corresponds to the timestamp for the NPT time given in the header
6898 * seqbase corresponds to the next sequence number we received. This number
6899 * indicates the first seqnum after the seek and should be used to discard
6900 * packets that are from before the seek.
6903 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
6908 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
6910 infos = g_strsplit (rtpinfo, ",", 0);
6911 for (i = 0; infos[i]; i++) {
6913 GstRTSPStream *stream;
6917 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
6919 /* init values, types of seqbase and timebase are bigger than needed so we
6920 * can store -1 as uninitialized values */
6925 /* parse url, find stream for url.
6926 * parse seq and rtptime. The seq number should be configured in the rtp
6927 * depayloader or session manager to detect gaps. Same for the rtptime, it
6928 * should be used to create an initial time newsegment. */
6929 fields = g_strsplit (infos[i], ";", 0);
6930 for (j = 0; fields[j]; j++) {
6931 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
6932 /* remove leading whitespace */
6933 fields[j] = g_strchug (fields[j]);
6934 if (g_str_has_prefix (fields[j], "url=")) {
6935 /* get the url and the stream */
6937 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
6938 } else if (g_str_has_prefix (fields[j], "seq=")) {
6939 seqbase = atoi (fields[j] + 4);
6940 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
6941 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
6944 g_strfreev (fields);
6945 /* now we need to store the values for the caps of the stream */
6946 if (stream != NULL) {
6947 GST_DEBUG_OBJECT (src,
6948 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
6949 stream, seqbase, timebase);
6951 /* we have a stream, configure detected params */
6952 stream->seqbase = seqbase;
6953 stream->timebase = timebase;
6962 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
6967 interval = strtoul (rtcp, NULL, 10);
6968 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
6973 interval *= GST_MSECOND;
6975 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6976 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6978 /* already (optionally) retrieved this when configuring manager */
6979 if (stream->session) {
6980 GObject *rtpsession = stream->session;
6982 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
6984 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
6988 /* now it happens that (Xenon) server sending this may also provide bogus
6989 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
6990 * and just use RTP-Info to sync */
6992 GObjectClass *klass;
6994 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
6995 if (g_object_class_find_property (klass, "rtcp-sync")) {
6996 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
6997 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
7003 gst_rtspsrc_get_float (const gchar * dstr)
7005 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
7007 /* canonicalise floating point string so we can handle float strings
7008 * in the form "24.930" or "24,930" irrespective of the current locale */
7009 g_strlcpy (s, dstr, sizeof (s));
7010 g_strdelimit (s, ",", '.');
7011 return g_ascii_strtod (s, NULL);
7015 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
7017 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
7019 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
7020 g_strlcpy (val_str, "now", sizeof (val_str));
7022 if (segment->position == 0) {
7023 g_strlcpy (val_str, "0", sizeof (val_str));
7025 g_ascii_dtostr (val_str, sizeof (val_str),
7026 ((gdouble) segment->position) / GST_SECOND);
7029 return g_strdup_printf ("npt=%s-", val_str);
7033 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
7037 stream->timebase = -1;
7038 stream->seqbase = -1;
7040 len = stream->ptmap->len;
7041 for (i = 0; i < len; i++) {
7042 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
7045 if (item->caps == NULL)
7048 item->caps = gst_caps_make_writable (item->caps);
7049 s = gst_caps_get_structure (item->caps, 0);
7050 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
7054 static GstRTSPResult
7055 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
7057 GstRTSPResult res = GST_RTSP_OK;
7059 if (src->state < GST_RTSP_STATE_READY) {
7060 res = GST_RTSP_ERROR;
7061 if (src->open_error) {
7062 GST_DEBUG_OBJECT (src, "the stream was in error");
7066 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
7068 if ((res = gst_rtspsrc_open (src, async)) < 0) {
7069 GST_DEBUG_OBJECT (src, "failed to open stream");
7078 static GstRTSPResult
7079 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
7081 GstRTSPMessage request = { 0 };
7082 GstRTSPMessage response = { 0 };
7083 GstRTSPResult res = GST_RTSP_OK;
7087 const gchar *control;
7089 GST_DEBUG_OBJECT (src, "PLAY...");
7091 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
7094 if (!(src->methods & GST_RTSP_PLAY))
7097 if (src->state == GST_RTSP_STATE_PLAYING)
7100 if (!src->conninfo.connection || !src->conninfo.connected)
7103 /* send some dummy packets before we activate the receive in the
7105 gst_rtspsrc_send_dummy_packets (src);
7107 /* require new SR packets */
7109 g_signal_emit_by_name (src->manager, "reset-sync", NULL);
7111 gst_rtspsrc_set_state (src, GST_STATE_PLAYING);
7113 /* construct a control url */
7114 control = get_aggregate_control (src);
7116 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7117 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7118 const gchar *setup_url;
7119 GstRTSPConnection *conn;
7121 /* try aggregate control first but do non-aggregate control otherwise */
7123 setup_url = control;
7124 else if ((setup_url = stream->conninfo.location) == NULL)
7127 if (src->conninfo.connection) {
7128 conn = src->conninfo.connection;
7129 } else if (stream->conninfo.connection) {
7130 conn = stream->conninfo.connection;
7136 res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
7138 goto create_request_failed;
7140 if (src->need_range) {
7141 hval = gen_range_header (src, segment);
7143 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
7145 /* store the newsegment event so it can be sent from the streaming thread. */
7146 if (src->start_segment)
7147 gst_event_unref (src->start_segment);
7148 src->start_segment = gst_event_new_segment (&src->segment);
7151 if (segment->rate != 1.0) {
7152 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
7154 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
7156 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
7158 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
7162 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
7164 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7167 /* seek may have silently failed as it is not supported */
7168 if (!(src->methods & GST_RTSP_PLAY)) {
7169 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
7170 /* obviously it is supported as we made it here */
7171 src->methods |= GST_RTSP_PLAY;
7172 src->seekable = FALSE;
7173 /* but there is nothing to parse in the response,
7174 * so convey we have no idea and not to expect anything particular */
7175 clear_rtp_base (src, stream);
7179 /* need to do for all streams */
7180 for (run = src->streams; run; run = g_list_next (run))
7181 clear_rtp_base (src, (GstRTSPStream *) run->data);
7183 /* NOTE the above also disables npt based eos detection */
7184 /* and below forces position to 0,
7185 * which is visible feedback we lost the plot */
7186 segment->start = segment->position = src->last_pos;
7189 gst_rtsp_message_unset (&request);
7191 /* parse RTP npt field. This is the current position in the stream (Normal
7192 * Play Time) and should be put in the NEWSEGMENT position field. */
7193 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
7195 gst_rtspsrc_parse_range (src, hval, segment);
7197 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
7198 segment->rate = 1.0;
7200 /* parse Speed header. This is the intended playback rate of the stream
7201 * and should be put in the NEWSEGMENT rate field. */
7202 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
7203 0) == GST_RTSP_OK) {
7204 segment->rate = gst_rtspsrc_get_float (hval);
7205 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
7206 &hval, 0) == GST_RTSP_OK) {
7207 segment->rate = gst_rtspsrc_get_float (hval);
7210 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
7211 * for the RTP packets. If this is not present, we assume all starts from 0...
7212 * This is info for the RTP session manager that we pass to it in caps. */
7214 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
7215 &hval, hval_idx++) == GST_RTSP_OK)
7216 gst_rtspsrc_parse_rtpinfo (src, hval);
7218 /* some servers indicate RTCP parameters in PLAY response,
7219 * rather than properly in SDP */
7220 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
7221 &hval, 0) == GST_RTSP_OK)
7222 gst_rtspsrc_handle_rtcp_interval (src, hval);
7224 gst_rtsp_message_unset (&response);
7226 /* early exit when we did aggregate control */
7230 /* configure the caps of the streams after we parsed all headers. Only reset
7231 * the manager object when we set a new Range header (we did a seek) */
7232 gst_rtspsrc_configure_caps (src, segment, src->need_range);
7234 /* set again when needed */
7235 src->need_range = FALSE;
7237 src->running = TRUE;
7238 src->base_time = -1;
7239 src->state = GST_RTSP_STATE_PLAYING;
7242 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
7243 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7244 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7245 stream->discont = TRUE;
7250 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
7257 GST_DEBUG_OBJECT (src, "failed to open stream");
7262 GST_DEBUG_OBJECT (src, "PLAY is not supported");
7267 GST_DEBUG_OBJECT (src, "we were already PLAYING");
7270 create_request_failed:
7272 gchar *str = gst_rtsp_strresult (res);
7274 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7275 ("Could not create request. (%s)", str));
7281 gchar *str = gst_rtsp_strresult (res);
7283 gst_rtsp_message_unset (&request);
7284 if (res != GST_RTSP_EINTR) {
7285 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7286 ("Could not send message. (%s)", str));
7288 GST_WARNING_OBJECT (src, "PLAY interrupted");
7295 static GstRTSPResult
7296 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
7298 GstRTSPResult res = GST_RTSP_OK;
7299 GstRTSPMessage request = { 0 };
7300 GstRTSPMessage response = { 0 };
7302 const gchar *control;
7304 GST_DEBUG_OBJECT (src, "PAUSE...");
7306 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
7309 if (!(src->methods & GST_RTSP_PAUSE))
7312 if (src->state == GST_RTSP_STATE_READY)
7315 if (!src->conninfo.connection || !src->conninfo.connected)
7318 /* construct a control url */
7319 control = get_aggregate_control (src);
7321 /* loop over the streams. We might exit the loop early when we could do an
7322 * aggregate control */
7323 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7324 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7325 GstRTSPConnection *conn;
7326 const gchar *setup_url;
7328 /* try aggregate control first but do non-aggregate control otherwise */
7330 setup_url = control;
7331 else if ((setup_url = stream->conninfo.location) == NULL)
7334 if (src->conninfo.connection) {
7335 conn = src->conninfo.connection;
7336 } else if (stream->conninfo.connection) {
7337 conn = stream->conninfo.connection;
7343 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
7344 ("Sending PAUSE request"));
7347 gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
7349 goto create_request_failed;
7351 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7354 gst_rtsp_message_unset (&request);
7355 gst_rtsp_message_unset (&response);
7357 /* exit early when we did agregate control */
7362 /* change element states now */
7363 gst_rtspsrc_set_state (src, GST_STATE_PAUSED);
7366 src->state = GST_RTSP_STATE_READY;
7370 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
7377 GST_DEBUG_OBJECT (src, "failed to open stream");
7382 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
7387 GST_DEBUG_OBJECT (src, "we were already PAUSED");
7390 create_request_failed:
7392 gchar *str = gst_rtsp_strresult (res);
7394 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7395 ("Could not create request. (%s)", str));
7401 gchar *str = gst_rtsp_strresult (res);
7403 gst_rtsp_message_unset (&request);
7404 if (res != GST_RTSP_EINTR) {
7405 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7406 ("Could not send message. (%s)", str));
7408 GST_WARNING_OBJECT (src, "PAUSE interrupted");
7416 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
7418 GstRTSPSrc *rtspsrc;
7420 rtspsrc = GST_RTSPSRC (bin);
7422 switch (GST_MESSAGE_TYPE (message)) {
7423 case GST_MESSAGE_EOS:
7424 gst_message_unref (message);
7426 case GST_MESSAGE_ELEMENT:
7428 const GstStructure *s = gst_message_get_structure (message);
7430 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
7431 gboolean ignore_timeout;
7433 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
7435 GST_OBJECT_LOCK (rtspsrc);
7436 ignore_timeout = rtspsrc->ignore_timeout;
7437 rtspsrc->ignore_timeout = TRUE;
7438 GST_OBJECT_UNLOCK (rtspsrc);
7440 /* we only act on the first udp timeout message, others are irrelevant
7441 * and can be ignored. */
7442 if (!ignore_timeout)
7443 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
7445 gst_message_unref (message);
7448 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7451 case GST_MESSAGE_ERROR:
7454 GstRTSPStream *stream;
7457 udpsrc = GST_MESSAGE_SRC (message);
7459 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
7460 GST_ELEMENT_NAME (udpsrc));
7462 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
7466 /* we ignore the RTCP udpsrc */
7467 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
7470 /* if we get error messages from the udp sources, that's not a problem as
7471 * long as not all of them error out. We also don't really know what the
7472 * problem is, the message does not give enough detail... */
7473 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
7474 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
7475 if (ret != GST_FLOW_OK)
7479 gst_message_unref (message);
7483 /* fatal but not our message, forward */
7484 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7489 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7495 /* the thread where everything happens */
7497 gst_rtspsrc_thread (GstRTSPSrc * src)
7501 GST_OBJECT_LOCK (src);
7502 cmd = src->pending_cmd;
7503 if (cmd == CMD_RECONNECT || cmd == CMD_PLAY || cmd == CMD_PAUSE
7504 || cmd == CMD_LOOP || cmd == CMD_OPEN)
7505 src->pending_cmd = CMD_LOOP;
7507 src->pending_cmd = CMD_WAIT;
7508 GST_DEBUG_OBJECT (src, "got command %d", cmd);
7510 /* we got the message command, so ensure communication is possible again */
7511 gst_rtspsrc_connection_flush (src, FALSE);
7513 src->busy_cmd = cmd;
7514 GST_OBJECT_UNLOCK (src);
7518 gst_rtspsrc_open (src, TRUE);
7521 gst_rtspsrc_play (src, &src->segment, TRUE);
7524 gst_rtspsrc_pause (src, TRUE);
7527 gst_rtspsrc_close (src, TRUE, FALSE);
7530 gst_rtspsrc_loop (src);
7533 gst_rtspsrc_reconnect (src, FALSE);
7539 GST_OBJECT_LOCK (src);
7540 /* and go back to sleep */
7541 if (src->pending_cmd == CMD_WAIT) {
7543 gst_task_pause (src->task);
7546 src->busy_cmd = CMD_WAIT;
7547 GST_OBJECT_UNLOCK (src);
7551 gst_rtspsrc_start (GstRTSPSrc * src)
7553 GST_DEBUG_OBJECT (src, "starting");
7555 GST_OBJECT_LOCK (src);
7557 src->pending_cmd = CMD_WAIT;
7559 if (src->task == NULL) {
7560 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src, NULL);
7561 if (src->task == NULL)
7564 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
7566 GST_OBJECT_UNLOCK (src);
7573 GST_OBJECT_UNLOCK (src);
7574 GST_ERROR_OBJECT (src, "failed to create task");
7580 gst_rtspsrc_stop (GstRTSPSrc * src)
7584 GST_DEBUG_OBJECT (src, "stopping");
7586 /* also cancels pending task */
7587 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_ALL);
7589 GST_OBJECT_LOCK (src);
7590 if ((task = src->task)) {
7592 GST_OBJECT_UNLOCK (src);
7594 gst_task_stop (task);
7596 /* make sure it is not running */
7597 GST_RTSP_STREAM_LOCK (src);
7598 GST_RTSP_STREAM_UNLOCK (src);
7600 /* now wait for the task to finish */
7601 gst_task_join (task);
7603 /* and free the task */
7604 gst_object_unref (GST_OBJECT (task));
7606 GST_OBJECT_LOCK (src);
7608 GST_OBJECT_UNLOCK (src);
7610 /* ensure synchronously all is closed and clean */
7611 gst_rtspsrc_close (src, FALSE, TRUE);
7616 static GstStateChangeReturn
7617 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
7619 GstRTSPSrc *rtspsrc;
7620 GstStateChangeReturn ret;
7622 rtspsrc = GST_RTSPSRC (element);
7624 switch (transition) {
7625 case GST_STATE_CHANGE_NULL_TO_READY:
7626 if (!gst_rtspsrc_start (rtspsrc))
7629 case GST_STATE_CHANGE_READY_TO_PAUSED:
7630 /* init some state */
7631 rtspsrc->cur_protocols = rtspsrc->protocols;
7632 /* first attempt, don't ignore timeouts */
7633 rtspsrc->ignore_timeout = FALSE;
7634 rtspsrc->open_error = FALSE;
7635 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
7637 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
7638 set_manager_buffer_mode (rtspsrc);
7640 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
7641 /* unblock the tcp tasks and make the loop waiting */
7642 if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
7643 /* make sure it is waiting before we send PAUSE or PLAY below */
7644 GST_RTSP_STREAM_LOCK (rtspsrc);
7645 GST_RTSP_STREAM_UNLOCK (rtspsrc);
7648 case GST_STATE_CHANGE_PAUSED_TO_READY:
7654 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
7655 if (ret == GST_STATE_CHANGE_FAILURE)
7658 switch (transition) {
7659 case GST_STATE_CHANGE_NULL_TO_READY:
7660 ret = GST_STATE_CHANGE_SUCCESS;
7662 case GST_STATE_CHANGE_READY_TO_PAUSED:
7663 ret = GST_STATE_CHANGE_NO_PREROLL;
7665 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
7666 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
7667 ret = GST_STATE_CHANGE_SUCCESS;
7669 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
7670 /* send pause request and keep the idle task around */
7671 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
7672 ret = GST_STATE_CHANGE_NO_PREROLL;
7674 case GST_STATE_CHANGE_PAUSED_TO_READY:
7675 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
7676 ret = GST_STATE_CHANGE_SUCCESS;
7678 case GST_STATE_CHANGE_READY_TO_NULL:
7679 gst_rtspsrc_stop (rtspsrc);
7680 ret = GST_STATE_CHANGE_SUCCESS;
7691 GST_DEBUG_OBJECT (rtspsrc, "start failed");
7692 return GST_STATE_CHANGE_FAILURE;
7697 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
7700 GstRTSPSrc *rtspsrc;
7702 rtspsrc = GST_RTSPSRC (element);
7704 if (GST_EVENT_IS_DOWNSTREAM (event)) {
7705 res = gst_rtspsrc_push_event (rtspsrc, event);
7707 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
7714 /*** GSTURIHANDLER INTERFACE *************************************************/
7717 gst_rtspsrc_uri_get_type (GType type)
7722 static const gchar *const *
7723 gst_rtspsrc_uri_get_protocols (GType type)
7725 static const gchar *protocols[] =
7726 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
7727 "rtsps", "rtspsu", "rtspst", "rtspsh", NULL
7734 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
7736 GstRTSPSrc *src = GST_RTSPSRC (handler);
7738 /* FIXME: make thread-safe */
7739 return g_strdup (src->conninfo.location);
7743 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
7748 GstRTSPUrl *newurl = NULL;
7749 GstSDPMessage *sdp = NULL;
7751 src = GST_RTSPSRC (handler);
7753 /* same URI, we're fine */
7754 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
7757 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
7758 if ((res = gst_sdp_message_new (&sdp) < 0))
7761 GST_DEBUG_OBJECT (src, "parsing SDP message");
7762 if ((res = gst_sdp_message_parse_uri (uri, sdp) < 0))
7766 GST_DEBUG_OBJECT (src, "parsing URI");
7767 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
7771 /* if worked, free previous and store new url object along with the original
7773 GST_DEBUG_OBJECT (src, "configuring URI");
7774 g_free (src->conninfo.location);
7775 src->conninfo.location = g_strdup (uri);
7776 gst_rtsp_url_free (src->conninfo.url);
7777 src->conninfo.url = newurl;
7778 g_free (src->conninfo.url_str);
7780 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
7782 src->conninfo.url_str = NULL;
7785 gst_sdp_message_free (src->sdp);
7787 src->from_sdp = sdp != NULL;
7789 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
7790 GST_DEBUG_OBJECT (src, "request uri is: %s",
7791 GST_STR_NULL (src->conninfo.url_str));
7798 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
7803 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", res);
7804 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7805 "Could not create SDP");
7810 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", res,
7811 GST_STR_NULL (uri));
7812 gst_sdp_message_free (sdp);
7813 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7819 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
7820 GST_STR_NULL (uri), res);
7821 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7822 "Invalid RTSP URI");
7828 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
7830 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
7832 iface->get_type = gst_rtspsrc_uri_get_type;
7833 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
7834 iface->get_uri = gst_rtspsrc_uri_get_uri;
7835 iface->set_uri = gst_rtspsrc_uri_set_uri;