2 * Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
3 * <2006> Lutz Mueller <lutz at topfrose dot de>
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
18 * Boston, MA 02110-1301, USA.
21 * Unless otherwise indicated, Source Code is licensed under MIT license.
22 * See further explanation attached in License Statement (distributed in the file
25 * Permission is hereby granted, free of charge, to any person obtaining a copy of
26 * this software and associated documentation files (the "Software"), to deal in
27 * the Software without restriction, including without limitation the rights to
28 * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
29 * of the Software, and to permit persons to whom the Software is furnished to do
30 * so, subject to the following conditions:
32 * The above copyright notice and this permission notice shall be included in all
33 * copies or substantial portions of the Software.
35 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
36 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
37 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
38 * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
39 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
40 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
44 * SECTION:element-rtspsrc
46 * Makes a connection to an RTSP server and read the data.
47 * rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
48 * RealMedia/Quicktime/Microsoft extensions.
50 * RTSP supports transport over TCP or UDP in unicast or multicast mode. By
51 * default rtspsrc will negotiate a connection in the following order:
52 * UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
53 * protocols can be controlled with the #GstRTSPSrc:protocols property.
55 * rtspsrc currently understands SDP as the format of the session description.
56 * For each stream listed in the SDP a new rtp_stream%d pad will be created
57 * with caps derived from the SDP media description. This is a caps of mime type
58 * "application/x-rtp" that can be connected to any available RTP depayloader
61 * rtspsrc will internally instantiate an RTP session manager element
62 * that will handle the RTCP messages to and from the server, jitter removal,
63 * packet reordering along with providing a clock for the pipeline.
64 * This feature is implemented using the gstrtpbin element.
66 * rtspsrc acts like a live source and will therefore only generate data in the
70 * <title>Example launch line</title>
72 * gst-launch-1.0 rtspsrc location=rtsp://some.server/url ! fakesink
73 * ]| Establish a connection to an RTSP server and send the raw RTP packets to a
77 * Last reviewed on 2006-08-18 (0.10.5)
86 #endif /* HAVE_UNISTD_H */
92 #include <gst/net/gstnet.h>
93 #include <gst/sdp/gstsdpmessage.h>
94 #include <gst/rtp/gstrtppayloads.h>
96 #include "gst/gst-i18n-plugin.h"
98 #include "gstrtspsrc.h"
100 GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
101 #define GST_CAT_DEFAULT (rtspsrc_debug)
103 static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
106 GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
108 /* templates used internally */
109 static GstStaticPadTemplate anysrctemplate =
110 GST_STATIC_PAD_TEMPLATE ("internalsrc_%u",
113 GST_STATIC_CAPS_ANY);
115 static GstStaticPadTemplate anysinktemplate =
116 GST_STATIC_PAD_TEMPLATE ("internalsink_%u",
119 GST_STATIC_CAPS_ANY);
123 SIGNAL_HANDLE_REQUEST,
125 SIGNAL_SELECT_STREAM,
130 enum _GstRtspSrcRtcpSyncMode
137 enum _GstRtspSrcBufferMode
146 #define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
148 gst_rtsp_src_buffer_mode_get_type (void)
150 static GType buffer_mode_type = 0;
151 static const GEnumValue buffer_modes[] = {
152 {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
153 {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
154 {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
155 {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
156 {BUFFER_MODE_SYNCED, "Synchronized sender and receiver clocks", "synced"},
160 if (!buffer_mode_type) {
162 g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
164 return buffer_mode_type;
167 #define DEFAULT_LOCATION NULL
168 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
169 #define DEFAULT_DEBUG FALSE
170 #define DEFAULT_RETRY 20
171 #define DEFAULT_TIMEOUT 5000000
172 #define DEFAULT_UDP_BUFFER_SIZE 0x80000
173 #define DEFAULT_TCP_TIMEOUT 20000000
174 #define DEFAULT_LATENCY_MS 2000
175 #define DEFAULT_DROP_ON_LATENCY FALSE
176 #define DEFAULT_CONNECTION_SPEED 0
177 #define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
178 #define DEFAULT_DO_RTCP TRUE
179 #define DEFAULT_DO_RTSP_KEEP_ALIVE TRUE
180 #define DEFAULT_PROXY NULL
181 #define DEFAULT_RTP_BLOCKSIZE 0
182 #define DEFAULT_USER_ID NULL
183 #define DEFAULT_USER_PW NULL
184 #define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO
185 #define DEFAULT_PORT_RANGE NULL
186 #define DEFAULT_SHORT_HEADER FALSE
187 #define DEFAULT_PROBATION 2
188 #define DEFAULT_UDP_RECONNECT TRUE
189 #define DEFAULT_MULTICAST_IFACE NULL
190 #define DEFAULT_NTP_SYNC FALSE
191 #define DEFAULT_USE_PIPELINE_CLOCK FALSE
192 #define DEFAULT_TLS_VALIDATION_FLAGS G_TLS_CERTIFICATE_VALIDATE_ALL
193 #define DEFAULT_TLS_DATABASE NULL
205 PROP_DROP_ON_LATENCY,
206 PROP_CONNECTION_SPEED,
209 PROP_DO_RTSP_KEEP_ALIVE,
218 PROP_UDP_BUFFER_SIZE,
222 PROP_MULTICAST_IFACE,
224 PROP_USE_PIPELINE_CLOCK,
226 PROP_TLS_VALIDATION_FLAGS,
231 #define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
233 gst_rtsp_nat_method_get_type (void)
235 static GType rtsp_nat_method_type = 0;
236 static const GEnumValue rtsp_nat_method[] = {
237 {GST_RTSP_NAT_NONE, "None", "none"},
238 {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
242 if (!rtsp_nat_method_type) {
243 rtsp_nat_method_type =
244 g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
246 return rtsp_nat_method_type;
249 static void gst_rtspsrc_finalize (GObject * object);
251 static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
252 const GValue * value, GParamSpec * pspec);
253 static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
254 GValue * value, GParamSpec * pspec);
256 static GstClock *gst_rtspsrc_provide_clock (GstElement * element);
258 static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
259 gpointer iface_data);
261 static void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
264 static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
265 static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
267 static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
269 static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
270 GstStateChange transition);
271 static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
272 static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
274 static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
275 GstRTSPMessage * response);
277 static gboolean gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd,
279 static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
280 GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
282 static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
283 static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
285 static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
286 static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
287 gboolean only_close);
289 static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
290 const gchar * uri, GError ** error);
291 static gchar *gst_rtspsrc_uri_get_uri (GstURIHandler * handler);
293 static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
294 static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
295 static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
296 GstRTSPStream * stream, GstEvent * event);
297 static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
298 static void gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush);
306 /* commands we send to out loop to notify it of events */
307 #define CMD_OPEN (1 << 0)
308 #define CMD_PLAY (1 << 1)
309 #define CMD_PAUSE (1 << 2)
310 #define CMD_CLOSE (1 << 3)
311 #define CMD_WAIT (1 << 4)
312 #define CMD_RECONNECT (1 << 5)
313 #define CMD_LOOP (1 << 6)
315 /* mask for all commands */
316 #define CMD_ALL ((CMD_LOOP << 1) - 1)
318 #define GST_ELEMENT_PROGRESS(el, type, code, text) \
320 gchar *__txt = _gst_element_error_printf text; \
321 gst_element_post_message (GST_ELEMENT_CAST (el), \
322 gst_message_new_progress (GST_OBJECT_CAST (el), \
323 GST_PROGRESS_TYPE_ ##type, code, __txt)); \
327 static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 };
329 #define gst_rtspsrc_parent_class parent_class
330 G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
331 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
334 default_select_stream (GstRTSPSrc * src, guint id, GstCaps * caps)
336 GST_DEBUG_OBJECT (src, "default handler");
341 select_stream_accum (GSignalInvocationHint * ihint,
342 GValue * return_accu, const GValue * handler_return, gpointer data)
346 myboolean = g_value_get_boolean (handler_return);
347 GST_DEBUG ("accum %d", myboolean);
348 g_value_set_boolean (return_accu, myboolean);
350 /* stop emission if FALSE */
355 gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
357 GObjectClass *gobject_class;
358 GstElementClass *gstelement_class;
359 GstBinClass *gstbin_class;
361 gobject_class = (GObjectClass *) klass;
362 gstelement_class = (GstElementClass *) klass;
363 gstbin_class = (GstBinClass *) klass;
365 GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
367 gobject_class->set_property = gst_rtspsrc_set_property;
368 gobject_class->get_property = gst_rtspsrc_get_property;
370 gobject_class->finalize = gst_rtspsrc_finalize;
372 g_object_class_install_property (gobject_class, PROP_LOCATION,
373 g_param_spec_string ("location", "RTSP Location",
374 "Location of the RTSP url to read",
375 DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
377 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
378 g_param_spec_flags ("protocols", "Protocols",
379 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
380 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
382 g_object_class_install_property (gobject_class, PROP_DEBUG,
383 g_param_spec_boolean ("debug", "Debug",
384 "Dump request and response messages to stdout",
385 DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
387 g_object_class_install_property (gobject_class, PROP_RETRY,
388 g_param_spec_uint ("retry", "Retry",
389 "Max number of retries when allocating RTP ports.",
390 0, G_MAXUINT16, DEFAULT_RETRY,
391 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
393 g_object_class_install_property (gobject_class, PROP_TIMEOUT,
394 g_param_spec_uint64 ("timeout", "Timeout",
395 "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
396 0, G_MAXUINT64, DEFAULT_TIMEOUT,
397 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
399 g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
400 g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
401 "Fail after timeout microseconds on TCP connections (0 = disabled)",
402 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
403 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
405 g_object_class_install_property (gobject_class, PROP_LATENCY,
406 g_param_spec_uint ("latency", "Buffer latency in ms",
407 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
408 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
410 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
411 g_param_spec_boolean ("drop-on-latency",
412 "Drop buffers when maximum latency is reached",
413 "Tells the jitterbuffer to never exceed the given latency in size",
414 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
416 g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
417 g_param_spec_uint64 ("connection-speed", "Connection Speed",
418 "Network connection speed in kbps (0 = unknown)",
419 0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
420 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
422 g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
423 g_param_spec_enum ("nat-method", "NAT Method",
424 "Method to use for traversing firewalls and NAT",
425 GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
426 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
429 * GstRTSPSrc:do-rtcp:
431 * Enable RTCP support. Some old server don't like RTCP and then this property
432 * needs to be set to FALSE.
434 g_object_class_install_property (gobject_class, PROP_DO_RTCP,
435 g_param_spec_boolean ("do-rtcp", "Do RTCP",
436 "Send RTCP packets, disable for old incompatible server.",
437 DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
440 * GstRTSPSrc:do-rtsp-keep-alive:
442 * Enable RTSP keep alive support. Some old server don't like RTSP
443 * keep alive and then this property needs to be set to FALSE.
445 g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
446 g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
447 "Send RTSP keep alive packets, disable for old incompatible server.",
448 DEFAULT_DO_RTSP_KEEP_ALIVE,
449 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
454 * Set the proxy parameters. This has to be a string of the format
455 * [http://][user:passwd@]host[:port].
457 g_object_class_install_property (gobject_class, PROP_PROXY,
458 g_param_spec_string ("proxy", "Proxy",
459 "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
460 DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
462 * GstRTSPSrc:proxy-id:
464 * Sets the proxy URI user id for authentication. If the URI set via the
465 * "proxy" property contains a user-id already, that will take precedence.
469 g_object_class_install_property (gobject_class, PROP_PROXY_ID,
470 g_param_spec_string ("proxy-id", "proxy-id",
471 "HTTP proxy URI user id for authentication", "",
472 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
474 * GstRTSPSrc:proxy-pw:
476 * Sets the proxy URI password for authentication. If the URI set via the
477 * "proxy" property contains a password already, that will take precedence.
481 g_object_class_install_property (gobject_class, PROP_PROXY_PW,
482 g_param_spec_string ("proxy-pw", "proxy-pw",
483 "HTTP proxy URI user password for authentication", "",
484 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
487 * GstRTSPSrc:rtp-blocksize:
489 * RTP package size to suggest to server.
491 g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
492 g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
493 "RTP package size to suggest to server (0 = disabled)",
494 0, 65536, DEFAULT_RTP_BLOCKSIZE,
495 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
497 g_object_class_install_property (gobject_class,
499 g_param_spec_string ("user-id", "user-id",
500 "RTSP location URI user id for authentication", DEFAULT_USER_ID,
501 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
502 g_object_class_install_property (gobject_class, PROP_USER_PW,
503 g_param_spec_string ("user-pw", "user-pw",
504 "RTSP location URI user password for authentication", DEFAULT_USER_PW,
505 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
508 * GstRTSPSrc:buffer-mode:
510 * Control the buffering and timestamping mode used by the jitterbuffer.
512 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
513 g_param_spec_enum ("buffer-mode", "Buffer Mode",
514 "Control the buffering algorithm in use",
515 GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
516 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
519 * GstRTSPSrc:port-range:
521 * Configure the client port numbers that can be used to recieve RTP and
524 g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
525 g_param_spec_string ("port-range", "Port range",
526 "Client port range that can be used to receive RTP and RTCP data, "
527 "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
528 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
531 * GstRTSPSrc:udp-buffer-size:
533 * Size of the kernel UDP receive buffer in bytes.
535 g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
536 g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
537 "Size of the kernel UDP receive buffer in bytes, 0=default",
538 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
539 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
542 * GstRTSPSrc:short-header:
544 * Only send the basic RTSP headers for broken encoders.
546 g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
547 g_param_spec_boolean ("short-header", "Short Header",
548 "Only send the basic RTSP headers for broken encoders",
549 DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
551 g_object_class_install_property (gobject_class, PROP_PROBATION,
552 g_param_spec_uint ("probation", "Number of probations",
553 "Consecutive packet sequence numbers to accept the source",
554 0, G_MAXUINT, DEFAULT_PROBATION,
555 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
557 g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
558 g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
559 "Reconnect to the server if RTSP connection is closed when doing UDP",
560 DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
562 g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
563 g_param_spec_string ("multicast-iface", "Multicast Interface",
564 "The network interface on which to join the multicast group",
565 DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
567 g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
568 g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
569 "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
570 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
572 g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
573 g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
574 "Use the pipeline running-time to set the NTP time in the RTCP SR messages",
575 DEFAULT_USE_PIPELINE_CLOCK,
576 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
578 g_object_class_install_property (gobject_class, PROP_SDES,
579 g_param_spec_boxed ("sdes", "SDES",
580 "The SDES items of this session",
581 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
584 * GstRTSPSrc::tls-validation-flags:
586 * TLS certificate validation flags used to validate server
591 g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
592 g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
593 "TLS certificate validation flags used to validate the server certificate",
594 G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS,
595 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
598 * GstRTSPSrc::tls-database:
600 * TLS database with anchor certificate authorities used to validate
601 * the server certificate.
605 g_object_class_install_property (gobject_class, PROP_TLS_DATABASE,
606 g_param_spec_object ("tls-database", "TLS database",
607 "TLS database with anchor certificate authorities used to validate the server certificate",
608 G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
611 * GstRTSPSrc::handle-request:
612 * @rtspsrc: a #GstRTSPSrc
613 * @request: a #GstRTSPMessage
614 * @response: a #GstRTSPMessage
616 * Handle a server request in @request and prepare @response.
618 * This signal is called from the streaming thread, you should therefore not
619 * do any state changes on @rtspsrc because this might deadlock. If you want
620 * to modify the state as a result of this signal, post a
621 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
626 gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST] =
627 g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
628 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2,
629 G_TYPE_POINTER, G_TYPE_POINTER);
632 * GstRTSPSrc::on-sdp:
633 * @rtspsrc: a #GstRTSPSrc
634 * @sdp: a #GstSDPMessage
636 * Emited when the client has retrieved the SDP and before it configures the
637 * streams in the SDP. @sdp can be inspected and modified.
639 * This signal is called from the streaming thread, you should therefore not
640 * do any state changes on @rtspsrc because this might deadlock. If you want
641 * to modify the state as a result of this signal, post a
642 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
647 gst_rtspsrc_signals[SIGNAL_ON_SDP] =
648 g_signal_new ("on-sdp", G_TYPE_FROM_CLASS (klass), 0,
649 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
650 GST_TYPE_SDP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
653 * GstRTSPSrc::select-stream:
654 * @rtspsrc: a #GstRTSPSrc
655 * @num: the stream number
656 * @caps: the stream caps
658 * Emited before the client decides to configure the stream @num with
661 * Returns: %TRUE when the stream should be selected, %FALSE when the stream
666 gst_rtspsrc_signals[SIGNAL_SELECT_STREAM] =
667 g_signal_new_class_handler ("select-stream", G_TYPE_FROM_CLASS (klass),
668 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP,
669 (GCallback) default_select_stream, select_stream_accum, NULL,
670 g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2, G_TYPE_UINT,
673 * GstRTSPSrc::new-manager:
674 * @rtspsrc: a #GstRTSPSrc
675 * @manager: a #GstElement
677 * Emited after a new manager (like rtpbin) was created and the default
678 * properties were configured.
682 gst_rtspsrc_signals[SIGNAL_NEW_MANAGER] =
683 g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass),
684 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
685 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
687 gstelement_class->send_event = gst_rtspsrc_send_event;
688 gstelement_class->provide_clock = gst_rtspsrc_provide_clock;
689 gstelement_class->change_state = gst_rtspsrc_change_state;
691 gst_element_class_add_pad_template (gstelement_class,
692 gst_static_pad_template_get (&rtptemplate));
694 gst_element_class_set_static_metadata (gstelement_class,
695 "RTSP packet receiver", "Source/Network",
696 "Receive data over the network via RTSP (RFC 2326)",
697 "Wim Taymans <wim@fluendo.com>, "
698 "Thijs Vermeir <thijs.vermeir@barco.com>, "
699 "Lutz Mueller <lutz@topfrose.de>");
701 gstbin_class->handle_message = gst_rtspsrc_handle_message;
703 gst_rtsp_ext_list_init ();
707 gst_rtspsrc_init (GstRTSPSrc * src)
709 src->conninfo.location = g_strdup (DEFAULT_LOCATION);
710 src->protocols = DEFAULT_PROTOCOLS;
711 src->debug = DEFAULT_DEBUG;
712 src->retry = DEFAULT_RETRY;
713 src->udp_timeout = DEFAULT_TIMEOUT;
714 gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
715 src->latency = DEFAULT_LATENCY_MS;
716 src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
717 src->connection_speed = DEFAULT_CONNECTION_SPEED;
718 src->nat_method = DEFAULT_NAT_METHOD;
719 src->do_rtcp = DEFAULT_DO_RTCP;
720 src->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
721 gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
722 src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
723 src->user_id = g_strdup (DEFAULT_USER_ID);
724 src->user_pw = g_strdup (DEFAULT_USER_PW);
725 src->buffer_mode = DEFAULT_BUFFER_MODE;
726 src->client_port_range.min = 0;
727 src->client_port_range.max = 0;
728 src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
729 src->short_header = DEFAULT_SHORT_HEADER;
730 src->probation = DEFAULT_PROBATION;
731 src->udp_reconnect = DEFAULT_UDP_RECONNECT;
732 src->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
733 src->ntp_sync = DEFAULT_NTP_SYNC;
734 src->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
736 src->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
737 src->tls_database = DEFAULT_TLS_DATABASE;
739 /* get a list of all extensions */
740 src->extensions = gst_rtsp_ext_list_get ();
742 /* connect to send signal */
743 gst_rtsp_ext_list_connect (src->extensions, "send",
744 (GCallback) gst_rtspsrc_send_cb, src);
746 /* protects the streaming thread in interleaved mode or the polling
747 * thread in UDP mode. */
748 g_rec_mutex_init (&src->stream_rec_lock);
750 /* protects our state changes from multiple invocations */
751 g_rec_mutex_init (&src->state_rec_lock);
753 src->state = GST_RTSP_STATE_INVALID;
755 GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
759 gst_rtspsrc_finalize (GObject * object)
763 rtspsrc = GST_RTSPSRC (object);
765 gst_rtsp_ext_list_free (rtspsrc->extensions);
766 g_free (rtspsrc->conninfo.location);
767 gst_rtsp_url_free (rtspsrc->conninfo.url);
768 g_free (rtspsrc->conninfo.url_str);
769 g_free (rtspsrc->user_id);
770 g_free (rtspsrc->user_pw);
771 g_free (rtspsrc->multi_iface);
774 gst_sdp_message_free (rtspsrc->sdp);
777 if (rtspsrc->provided_clock)
778 gst_object_unref (rtspsrc->provided_clock);
781 gst_structure_free (rtspsrc->sdes);
783 if (rtspsrc->tls_database)
784 g_object_unref (rtspsrc->tls_database);
787 g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
788 g_rec_mutex_clear (&rtspsrc->state_rec_lock);
790 G_OBJECT_CLASS (parent_class)->finalize (object);
794 gst_rtspsrc_provide_clock (GstElement * element)
796 GstRTSPSrc *src = GST_RTSPSRC (element);
799 if ((clock = src->provided_clock) != NULL)
800 gst_object_ref (clock);
805 /* a proxy string of the format [user:passwd@]host[:port] */
807 gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
811 g_free (rtsp->proxy_user);
812 rtsp->proxy_user = NULL;
813 g_free (rtsp->proxy_passwd);
814 rtsp->proxy_passwd = NULL;
815 g_free (rtsp->proxy_host);
816 rtsp->proxy_host = NULL;
817 rtsp->proxy_port = 0;
824 /* we allow http:// in front but ignore it */
825 if (g_str_has_prefix (p, "http://"))
828 at = strchr (p, '@');
830 /* look for user:passwd */
831 col = strchr (proxy, ':');
832 if (col == NULL || col > at)
835 rtsp->proxy_user = g_strndup (p, col - p);
837 rtsp->proxy_passwd = g_strndup (col, at - col);
842 if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
843 rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
844 if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
845 rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
846 if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
847 GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
848 GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
851 col = strchr (p, ':');
854 /* everything before the colon is the hostname */
855 rtsp->proxy_host = g_strndup (p, col - p);
857 rtsp->proxy_port = strtoul (p, (char **) &p, 10);
859 rtsp->proxy_host = g_strdup (p);
860 rtsp->proxy_port = 8080;
866 gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
868 rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
869 rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
872 rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
874 rtspsrc->ptcp_timeout = NULL;
878 gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
883 rtspsrc = GST_RTSPSRC (object);
887 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
888 g_value_get_string (value), NULL);
891 rtspsrc->protocols = g_value_get_flags (value);
894 rtspsrc->debug = g_value_get_boolean (value);
897 rtspsrc->retry = g_value_get_uint (value);
900 rtspsrc->udp_timeout = g_value_get_uint64 (value);
902 case PROP_TCP_TIMEOUT:
903 gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
906 rtspsrc->latency = g_value_get_uint (value);
908 case PROP_DROP_ON_LATENCY:
909 rtspsrc->drop_on_latency = g_value_get_boolean (value);
911 case PROP_CONNECTION_SPEED:
912 rtspsrc->connection_speed = g_value_get_uint64 (value);
914 case PROP_NAT_METHOD:
915 rtspsrc->nat_method = g_value_get_enum (value);
918 rtspsrc->do_rtcp = g_value_get_boolean (value);
920 case PROP_DO_RTSP_KEEP_ALIVE:
921 rtspsrc->do_rtsp_keep_alive = g_value_get_boolean (value);
924 gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
927 if (rtspsrc->prop_proxy_id)
928 g_free (rtspsrc->prop_proxy_id);
929 rtspsrc->prop_proxy_id = g_value_dup_string (value);
932 if (rtspsrc->prop_proxy_pw)
933 g_free (rtspsrc->prop_proxy_pw);
934 rtspsrc->prop_proxy_pw = g_value_dup_string (value);
936 case PROP_RTP_BLOCKSIZE:
937 rtspsrc->rtp_blocksize = g_value_get_uint (value);
940 if (rtspsrc->user_id)
941 g_free (rtspsrc->user_id);
942 rtspsrc->user_id = g_value_dup_string (value);
945 if (rtspsrc->user_pw)
946 g_free (rtspsrc->user_pw);
947 rtspsrc->user_pw = g_value_dup_string (value);
949 case PROP_BUFFER_MODE:
950 rtspsrc->buffer_mode = g_value_get_enum (value);
952 case PROP_PORT_RANGE:
956 str = g_value_get_string (value);
958 sscanf (str, "%u-%u",
959 &rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
961 rtspsrc->client_port_range.min = 0;
962 rtspsrc->client_port_range.max = 0;
966 case PROP_UDP_BUFFER_SIZE:
967 rtspsrc->udp_buffer_size = g_value_get_int (value);
969 case PROP_SHORT_HEADER:
970 rtspsrc->short_header = g_value_get_boolean (value);
973 rtspsrc->probation = g_value_get_uint (value);
975 case PROP_UDP_RECONNECT:
976 rtspsrc->udp_reconnect = g_value_get_boolean (value);
978 case PROP_MULTICAST_IFACE:
979 g_free (rtspsrc->multi_iface);
981 if (g_value_get_string (value) == NULL)
982 rtspsrc->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
984 rtspsrc->multi_iface = g_value_dup_string (value);
987 rtspsrc->ntp_sync = g_value_get_boolean (value);
989 case PROP_USE_PIPELINE_CLOCK:
990 rtspsrc->use_pipeline_clock = g_value_get_boolean (value);
993 rtspsrc->sdes = g_value_dup_boxed (value);
995 case PROP_TLS_VALIDATION_FLAGS:
996 rtspsrc->tls_validation_flags = g_value_get_flags (value);
998 case PROP_TLS_DATABASE:
999 g_clear_object (&rtspsrc->tls_database);
1000 rtspsrc->tls_database = g_value_dup_object (value);
1003 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1009 gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
1012 GstRTSPSrc *rtspsrc;
1014 rtspsrc = GST_RTSPSRC (object);
1018 g_value_set_string (value, rtspsrc->conninfo.location);
1020 case PROP_PROTOCOLS:
1021 g_value_set_flags (value, rtspsrc->protocols);
1024 g_value_set_boolean (value, rtspsrc->debug);
1027 g_value_set_uint (value, rtspsrc->retry);
1030 g_value_set_uint64 (value, rtspsrc->udp_timeout);
1032 case PROP_TCP_TIMEOUT:
1036 timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
1037 rtspsrc->tcp_timeout.tv_usec;
1038 g_value_set_uint64 (value, timeout);
1042 g_value_set_uint (value, rtspsrc->latency);
1044 case PROP_DROP_ON_LATENCY:
1045 g_value_set_boolean (value, rtspsrc->drop_on_latency);
1047 case PROP_CONNECTION_SPEED:
1048 g_value_set_uint64 (value, rtspsrc->connection_speed);
1050 case PROP_NAT_METHOD:
1051 g_value_set_enum (value, rtspsrc->nat_method);
1054 g_value_set_boolean (value, rtspsrc->do_rtcp);
1056 case PROP_DO_RTSP_KEEP_ALIVE:
1057 g_value_set_boolean (value, rtspsrc->do_rtsp_keep_alive);
1063 if (rtspsrc->proxy_host) {
1065 g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
1069 g_value_take_string (value, str);
1073 g_value_set_string (value, rtspsrc->prop_proxy_id);
1076 g_value_set_string (value, rtspsrc->prop_proxy_pw);
1078 case PROP_RTP_BLOCKSIZE:
1079 g_value_set_uint (value, rtspsrc->rtp_blocksize);
1082 g_value_set_string (value, rtspsrc->user_id);
1085 g_value_set_string (value, rtspsrc->user_pw);
1087 case PROP_BUFFER_MODE:
1088 g_value_set_enum (value, rtspsrc->buffer_mode);
1090 case PROP_PORT_RANGE:
1094 if (rtspsrc->client_port_range.min != 0) {
1095 str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
1096 rtspsrc->client_port_range.max);
1100 g_value_take_string (value, str);
1103 case PROP_UDP_BUFFER_SIZE:
1104 g_value_set_int (value, rtspsrc->udp_buffer_size);
1106 case PROP_SHORT_HEADER:
1107 g_value_set_boolean (value, rtspsrc->short_header);
1109 case PROP_PROBATION:
1110 g_value_set_uint (value, rtspsrc->probation);
1112 case PROP_UDP_RECONNECT:
1113 g_value_set_boolean (value, rtspsrc->udp_reconnect);
1115 case PROP_MULTICAST_IFACE:
1116 g_value_set_string (value, rtspsrc->multi_iface);
1119 g_value_set_boolean (value, rtspsrc->ntp_sync);
1121 case PROP_USE_PIPELINE_CLOCK:
1122 g_value_set_boolean (value, rtspsrc->use_pipeline_clock);
1125 g_value_set_boxed (value, rtspsrc->sdes);
1127 case PROP_TLS_VALIDATION_FLAGS:
1128 g_value_set_flags (value, rtspsrc->tls_validation_flags);
1130 case PROP_TLS_DATABASE:
1131 g_value_set_object (value, rtspsrc->tls_database);
1134 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1140 find_stream_by_id (GstRTSPStream * stream, gint * id)
1142 if (stream->id == *id)
1149 find_stream_by_channel (GstRTSPStream * stream, gint * channel)
1151 if (stream->channel[0] == *channel || stream->channel[1] == *channel)
1158 find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
1160 GstElement *src = (GstElement *) a;
1162 if (stream->udpsrc[0] == src)
1164 if (stream->udpsrc[1] == src)
1171 find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
1173 if (stream->conninfo.location) {
1174 /* check qualified setup_url */
1175 if (!strcmp (stream->conninfo.location, (gchar *) a))
1178 if (stream->control_url) {
1179 /* check original control_url */
1180 if (!strcmp (stream->control_url, (gchar *) a))
1183 /* check if qualified setup_url ends with string */
1184 if (g_str_has_suffix (stream->control_url, (gchar *) a))
1191 static GstRTSPStream *
1192 find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
1196 /* find and get stream */
1197 if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
1198 return (GstRTSPStream *) lstream->data;
1203 static const GstSDPBandwidth *
1204 gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1205 const GstSDPMedia * media, const gchar * type)
1209 /* first look in the media specific section */
1210 len = gst_sdp_media_bandwidths_len (media);
1211 for (i = 0; i < len; i++) {
1212 const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
1214 if (strcmp (bw->bwtype, type) == 0)
1217 /* then look in the message specific section */
1218 len = gst_sdp_message_bandwidths_len (sdp);
1219 for (i = 0; i < len; i++) {
1220 const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
1222 if (strcmp (bw->bwtype, type) == 0)
1229 gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1230 const GstSDPMedia * media, GstRTSPStream * stream)
1232 const GstSDPBandwidth *bw;
1234 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
1235 stream->as_bandwidth = bw->bandwidth;
1237 stream->as_bandwidth = -1;
1239 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
1240 stream->rr_bandwidth = bw->bandwidth;
1242 stream->rr_bandwidth = -1;
1244 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
1245 stream->rs_bandwidth = bw->bandwidth;
1247 stream->rs_bandwidth = -1;
1251 gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
1252 const GstSDPConnection * conn)
1254 if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
1257 if (conn->addrtype == NULL)
1260 /* check for IPV6 */
1261 if (strcmp (conn->addrtype, "IP4") == 0)
1262 stream->is_ipv6 = FALSE;
1263 else if (strcmp (conn->addrtype, "IP6") == 0)
1264 stream->is_ipv6 = TRUE;
1269 g_free (stream->destination);
1270 stream->destination = g_strdup (conn->address);
1272 /* check for multicast */
1273 stream->is_multicast =
1274 gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
1276 stream->ttl = conn->ttl;
1279 /* Go over the connections for a stream.
1280 * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
1282 * - If we are dealing with a localhost address, we disable multicast
1285 gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
1286 const GstSDPMedia * media, GstRTSPStream * stream)
1288 const GstSDPConnection *conn;
1291 /* first look in the media specific section */
1292 len = gst_sdp_media_connections_len (media);
1293 for (i = 0; i < len; i++) {
1294 conn = gst_sdp_media_get_connection (media, i);
1296 gst_rtspsrc_do_stream_connection (src, stream, conn);
1298 /* then look in the message specific section */
1299 if ((conn = gst_sdp_message_get_connection (sdp))) {
1300 gst_rtspsrc_do_stream_connection (src, stream, conn);
1305 /* m=<media> <UDP port> RTP/AVP <payload>
1308 gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
1309 const GstSDPMedia * media, GstRTSPStream * stream)
1315 proto = gst_sdp_media_get_proto (media);
1319 if (g_str_equal (proto, "RTP/AVP"))
1320 stream->profile = GST_RTSP_PROFILE_AVP;
1321 else if (g_str_equal (proto, "RTP/SAVP"))
1322 stream->profile = GST_RTSP_PROFILE_SAVP;
1323 else if (g_str_equal (proto, "RTP/AVPF"))
1324 stream->profile = GST_RTSP_PROFILE_AVPF;
1325 else if (g_str_equal (proto, "RTP/SAVPF"))
1326 stream->profile = GST_RTSP_PROFILE_SAVPF;
1330 len = gst_sdp_media_formats_len (media);
1331 for (i = 0; i < len; i++) {
1338 pt = atoi (gst_sdp_media_get_format (media, i));
1340 GST_DEBUG_OBJECT (src, " looking at %d pt: %d", i, pt);
1343 caps = gst_rtspsrc_media_to_caps (pt, media);
1345 GST_WARNING_OBJECT (src, " skipping pt %d without caps", pt);
1349 /* do some tweaks */
1350 s = gst_caps_get_structure (caps, 0);
1351 if ((enc = gst_structure_get_string (s, "encoding-name"))) {
1352 stream->is_real = (strstr (enc, "-REAL") != NULL);
1353 if (strcmp (enc, "X-ASF-PF") == 0)
1354 stream->container = TRUE;
1356 GST_DEBUG ("mapping sdp session level attributes to caps");
1357 gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, caps);
1358 GST_DEBUG ("mapping sdp media level attributes to caps");
1359 gst_rtspsrc_sdp_attributes_to_caps (media->attributes, caps);
1361 /* the first pt will be the default */
1362 if (stream->ptmap->len == 0)
1363 stream->default_pt = pt;
1367 g_array_append_val (stream->ptmap, item);
1373 GST_ERROR_OBJECT (src, "can't find proto in media");
1378 GST_ERROR_OBJECT (src, "unknown proto in media %s", proto);
1383 static const gchar *
1384 get_aggregate_control (GstRTSPSrc * src)
1389 base = src->control;
1390 else if (src->content_base)
1391 base = src->content_base;
1392 else if (src->conninfo.url_str)
1393 base = src->conninfo.url_str;
1401 clear_ptmap_item (PtMapItem * item)
1404 gst_caps_unref (item->caps);
1407 static GstRTSPStream *
1408 gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
1410 GstRTSPStream *stream;
1411 const gchar *control_url;
1412 const GstSDPMedia *media;
1414 /* get media, should not return NULL */
1415 media = gst_sdp_message_get_media (sdp, idx);
1419 stream = g_new0 (GstRTSPStream, 1);
1420 stream->parent = src;
1421 /* we mark the pad as not linked, we will mark it as OK when we add the pad to
1423 stream->last_ret = GST_FLOW_NOT_LINKED;
1424 stream->added = FALSE;
1425 stream->setup = FALSE;
1426 stream->skipped = FALSE;
1428 stream->eos = FALSE;
1429 stream->discont = TRUE;
1430 stream->seqbase = -1;
1431 stream->timebase = -1;
1432 stream->profile = GST_RTSP_PROFILE_AVP;
1433 stream->ptmap = g_array_new (FALSE, FALSE, sizeof (PtMapItem));
1434 g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);
1436 /* collect bandwidth information for this steam. FIXME, configure in the RTP
1437 * session manager to scale RTCP. */
1438 gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
1440 /* collect connection info */
1441 gst_rtspsrc_collect_connections (src, sdp, media, stream);
1443 /* make the payload type map */
1444 gst_rtspsrc_collect_payloads (src, sdp, media, stream);
1446 /* collect port number */
1447 stream->port = gst_sdp_media_get_port (media);
1449 /* get control url to construct the setup url. The setup url is used to
1450 * configure the transport of the stream and is used to identity the stream in
1451 * the RTP-Info header field returned from PLAY. */
1452 control_url = gst_sdp_media_get_attribute_val (media, "control");
1453 if (control_url == NULL)
1454 control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
1456 GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
1457 GST_DEBUG_OBJECT (src, " port: %d", stream->port);
1458 GST_DEBUG_OBJECT (src, " container: %d", stream->container);
1459 GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
1461 if (control_url != NULL) {
1462 stream->control_url = g_strdup (control_url);
1463 /* Build a fully qualified url using the content_base if any or by prefixing
1464 * the original request.
1465 * If the control_url starts with a '/' or a non rtsp: protocol we will most
1466 * likely build a URL that the server will fail to understand, this is ok,
1467 * we will fail then. */
1468 if (g_str_has_prefix (control_url, "rtsp://"))
1469 stream->conninfo.location = g_strdup (control_url);
1474 if (g_strcmp0 (control_url, "*") == 0)
1477 base = get_aggregate_control (src);
1479 /* check if the base ends or control starts with / */
1480 has_slash = g_str_has_prefix (control_url, "/");
1481 has_slash = has_slash || g_str_has_suffix (base, "/");
1483 /* concatenate the two strings, insert / when not present */
1484 stream->conninfo.location =
1485 g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
1488 GST_DEBUG_OBJECT (src, " setup: %s",
1489 GST_STR_NULL (stream->conninfo.location));
1491 /* we keep track of all streams */
1492 src->streams = g_list_append (src->streams, stream);
1500 gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
1504 GST_DEBUG_OBJECT (src, "free stream %p", stream);
1506 g_array_free (stream->ptmap, TRUE);
1508 g_free (stream->destination);
1509 g_free (stream->control_url);
1510 g_free (stream->conninfo.location);
1512 for (i = 0; i < 2; i++) {
1513 if (stream->udpsrc[i]) {
1514 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
1515 gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
1516 gst_object_unref (stream->udpsrc[i]);
1517 stream->udpsrc[i] = NULL;
1519 if (stream->channelpad[i]) {
1520 gst_object_unref (stream->channelpad[i]);
1521 stream->channelpad[i] = NULL;
1523 if (stream->udpsink[i]) {
1524 gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
1525 gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
1526 gst_object_unref (stream->udpsink[i]);
1527 stream->udpsink[i] = NULL;
1530 if (stream->fakesrc) {
1531 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
1532 gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
1533 gst_object_unref (stream->fakesrc);
1534 stream->fakesrc = NULL;
1536 if (stream->srcpad) {
1537 gst_pad_set_active (stream->srcpad, FALSE);
1538 if (stream->added) {
1539 gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
1540 stream->added = FALSE;
1542 stream->srcpad = NULL;
1544 if (stream->rtcppad) {
1545 gst_object_unref (stream->rtcppad);
1546 stream->rtcppad = NULL;
1548 if (stream->session) {
1549 g_object_unref (stream->session);
1550 stream->session = NULL;
1556 gst_rtspsrc_cleanup (GstRTSPSrc * src)
1560 GST_DEBUG_OBJECT (src, "cleanup");
1562 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1563 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1565 gst_rtspsrc_stream_free (src, stream);
1567 g_list_free (src->streams);
1568 src->streams = NULL;
1570 if (src->manager_sig_id) {
1571 g_signal_handler_disconnect (src->manager, src->manager_sig_id);
1572 src->manager_sig_id = 0;
1574 gst_element_set_state (src->manager, GST_STATE_NULL);
1575 gst_bin_remove (GST_BIN_CAST (src), src->manager);
1576 src->manager = NULL;
1579 gst_structure_free (src->props);
1582 g_free (src->content_base);
1583 src->content_base = NULL;
1585 g_free (src->control);
1586 src->control = NULL;
1589 gst_rtsp_range_free (src->range);
1592 /* don't clear the SDP when it was used in the url */
1593 if (src->sdp && !src->from_sdp) {
1594 gst_sdp_message_free (src->sdp);
1597 if (src->start_segment) {
1598 gst_event_unref (src->start_segment);
1599 src->start_segment = NULL;
1601 if (src->provided_clock) {
1602 gst_object_unref (src->provided_clock);
1603 src->provided_clock = NULL;
1607 #define PARSE_INT(p, del, res) \
1610 p = strstr (p, del); \
1620 #define PARSE_STRING(p, del, res) \
1623 p = strstr (p, del); \
1635 #define SKIP_SPACES(p) \
1636 while (*p && g_ascii_isspace (*p)) \
1641 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1644 gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
1645 gint * rate, gchar ** params)
1649 p = (gchar *) rtpmap;
1651 PARSE_INT (p, " ", *payload);
1659 PARSE_STRING (p, "/", *name);
1660 if (*name == NULL) {
1661 GST_DEBUG ("no rate, name %s", p);
1662 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
1663 * streams seem to omit the rate. */
1670 p = strstr (p, "/");
1688 * Mapping SDP attributes to caps
1690 * prepend 'a-' to IANA registered sdp attributes names
1691 * (ie: not prefixed with 'x-') in order to avoid
1692 * collision with gstreamer standard caps properties names
1695 gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
1697 if (attributes->len > 0) {
1701 s = gst_caps_get_structure (caps, 0);
1703 for (i = 0; i < attributes->len; i++) {
1704 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
1705 gchar *tofree, *key;
1709 /* skip some of the attribute we already handle */
1710 if (!strcmp (key, "fmtp"))
1712 if (!strcmp (key, "rtpmap"))
1714 if (!strcmp (key, "control"))
1716 if (!strcmp (key, "range"))
1719 /* string must be valid UTF8 */
1720 if (!g_utf8_validate (attr->value, -1, NULL))
1723 if (!g_str_has_prefix (key, "x-"))
1724 tofree = key = g_strdup_printf ("a-%s", key);
1728 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
1729 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
1735 static const gchar *
1736 rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
1745 if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
1748 if (sscanf (attr, "%d ", &val) != 1)
1758 * Mapping of caps to and from SDP fields:
1760 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1761 * a=fmtp:<payload> <param>[=<value>];...
1764 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
1767 const gchar *rtpmap;
1771 gchar *params = NULL;
1777 /* get and parse rtpmap */
1778 rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
1781 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
1783 g_warning ("error parsing rtpmap, ignoring");
1787 /* dynamic payloads need rtpmap or we fail */
1788 if (rtpmap == NULL && pt >= 96)
1791 /* check if we have a rate, if not, we need to look up the rate from the
1792 * default rates based on the payload types. */
1794 const GstRTPPayloadInfo *info;
1796 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
1797 /* dynamic types, use media and encoding_name */
1798 tmp = g_ascii_strdown (media->media, -1);
1799 info = gst_rtp_payload_info_for_name (tmp, name);
1802 /* static types, use payload type */
1803 info = gst_rtp_payload_info_for_pt (pt);
1807 if ((rate = info->clock_rate) == 0)
1810 /* we fail if we cannot find one */
1815 tmp = g_ascii_strdown (media->media, -1);
1816 caps = gst_caps_new_simple ("application/x-unknown",
1817 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
1819 s = gst_caps_get_structure (caps, 0);
1821 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
1823 /* encoding name must be upper case */
1825 tmp = g_ascii_strup (name, -1);
1826 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
1830 /* params must be lower case */
1831 if (params != NULL) {
1832 tmp = g_ascii_strdown (params, -1);
1833 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
1837 /* parse optional fmtp: field */
1838 if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
1844 /* p is now of the format <payload> <param>[=<value>];... */
1845 PARSE_INT (p, " ", payload);
1846 if (payload != -1 && payload == pt) {
1850 /* <param>[=<value>] are separated with ';' */
1851 pairs = g_strsplit (p, ";", 0);
1852 for (i = 0; pairs[i]; i++) {
1854 const gchar *val, *key;
1856 /* the key may not have a '=', the value can have other '='s */
1857 valpos = strstr (pairs[i], "=");
1859 /* we have a '=' and thus a value, remove the '=' with \0 */
1861 /* value is everything between '=' and ';'. We split the pairs at ;
1862 * boundaries so we can take the remainder of the value. Some servers
1863 * put spaces around the value which we strip off here. Alternatively
1864 * we could strip those spaces in the depayloaders should these spaces
1865 * actually carry any meaning in the future. */
1866 val = g_strstrip (valpos + 1);
1868 /* simple <param>;.. is translated into <param>=1;... */
1871 /* strip the key of spaces, convert key to lowercase but not the value. */
1872 key = g_strstrip (pairs[i]);
1873 if (strlen (key) > 1) {
1874 tmp = g_ascii_strdown (key, -1);
1875 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
1887 g_warning ("rtpmap type not given for dynamic payload %d", pt);
1892 g_warning ("rate unknown for payload type %d", pt);
1898 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
1899 gint * rtpport, gint * rtcpport)
1902 GstStateChangeReturn ret;
1903 GstElement *udpsrc0, *udpsrc1;
1904 gint tmp_rtp, tmp_rtcp;
1908 src = stream->parent;
1914 /* Start at next port */
1915 tmp_rtp = src->next_port_num;
1917 if (stream->is_ipv6)
1918 host = "udp://[::0]";
1920 host = "udp://0.0.0.0";
1922 /* try to allocate 2 UDP ports, the RTP port should be an even
1923 * number and the RTCP port should be the next (uneven) port */
1926 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
1927 tmp_rtp >= src->client_port_range.max)
1930 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
1931 if (udpsrc0 == NULL)
1932 goto no_udp_protocol;
1933 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
1935 if (src->udp_buffer_size != 0)
1936 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
1939 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
1940 if (ret == GST_STATE_CHANGE_FAILURE) {
1942 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
1945 if (++count > src->retry)
1948 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1949 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1950 gst_object_unref (udpsrc0);
1953 GST_DEBUG_OBJECT (src, "retry %d", count);
1956 goto no_udp_protocol;
1959 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
1960 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
1962 /* check if port is even */
1963 if ((tmp_rtp & 0x01) != 0) {
1964 /* port not even, close and allocate another */
1965 if (++count > src->retry)
1968 GST_DEBUG_OBJECT (src, "RTP port not even");
1970 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1971 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1972 gst_object_unref (udpsrc0);
1975 GST_DEBUG_OBJECT (src, "retry %d", count);
1980 /* allocate port+1 for RTCP now */
1981 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
1982 if (udpsrc1 == NULL)
1983 goto no_udp_rtcp_protocol;
1986 tmp_rtcp = tmp_rtp + 1;
1987 if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
1990 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
1992 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
1993 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
1994 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
1995 if (ret == GST_STATE_CHANGE_FAILURE) {
1996 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
1998 if (++count > src->retry)
2001 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2002 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2003 gst_object_unref (udpsrc0);
2006 GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
2007 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2008 gst_object_unref (udpsrc1);
2012 GST_DEBUG_OBJECT (src, "retry %d", count);
2016 /* all fine, do port check */
2017 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
2018 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
2020 /* this should not happen... */
2021 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
2024 /* we keep these elements, we configure all in configure_transport when the
2025 * server told us to really use the UDP ports. */
2026 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
2027 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
2028 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
2029 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
2031 /* keep track of next available port number when we have a range
2033 if (src->next_port_num != 0)
2034 src->next_port_num = tmp_rtcp + 1;
2041 GST_DEBUG_OBJECT (src, "could not get UDP source");
2046 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
2050 no_udp_rtcp_protocol:
2052 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
2057 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
2058 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
2064 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2065 gst_object_unref (udpsrc0);
2068 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2069 gst_object_unref (udpsrc1);
2076 gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
2081 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
2083 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2084 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2087 for (i = 0; i < 2; i++) {
2088 if (stream->udpsrc[i])
2089 gst_element_set_state (stream->udpsrc[i], state);
2095 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
2102 event = gst_event_new_flush_start ();
2103 GST_DEBUG_OBJECT (src, "start flush");
2105 state = GST_STATE_PAUSED;
2107 event = gst_event_new_flush_stop (FALSE);
2108 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
2111 state = GST_STATE_PLAYING;
2113 state = GST_STATE_PAUSED;
2115 gst_rtspsrc_push_event (src, event);
2116 gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
2117 gst_rtspsrc_set_state (src, state);
2120 static GstRTSPResult
2121 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
2122 GstRTSPMessage * message, GTimeVal * timeout)
2127 ret = gst_rtsp_connection_send (conn, message, timeout);
2129 ret = GST_RTSP_ERROR;
2134 static GstRTSPResult
2135 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
2136 GstRTSPMessage * message, GTimeVal * timeout)
2141 ret = gst_rtsp_connection_receive (conn, message, timeout);
2143 ret = GST_RTSP_ERROR;
2149 gst_rtspsrc_get_position (GstRTSPSrc * src)
2154 query = gst_query_new_position (GST_FORMAT_TIME);
2155 /* should be known somewhere down the stream (e.g. jitterbuffer) */
2156 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2157 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2161 if (stream->srcpad) {
2162 if (gst_pad_query (stream->srcpad, query)) {
2163 gst_query_parse_position (query, &fmt, &pos);
2164 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
2165 GST_TIME_ARGS (pos));
2166 src->last_pos = pos;
2176 gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
2178 src->state = GST_RTSP_STATE_SEEKING;
2179 /* PLAY will add the range header now. */
2180 src->need_range = TRUE;
2186 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
2191 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
2193 gboolean flush, skip;
2196 GstSegment seeksegment = { 0, };
2200 GST_DEBUG_OBJECT (src, "doing seek with event");
2202 gst_event_parse_seek (event, &rate, &format, &flags,
2203 &cur_type, &cur, &stop_type, &stop);
2205 /* no negative rates yet */
2209 /* we need TIME format */
2210 if (format != src->segment.format)
2213 GST_DEBUG_OBJECT (src, "doing seek without event");
2215 cur_type = GST_SEEK_TYPE_SET;
2216 stop_type = GST_SEEK_TYPE_SET;
2219 /* get flush flag */
2220 flush = flags & GST_SEEK_FLAG_FLUSH;
2221 skip = flags & GST_SEEK_FLAG_SKIP;
2223 /* now we need to make sure the streaming thread is stopped. We do this by
2224 * either sending a FLUSH_START event downstream which will cause the
2225 * streaming thread to stop with a WRONG_STATE.
2226 * For a non-flushing seek we simply pause the task, which will happen as soon
2227 * as it completes one iteration (and thus might block when the sink is
2228 * blocking in preroll). */
2230 GST_DEBUG_OBJECT (src, "starting flush");
2231 gst_rtspsrc_flush (src, TRUE, FALSE);
2234 gst_task_pause (src->task);
2238 /* we should now be able to grab the streaming thread because we stopped it
2239 * with the above flush/pause code */
2240 GST_RTSP_STREAM_LOCK (src);
2242 GST_DEBUG_OBJECT (src, "stopped streaming");
2244 /* stop flushing the rtsp connection so we can send PAUSE/PLAY below */
2245 gst_rtspsrc_connection_flush (src, FALSE);
2247 /* copy segment, we need this because we still need the old
2248 * segment when we close the current segment. */
2249 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
2251 /* configure the seek parameters in the seeksegment. We will then have the
2252 * right values in the segment to perform the seek */
2254 GST_DEBUG_OBJECT (src, "configuring seek");
2255 gst_segment_do_seek (&seeksegment, rate, format, flags,
2256 cur_type, cur, stop_type, stop, &update);
2259 /* figure out the last position we need to play. If it's configured (stop !=
2260 * -1), use that, else we play until the total duration of the file */
2261 if ((stop = seeksegment.stop) == -1)
2262 stop = seeksegment.duration;
2264 playing = (src->state == GST_RTSP_STATE_PLAYING);
2266 /* if we were playing, pause first */
2268 /* obtain current position in case seek fails */
2269 gst_rtspsrc_get_position (src);
2270 gst_rtspsrc_pause (src, FALSE);
2274 gst_rtspsrc_do_seek (src, &seeksegment);
2276 /* and continue playing */
2278 gst_rtspsrc_play (src, &seeksegment, FALSE);
2280 /* prepare for streaming again */
2282 /* if we started flush, we stop now */
2283 GST_DEBUG_OBJECT (src, "stopping flush");
2284 gst_rtspsrc_flush (src, FALSE, playing);
2287 /* now we did the seek and can activate the new segment values */
2288 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
2290 /* if we're doing a segment seek, post a SEGMENT_START message */
2291 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2292 gst_element_post_message (GST_ELEMENT_CAST (src),
2293 gst_message_new_segment_start (GST_OBJECT_CAST (src),
2294 src->segment.format, src->segment.position));
2297 /* now create the newsegment */
2298 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
2299 " to %" G_GINT64_FORMAT, src->segment.position, stop);
2302 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
2303 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2304 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2305 stream->discont = TRUE;
2308 GST_RTSP_STREAM_UNLOCK (src);
2315 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
2320 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
2326 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
2330 gboolean res = TRUE;
2333 src = GST_RTSPSRC_CAST (parent);
2335 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
2336 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
2338 switch (GST_EVENT_TYPE (event)) {
2339 case GST_EVENT_SEEK:
2340 res = gst_rtspsrc_perform_seek (src, event);
2344 case GST_EVENT_NAVIGATION:
2345 case GST_EVENT_LATENCY:
2353 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
2354 res = gst_pad_send_event (target, event);
2355 gst_object_unref (target);
2357 gst_event_unref (event);
2360 gst_event_unref (event);
2366 /* this is the final event function we receive on the internal source pad when
2367 * we deal with TCP connections */
2369 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
2374 GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
2376 switch (GST_EVENT_TYPE (event)) {
2377 case GST_EVENT_SEEK:
2379 case GST_EVENT_NAVIGATION:
2380 case GST_EVENT_LATENCY:
2382 gst_event_unref (event);
2389 /* this is the final query function we receive on the internal source pad when
2390 * we deal with TCP connections */
2392 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
2396 gboolean res = TRUE;
2398 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
2400 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2401 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2403 switch (GST_QUERY_TYPE (query)) {
2404 case GST_QUERY_POSITION:
2409 case GST_QUERY_DURATION:
2413 gst_query_parse_duration (query, &format, NULL);
2416 case GST_FORMAT_TIME:
2417 gst_query_set_duration (query, format, src->segment.duration);
2425 case GST_QUERY_LATENCY:
2427 /* we are live with a min latency of 0 and unlimited max latency, this
2428 * result will be updated by the session manager if there is any. */
2429 gst_query_set_latency (query, TRUE, 0, -1);
2439 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2441 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2445 gboolean res = FALSE;
2447 src = GST_RTSPSRC_CAST (parent);
2449 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2450 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2452 switch (GST_QUERY_TYPE (query)) {
2453 case GST_QUERY_DURATION:
2457 gst_query_parse_duration (query, &format, NULL);
2460 case GST_FORMAT_TIME:
2461 gst_query_set_duration (query, format, src->segment.duration);
2469 case GST_QUERY_SEEKING:
2473 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2474 if (format == GST_FORMAT_TIME) {
2476 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2478 /* seeking without duration is unlikely */
2479 seekable = seekable && src->seekable && src->segment.duration &&
2480 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2482 /* FIXME ?? should we have 0 and segment.duration here; see demuxers */
2483 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
2484 src->segment.start, src->segment.stop);
2493 uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
2495 gst_query_set_uri (query, uri);
2503 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2505 /* forward the query to the proxy target pad */
2507 res = gst_pad_query (target, query);
2508 gst_object_unref (target);
2517 /* callback for RTCP messages to be sent to the server when operating in TCP
2519 static GstFlowReturn
2520 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2523 GstRTSPStream *stream;
2524 GstFlowReturn res = GST_FLOW_OK;
2529 GstRTSPMessage message = { 0 };
2530 GstRTSPConnection *conn;
2532 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2533 src = stream->parent;
2535 gst_buffer_map (buffer, &map, GST_MAP_READ);
2539 gst_rtsp_message_init_data (&message, stream->channel[1]);
2541 /* lend the body data to the message */
2542 gst_rtsp_message_take_body (&message, data, size);
2544 if (stream->conninfo.connection)
2545 conn = stream->conninfo.connection;
2547 conn = src->conninfo.connection;
2549 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2550 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2551 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2553 /* and steal it away again because we will free it when unreffing the
2555 gst_rtsp_message_steal_body (&message, &data, &size);
2556 gst_rtsp_message_unset (&message);
2558 gst_buffer_unmap (buffer, &map);
2559 gst_buffer_unref (buffer);
2564 static GstPadProbeReturn
2565 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2567 GstRTSPSrc *src = user_data;
2569 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2570 GST_DEBUG_PAD_NAME (pad));
2572 /* activate the streams */
2573 GST_OBJECT_LOCK (src);
2574 if (!src->need_activate)
2577 src->need_activate = FALSE;
2578 GST_OBJECT_UNLOCK (src);
2580 gst_rtspsrc_activate_streams (src);
2582 return GST_PAD_PROBE_OK;
2586 GST_OBJECT_UNLOCK (src);
2587 return GST_PAD_PROBE_OK;
2591 /* this callback is called when the session manager generated a new src pad with
2592 * payloaded RTP packets. We simply ghost the pad here. */
2594 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2597 GstPadTemplate *template;
2600 GstRTSPStream *stream;
2603 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2605 GST_RTSP_STATE_LOCK (src);
2607 name = gst_object_get_name (GST_OBJECT_CAST (pad));
2608 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
2609 goto unknown_stream;
2611 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);
2613 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
2615 goto unknown_stream;
2618 stream->ssrc = ssrc;
2620 /* we'll add it later see below */
2621 stream->added = TRUE;
2623 /* check if we added all streams */
2625 for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
2626 GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;
2628 GST_DEBUG_OBJECT (src, "stream %p, container %d, added %d, setup %d",
2629 ostream, ostream->container, ostream->added, ostream->setup);
2631 /* if we find a stream for which we did a setup that is not added, we
2632 * need to wait some more */
2633 if (ostream->setup && !ostream->added) {
2638 GST_RTSP_STATE_UNLOCK (src);
2640 /* create a new pad we will use to stream to */
2641 template = gst_static_pad_template_get (&rtptemplate);
2642 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
2643 gst_object_unref (template);
2646 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
2647 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
2648 gst_pad_set_active (stream->srcpad, TRUE);
2649 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
2652 GST_DEBUG_OBJECT (src, "We added all streams");
2653 /* when we get here, all stream are added and we can fire the no-more-pads
2655 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
2663 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
2664 GST_RTSP_STATE_UNLOCK (src);
2671 stream_get_caps_for_pt (GstRTSPStream * stream, guint pt)
2675 len = stream->ptmap->len;
2676 for (i = 0; i < len; i++) {
2677 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
2685 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
2687 GstRTSPStream *stream;
2690 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
2692 GST_RTSP_STATE_LOCK (src);
2693 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2695 goto unknown_stream;
2697 if ((caps = stream_get_caps_for_pt (stream, pt)))
2698 gst_caps_ref (caps);
2699 GST_RTSP_STATE_UNLOCK (src);
2705 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
2706 GST_RTSP_STATE_UNLOCK (src);
2712 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
2714 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
2720 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
2726 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
2732 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2734 GstRTSPSrc *src = stream->parent;
2737 g_object_get (source, "ssrc", &ssrc, NULL);
2739 GST_DEBUG_OBJECT (src, "source %08x, stream %08x, session %u received BYE",
2740 ssrc, stream->ssrc, stream->id);
2742 if (ssrc == stream->ssrc)
2743 gst_rtspsrc_do_stream_eos (src, stream);
2747 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2749 GstRTSPSrc *src = stream->parent;
2752 g_object_get (source, "ssrc", &ssrc, NULL);
2754 GST_WARNING_OBJECT (src, "source %08x, stream %08x in session %u timed out",
2755 ssrc, stream->ssrc, stream->id);
2757 if (ssrc == stream->ssrc)
2758 gst_rtspsrc_do_stream_eos (src, stream);
2762 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
2764 GstRTSPStream *stream;
2766 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
2768 /* get stream for session */
2769 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2771 gst_rtspsrc_do_stream_eos (src, stream);
2776 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
2778 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
2783 set_manager_buffer_mode (GstRTSPSrc * src)
2785 GObjectClass *klass;
2787 if (src->manager == NULL)
2790 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
2792 if (!g_object_class_find_property (klass, "buffer-mode"))
2795 if (src->buffer_mode != BUFFER_MODE_AUTO) {
2796 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
2801 GST_DEBUG_OBJECT (src,
2802 "auto buffering mode, have clock %" GST_PTR_FORMAT, src->provided_clock);
2804 if (src->provided_clock) {
2805 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
2807 if (clock == src->provided_clock) {
2808 GST_DEBUG_OBJECT (src, "selected synced");
2809 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SYNCED, NULL);
2812 gst_object_unref (clock);
2817 /* Otherwise fall-through and use another buffer mode */
2819 gst_object_unref (clock);
2822 GST_DEBUG_OBJECT (src, "auto buffering mode");
2823 if (src->use_buffering) {
2824 GST_DEBUG_OBJECT (src, "selected buffer");
2825 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER, NULL);
2827 GST_DEBUG_OBJECT (src, "selected slave");
2828 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
2832 /* try to get and configure a manager */
2834 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
2835 GstRTSPTransport * transport)
2837 const gchar *manager;
2839 GstStateChangeReturn ret;
2841 /* find a manager */
2842 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
2846 GST_DEBUG_OBJECT (src, "using manager %s", manager);
2848 /* configure the manager */
2849 if (src->manager == NULL) {
2850 GObjectClass *klass;
2852 if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
2854 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
2858 goto use_no_manager;
2860 if (!(src->manager = gst_element_factory_make (manager, "manager")))
2861 goto manager_failed;
2864 /* we manage this element */
2865 gst_element_set_locked_state (src->manager, TRUE);
2866 gst_bin_add (GST_BIN_CAST (src), src->manager);
2868 ret = gst_element_set_state (src->manager, GST_STATE_PAUSED);
2869 if (ret == GST_STATE_CHANGE_FAILURE)
2870 goto start_manager_failure;
2872 g_object_set (src->manager, "latency", src->latency, NULL);
2874 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
2876 if (g_object_class_find_property (klass, "ntp-sync")) {
2877 g_object_set (src->manager, "ntp-sync", src->ntp_sync, NULL);
2880 if (g_object_class_find_property (klass, "use-pipeline-clock")) {
2881 g_object_set (src->manager, "use-pipeline-clock",
2882 src->use_pipeline_clock, NULL);
2885 if (src->sdes && g_object_class_find_property (klass, "sdes")) {
2886 g_object_set (src->manager, "sdes", src->sdes, NULL);
2889 if (g_object_class_find_property (klass, "drop-on-latency")) {
2890 g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
2894 /* buffer mode pauses are handled by adding offsets to buffer times,
2895 * but some depayloaders may have a hard time syncing output times
2896 * with such input times, e.g. container ones, most notably ASF */
2897 /* TODO alternatives are having an event that indicates these shifts,
2898 * or having rtsp extensions provide suggestion on buffer mode */
2899 /* valid duration implies not likely live pipeline,
2900 * so slaving in jitterbuffer does not make much sense
2901 * (and might mess things up due to bursts) */
2902 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
2903 src->segment.duration && !stream->container) {
2904 src->use_buffering = TRUE;
2906 src->use_buffering = FALSE;
2909 set_manager_buffer_mode (src);
2911 /* connect to signals */
2912 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
2914 src->manager_sig_id =
2915 g_signal_connect (src->manager, "pad-added",
2916 (GCallback) new_manager_pad, src);
2917 src->manager_ptmap_id =
2918 g_signal_connect (src->manager, "request-pt-map",
2919 (GCallback) request_pt_map, src);
2921 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
2924 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_NEW_MANAGER], 0,
2928 /* we stream directly to the manager, get some pads. Each RTSP stream goes
2929 * into a separate RTP session. */
2930 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
2931 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
2933 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
2934 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
2937 /* now configure the bandwidth in the manager */
2938 if (g_signal_lookup ("get-internal-session",
2939 G_OBJECT_TYPE (src->manager)) != 0) {
2940 GObject *rtpsession;
2942 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
2945 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
2947 stream->session = rtpsession;
2949 if (stream->as_bandwidth != -1) {
2950 GST_INFO_OBJECT (src, "setting AS: %f",
2951 (gdouble) (stream->as_bandwidth * 1000));
2952 g_object_set (rtpsession, "bandwidth",
2953 (gdouble) (stream->as_bandwidth * 1000), NULL);
2955 if (stream->rr_bandwidth != -1) {
2956 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
2957 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
2960 if (stream->rs_bandwidth != -1) {
2961 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
2962 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
2966 g_object_set (rtpsession, "probation", src->probation, NULL);
2968 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2970 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
2972 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
2974 g_signal_connect (rtpsession, "on-ssrc-active",
2975 (GCallback) on_ssrc_active, stream);
2986 GST_DEBUG_OBJECT (src, "cannot get a session manager");
2991 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
2994 start_manager_failure:
2996 GST_DEBUG_OBJECT (src, "could not start session manager");
3001 /* free the UDP sources allocated when negotiating a transport.
3002 * This function is called when the server negotiated to a transport where the
3003 * UDP sources are not needed anymore, such as TCP or multicast. */
3005 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
3009 for (i = 0; i < 2; i++) {
3010 if (stream->udpsrc[i]) {
3011 GST_DEBUG ("free UDP source %d for stream %p", i, stream);
3012 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
3013 gst_object_unref (stream->udpsrc[i]);
3014 stream->udpsrc[i] = NULL;
3019 /* for TCP, create pads to send and receive data to and from the manager and to
3020 * intercept various events and queries
3023 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
3024 GstRTSPTransport * transport, GstPad ** outpad)
3027 GstPadTemplate *template;
3028 GstPad *pad0, *pad1;
3030 /* configure for interleaved delivery, nothing needs to be done
3031 * here, the loop function will call the chain functions of the
3032 * session manager. */
3033 stream->channel[0] = transport->interleaved.min;
3034 stream->channel[1] = transport->interleaved.max;
3035 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
3036 stream->channel[0], stream->channel[1]);
3038 /* we can remove the allocated UDP ports now */
3039 gst_rtspsrc_stream_free_udp (stream);
3041 /* no session manager, send data to srcpad directly */
3042 if (!stream->channelpad[0]) {
3043 GST_DEBUG_OBJECT (src, "no manager, creating pad");
3045 /* create a new pad we will use to stream to */
3046 name = g_strdup_printf ("stream_%u", stream->id);
3047 template = gst_static_pad_template_get (&rtptemplate);
3048 stream->channelpad[0] = gst_pad_new_from_template (template, name);
3049 gst_object_unref (template);
3052 /* set caps and activate */
3053 gst_pad_use_fixed_caps (stream->channelpad[0]);
3054 gst_pad_set_active (stream->channelpad[0], TRUE);
3056 *outpad = gst_object_ref (stream->channelpad[0]);
3058 GST_DEBUG_OBJECT (src, "using manager source pad");
3060 template = gst_static_pad_template_get (&anysrctemplate);
3062 /* allocate pads for sending the channel data into the manager */
3063 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
3064 gst_pad_link_full (pad0, stream->channelpad[0], GST_PAD_LINK_CHECK_NOTHING);
3065 gst_object_unref (stream->channelpad[0]);
3066 stream->channelpad[0] = pad0;
3067 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
3068 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
3069 gst_pad_set_element_private (pad0, src);
3070 gst_pad_set_active (pad0, TRUE);
3072 if (stream->channelpad[1]) {
3073 /* if we have a sinkpad for the other channel, create a pad and link to the
3075 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
3076 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
3077 gst_pad_link_full (pad1, stream->channelpad[1],
3078 GST_PAD_LINK_CHECK_NOTHING);
3079 gst_object_unref (stream->channelpad[1]);
3080 stream->channelpad[1] = pad1;
3081 gst_pad_set_active (pad1, TRUE);
3083 gst_object_unref (template);
3085 /* setup RTCP transport back to the server if we have to. */
3086 if (src->manager && src->do_rtcp) {
3089 template = gst_static_pad_template_get (&anysinktemplate);
3091 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
3092 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
3093 gst_pad_set_element_private (stream->rtcppad, stream);
3094 gst_pad_set_active (stream->rtcppad, TRUE);
3096 /* get session RTCP pad */
3097 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3098 pad = gst_element_get_request_pad (src->manager, name);
3103 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3104 gst_object_unref (pad);
3107 gst_object_unref (template);
3113 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
3114 GstRTSPTransport * transport, const gchar ** destination, gint * min,
3115 gint * max, guint * ttl)
3117 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3119 if (!(*destination = transport->destination))
3120 *destination = stream->destination;
3123 /* transport first */
3124 *min = transport->port.min;
3125 *max = transport->port.max;
3126 if (*min == -1 && *max == -1) {
3127 /* then try from SDP */
3128 if (stream->port != 0) {
3129 *min = stream->port;
3130 *max = stream->port + 1;
3136 if (!(*ttl = transport->ttl))
3141 /* first take the source, then the endpoint to figure out where to send
3143 if (!(*destination = transport->source)) {
3144 if (src->conninfo.connection)
3145 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
3146 else if (stream->conninfo.connection)
3148 gst_rtsp_connection_get_ip (stream->conninfo.connection);
3152 /* for unicast we only expect the ports here */
3153 *min = transport->server_port.min;
3154 *max = transport->server_port.max;
3159 /* For multicast create UDP sources and join the multicast group. */
3161 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
3162 GstRTSPTransport * transport, GstPad ** outpad)
3165 const gchar *destination;
3168 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
3170 /* we can remove the allocated UDP ports now */
3171 gst_rtspsrc_stream_free_udp (stream);
3173 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
3176 /* we need a destination now */
3177 if (destination == NULL)
3178 goto no_destination;
3180 /* we really need ports now or we won't be able to receive anything at all */
3181 if (min == -1 && max == -1)
3184 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
3185 destination, min, max);
3187 /* creating UDP source for RTP */
3189 uri = g_strdup_printf ("udp://%s:%d", destination, min);
3191 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3193 if (stream->udpsrc[0] == NULL)
3196 /* take ownership */
3197 gst_object_ref_sink (stream->udpsrc[0]);
3199 if (src->udp_buffer_size != 0)
3200 g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
3201 src->udp_buffer_size, NULL);
3203 if (src->multi_iface != NULL)
3204 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3205 src->multi_iface, NULL);
3208 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3209 gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
3212 /* creating another UDP source for RTCP */
3216 uri = g_strdup_printf ("udp://%s:%d", destination, max);
3218 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3220 if (stream->udpsrc[1] == NULL)
3223 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3224 stream->profile == GST_RTSP_PROFILE_SAVPF)
3225 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3227 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3228 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3229 gst_caps_unref (caps);
3231 /* take ownership */
3232 gst_object_ref_sink (stream->udpsrc[1]);
3234 if (src->multi_iface != NULL)
3235 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3236 src->multi_iface, NULL);
3238 gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
3245 GST_DEBUG_OBJECT (src, "no UDP source element found");
3250 GST_DEBUG_OBJECT (src, "no destination found");
3255 GST_DEBUG_OBJECT (src, "no ports found");
3260 /* configure the remainder of the UDP ports */
3262 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
3263 GstRTSPTransport * transport, GstPad ** outpad)
3265 /* we manage the UDP elements now. For unicast, the UDP sources where
3266 * allocated in the stream when we suggested a transport. */
3267 if (stream->udpsrc[0]) {
3268 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3269 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
3271 GST_DEBUG_OBJECT (src, "setting up UDP source");
3273 /* configure a timeout on the UDP port. When the timeout message is
3274 * posted, we assume UDP transport is not possible. We reconnect using TCP
3276 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
3277 src->udp_timeout * 1000, NULL);
3279 /* get output pad of the UDP source. */
3280 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
3282 /* save it so we can unblock */
3283 stream->blockedpad = *outpad;
3285 /* configure pad block on the pad. As soon as there is dataflow on the
3286 * UDP source, we know that UDP is not blocked by a firewall and we can
3287 * configure all the streams to let the application autoplug decoders. */
3289 gst_pad_add_probe (stream->blockedpad,
3290 GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, pad_blocked, src, NULL);
3292 if (stream->channelpad[0]) {
3293 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
3294 /* configure for UDP delivery, we need to connect the UDP pads to
3295 * the session plugin. */
3296 gst_pad_link_full (*outpad, stream->channelpad[0],
3297 GST_PAD_LINK_CHECK_NOTHING);
3298 gst_object_unref (*outpad);
3300 /* we connected to pad-added signal to get pads from the manager */
3302 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
3307 if (stream->udpsrc[1]) {
3310 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
3311 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
3313 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3314 stream->profile == GST_RTSP_PROFILE_SAVPF)
3315 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3317 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3318 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3319 gst_caps_unref (caps);
3321 if (stream->channelpad[1]) {
3324 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
3326 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
3327 gst_pad_link_full (pad, stream->channelpad[1],
3328 GST_PAD_LINK_CHECK_NOTHING);
3329 gst_object_unref (pad);
3331 /* leave unlinked */
3337 /* configure the UDP sink back to the server for status reports */
3339 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
3340 GstRTSPStream * stream, GstRTSPTransport * transport)
3343 gint rtp_port, rtcp_port;
3344 gboolean do_rtp, do_rtcp;
3345 const gchar *destination;
3350 /* get transport info */
3351 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
3352 &rtp_port, &rtcp_port, &ttl);
3354 /* see what we need to do */
3355 do_rtp = (rtp_port != -1);
3356 /* it's possible that the server does not want us to send RTCP in which case
3358 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
3360 /* we need a destination when we have RTP or RTCP ports */
3361 if (destination == NULL && (do_rtp || do_rtcp))
3362 goto no_destination;
3364 /* try to construct the fakesrc to the RTP port of the server to open up any
3367 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
3370 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
3371 stream->udpsink[0] =
3372 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3374 if (stream->udpsink[0] == NULL)
3375 goto no_sink_element;
3377 /* don't join multicast group, we will have the source socket do that */
3378 /* no sync or async state changes needed */
3379 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
3380 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3382 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3384 if (stream->udpsrc[0]) {
3385 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
3386 * so that NAT firewalls will open a hole for us */
3387 g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
3388 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
3389 /* configure socket and make sure udpsink does not close it when shutting
3390 * down, it belongs to udpsrc after all. */
3391 g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
3392 "close-socket", FALSE, NULL);
3393 g_object_unref (socket);
3396 /* the source for the dummy packets to open up NAT */
3397 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
3398 if (stream->fakesrc == NULL)
3399 goto no_fakesrc_element;
3401 /* random data in 5 buffers, a size of 200 bytes should be fine */
3402 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
3403 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
3405 /* we don't want to consider this a sink */
3406 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
3408 /* keep everything locked */
3409 gst_element_set_locked_state (stream->udpsink[0], TRUE);
3410 gst_element_set_locked_state (stream->fakesrc, TRUE);
3412 gst_object_ref (stream->udpsink[0]);
3413 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
3414 gst_object_ref (stream->fakesrc);
3415 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
3417 gst_element_link_pads_full (stream->fakesrc, "src", stream->udpsink[0],
3418 "sink", GST_PAD_LINK_CHECK_NOTHING);
3421 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
3424 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
3425 stream->udpsink[1] =
3426 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3428 if (stream->udpsink[1] == NULL)
3429 goto no_sink_element;
3431 /* don't join multicast group, we will have the source socket do that */
3432 /* no sync or async state changes needed */
3433 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
3434 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3436 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3438 if (stream->udpsrc[1]) {
3439 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
3440 * because some servers check the port number of where it sends RTCP to identify
3441 * the RTCP packets it receives */
3442 g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
3443 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
3444 /* configure socket and make sure udpsink does not close it when shutting
3445 * down, it belongs to udpsrc after all. */
3446 g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
3447 "close-socket", FALSE, NULL);
3448 g_object_unref (socket);
3451 /* we don't want to consider this a sink */
3452 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
3454 /* we keep this playing always */
3455 gst_element_set_locked_state (stream->udpsink[1], TRUE);
3456 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
3458 gst_object_ref (stream->udpsink[1]);
3459 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
3461 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
3463 /* get session RTCP pad */
3464 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3465 pad = gst_element_get_request_pad (src->manager, name);
3470 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3471 gst_object_unref (pad);
3480 GST_DEBUG_OBJECT (src, "no destination address specified");
3485 GST_DEBUG_OBJECT (src, "no UDP sink element found");
3490 GST_DEBUG_OBJECT (src, "no fakesrc element found");
3495 /* sets up all elements needed for streaming over the specified transport.
3496 * Does not yet expose the element pads, this will be done when there is actuall
3497 * dataflow detected, which might never happen when UDP is blocked in a
3498 * firewall, for example.
3501 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
3502 GstRTSPTransport * transport)
3505 GstPad *outpad = NULL;
3506 GstPadTemplate *template;
3508 const gchar *media_type;
3511 src = stream->parent;
3513 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
3515 /* get the proper media type for this stream now */
3516 if (gst_rtsp_transport_get_media_type (transport, &media_type) < 0)
3517 goto unknown_transport;
3519 goto unknown_transport;
3521 /* configure the final media type */
3522 GST_DEBUG_OBJECT (src, "setting media type to %s", media_type);
3524 len = stream->ptmap->len;
3525 for (i = 0; i < len; i++) {
3527 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
3529 if (item->caps == NULL)
3532 s = gst_caps_get_structure (item->caps, 0);
3533 gst_structure_set_name (s, media_type);
3536 /* try to get and configure a manager, channelpad[0-1] will be configured with
3537 * the pads for the manager, or NULL when no manager is needed. */
3538 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
3541 switch (transport->lower_transport) {
3542 case GST_RTSP_LOWER_TRANS_TCP:
3543 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
3544 goto transport_failed;
3546 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3547 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
3548 goto transport_failed;
3549 /* fallthrough, the rest is the same for UDP and MCAST */
3550 case GST_RTSP_LOWER_TRANS_UDP:
3551 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
3552 goto transport_failed;
3553 /* configure udpsinks back to the server for RTCP messages and for the
3554 * dummy RTP messages to open NAT. */
3555 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
3556 goto transport_failed;
3559 goto unknown_transport;
3563 GST_DEBUG_OBJECT (src, "creating ghostpad");
3565 gst_pad_use_fixed_caps (outpad);
3567 /* create ghostpad, don't add just yet, this will be done when we activate
3569 name = g_strdup_printf ("stream_%u", stream->id);
3570 template = gst_static_pad_template_get (&rtptemplate);
3571 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
3572 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
3573 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
3574 gst_object_unref (template);
3577 gst_object_unref (outpad);
3579 /* mark pad as ok */
3580 stream->last_ret = GST_FLOW_OK;
3587 GST_DEBUG_OBJECT (src, "failed to configure transport");
3592 GST_DEBUG_OBJECT (src, "unknown transport");
3597 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3602 /* send a couple of dummy random packets on the receiver RTP port to the server,
3603 * this should make a firewall think we initiated the data transfer and
3604 * hopefully allow packets to go from the sender port to our RTP receiver port */
3606 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
3610 if (src->nat_method != GST_RTSP_NAT_DUMMY)
3613 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3614 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3616 if (stream->fakesrc && stream->udpsink[0]) {
3617 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
3618 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
3619 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
3620 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
3621 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
3627 /* Adds the source pads of all configured streams to the element.
3628 * This code is performed when we detected dataflow.
3630 * We detect dataflow from either the _loop function or with pad probes on the
3634 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
3638 GST_DEBUG_OBJECT (src, "activating streams");
3640 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3641 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3643 if (stream->udpsrc[0]) {
3644 /* remove timeout, we are streaming now and timeouts will be handled by
3645 * the session manager and jitter buffer */
3646 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
3648 if (stream->srcpad) {
3649 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
3650 gst_pad_set_active (stream->srcpad, TRUE);
3652 /* if we don't have a session manager, set the caps now. If we have a
3653 * session, we will get a notification of the pad and the caps. */
3654 if (!src->manager) {
3657 caps = stream_get_caps_for_pt (stream, stream->default_pt);
3658 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
3659 gst_pad_set_caps (stream->srcpad, caps);
3662 if (!stream->added) {
3663 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
3664 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
3665 stream->added = TRUE;
3670 /* unblock all pads */
3671 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3672 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3674 if (stream->blockid) {
3675 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
3676 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
3677 stream->blockid = 0;
3685 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
3686 gboolean reset_manager)
3689 guint64 start, stop;
3690 gdouble play_speed, play_scale;
3692 GST_DEBUG_OBJECT (src, "configuring stream caps");
3694 start = segment->position;
3695 stop = segment->duration;
3696 play_speed = segment->rate;
3697 play_scale = segment->applied_rate;
3699 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3700 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3706 len = stream->ptmap->len;
3707 for (j = 0; j < len; j++) {
3709 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, j);
3711 if (item->caps == NULL)
3714 caps = gst_caps_make_writable (item->caps);
3716 if (stream->timebase != -1)
3717 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
3718 (guint) stream->timebase, NULL);
3719 if (stream->seqbase != -1)
3720 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
3721 (guint) stream->seqbase, NULL);
3722 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
3724 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
3725 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
3726 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
3729 GST_DEBUG_OBJECT (src, "stream %p, pt %d, caps %" GST_PTR_FORMAT, stream,
3733 if (reset_manager && src->manager) {
3734 GST_DEBUG_OBJECT (src, "clear session");
3735 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
3739 static GstFlowReturn
3740 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
3745 /* store the value */
3746 stream->last_ret = ret;
3748 /* if it's success we can return the value right away */
3749 if (ret == GST_FLOW_OK)
3752 /* any other error that is not-linked can be returned right
3754 if (ret != GST_FLOW_NOT_LINKED)
3757 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
3758 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3759 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3761 ret = ostream->last_ret;
3762 /* some other return value (must be SUCCESS but we can return
3763 * other values as well) */
3764 if (ret != GST_FLOW_NOT_LINKED)
3767 /* if we get here, all other pads were unlinked and we return
3768 * NOT_LINKED then */
3774 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
3777 gboolean res = TRUE;
3779 /* only streams that have a connection to the outside world */
3783 if (stream->udpsrc[0]) {
3784 gst_event_ref (event);
3785 res = gst_element_send_event (stream->udpsrc[0], event);
3786 } else if (stream->channelpad[0]) {
3787 gst_event_ref (event);
3788 if (GST_PAD_IS_SRC (stream->channelpad[0]))
3789 res = gst_pad_push_event (stream->channelpad[0], event);
3791 res = gst_pad_send_event (stream->channelpad[0], event);
3794 if (stream->udpsrc[1]) {
3795 gst_event_ref (event);
3796 res &= gst_element_send_event (stream->udpsrc[1], event);
3797 } else if (stream->channelpad[1]) {
3798 gst_event_ref (event);
3799 if (GST_PAD_IS_SRC (stream->channelpad[1]))
3800 res &= gst_pad_push_event (stream->channelpad[1], event);
3802 res &= gst_pad_send_event (stream->channelpad[1], event);
3806 gst_event_unref (event);
3812 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
3815 gboolean res = TRUE;
3817 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3818 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3820 gst_event_ref (event);
3821 res &= gst_rtspsrc_stream_push_event (src, ostream, event);
3823 gst_event_unref (event);
3828 static GstRTSPResult
3829 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3834 if (info->connection == NULL) {
3835 if (info->url == NULL) {
3836 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
3837 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
3841 /* create connection */
3842 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
3843 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
3844 goto could_not_create;
3847 g_free (info->url_str);
3848 info->url_str = gst_rtsp_url_get_request_uri (info->url);
3850 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
3852 if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
3853 if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
3854 src->tls_validation_flags))
3855 GST_WARNING_OBJECT (src, "Unable to set TLS validation flags");
3857 if (src->tls_database)
3858 gst_rtsp_connection_set_tls_database (info->connection,
3862 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
3863 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
3865 if (src->proxy_host) {
3866 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
3868 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
3873 if (!info->connected) {
3876 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
3877 ("Connecting to %s", info->location));
3878 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
3880 gst_rtsp_connection_connect (info->connection,
3881 src->ptcp_timeout)) < 0)
3882 goto could_not_connect;
3884 info->connected = TRUE;
3891 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
3896 gchar *str = gst_rtsp_strresult (res);
3897 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
3903 gchar *str = gst_rtsp_strresult (res);
3904 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
3910 static GstRTSPResult
3911 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
3914 GST_RTSP_STATE_LOCK (src);
3915 if (info->connected) {
3916 GST_DEBUG_OBJECT (src, "closing connection...");
3917 gst_rtsp_connection_close (info->connection);
3918 info->connected = FALSE;
3920 if (free && info->connection) {
3921 /* free connection */
3922 GST_DEBUG_OBJECT (src, "freeing connection...");
3923 gst_rtsp_connection_free (info->connection);
3924 info->connection = NULL;
3926 GST_RTSP_STATE_UNLOCK (src);
3930 static GstRTSPResult
3931 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3936 GST_DEBUG_OBJECT (src, "reconnecting connection...");
3937 gst_rtsp_conninfo_close (src, info, FALSE);
3938 res = gst_rtsp_conninfo_connect (src, info, async);
3944 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
3948 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
3949 GST_RTSP_STATE_LOCK (src);
3950 if (src->conninfo.connection && src->conninfo.flushing != flush) {
3951 GST_DEBUG_OBJECT (src, "connection flush");
3952 gst_rtsp_connection_flush (src->conninfo.connection, flush);
3953 src->conninfo.flushing = flush;
3955 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3956 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3957 if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
3958 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
3959 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
3960 stream->conninfo.flushing = flush;
3963 GST_RTSP_STATE_UNLOCK (src);
3966 /* FIXME, handle server request, reply with OK, for now */
3967 static GstRTSPResult
3968 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
3969 GstRTSPMessage * request)
3971 GstRTSPMessage response = { 0 };
3974 GST_DEBUG_OBJECT (src, "got server request message");
3977 gst_rtsp_message_dump (request);
3979 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
3981 if (res == GST_RTSP_ENOTIMPL) {
3982 /* default implementation, send OK */
3983 GST_DEBUG_OBJECT (src, "prepare OK reply");
3985 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
3990 /* let app parse and reply */
3991 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST],
3992 0, request, &response);
3995 gst_rtsp_message_dump (&response);
3997 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
4001 gst_rtsp_message_unset (&response);
4002 } else if (res == GST_RTSP_EEOF)
4010 gst_rtsp_message_unset (&response);
4015 /* send server keep-alive */
4016 static GstRTSPResult
4017 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
4019 GstRTSPMessage request = { 0 };
4021 GstRTSPMethod method;
4022 const gchar *control;
4024 if (src->do_rtsp_keep_alive == FALSE) {
4025 GST_DEBUG_OBJECT (src, "do-rtsp-keep-alive is FALSE, not sending.");
4026 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4030 GST_DEBUG_OBJECT (src, "creating server keep-alive");
4032 /* find a method to use for keep-alive */
4033 if (src->methods & GST_RTSP_GET_PARAMETER)
4034 method = GST_RTSP_GET_PARAMETER;
4036 method = GST_RTSP_OPTIONS;
4038 control = get_aggregate_control (src);
4039 if (control == NULL)
4042 res = gst_rtsp_message_init_request (&request, method, control);
4047 gst_rtsp_message_dump (&request);
4050 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
4055 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4056 gst_rtsp_message_unset (&request);
4063 GST_WARNING_OBJECT (src, "no control url to send keepalive");
4068 gchar *str = gst_rtsp_strresult (res);
4070 gst_rtsp_message_unset (&request);
4071 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
4072 ("Could not send keep-alive. (%s)", str));
4078 static GstFlowReturn
4079 gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
4081 GstFlowReturn ret = GST_FLOW_OK;
4083 GstRTSPStream *stream;
4084 GstPad *outpad = NULL;
4091 channel = message->type_data.data.channel;
4093 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
4095 goto unknown_stream;
4097 if (channel == stream->channel[0]) {
4098 outpad = stream->channelpad[0];
4100 } else if (channel == stream->channel[1]) {
4101 outpad = stream->channelpad[1];
4107 /* take a look at the body to figure out what we have */
4108 gst_rtsp_message_get_body (message, &data, &size);
4110 goto invalid_length;
4112 /* channels are not correct on some servers, do extra check */
4113 if (data[1] >= 200 && data[1] <= 204) {
4114 /* hmm RTCP message switch to the RTCP pad of the same stream. */
4115 outpad = stream->channelpad[1];
4119 /* we have no clue what this is, just ignore then. */
4121 goto unknown_stream;
4123 /* take the message body for further processing */
4124 gst_rtsp_message_steal_body (message, &data, &size);
4126 /* strip the trailing \0 */
4129 buf = gst_buffer_new ();
4130 gst_buffer_append_memory (buf,
4131 gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
4133 /* don't need message anymore */
4134 gst_rtsp_message_unset (message);
4136 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
4139 if (src->need_activate) {
4145 guint group_id = gst_util_group_id_next ();
4147 /* generate an SHA256 sum of the URI */
4148 cs = g_checksum_new (G_CHECKSUM_SHA256);
4149 uri = src->conninfo.location;
4150 g_checksum_update (cs, (const guchar *) uri, strlen (uri));
4152 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4153 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4156 g_strdup_printf ("%s/%d", g_checksum_get_string (cs), ostream->id);
4157 event = gst_event_new_stream_start (stream_id);
4158 gst_event_set_group_id (event, group_id);
4161 gst_rtspsrc_stream_push_event (src, ostream, event);
4163 g_checksum_free (cs);
4165 gst_rtspsrc_activate_streams (src);
4166 src->need_activate = FALSE;
4168 if ((event = src->start_segment) != NULL) {
4169 src->start_segment = NULL;
4170 gst_rtspsrc_push_event (src, event);
4173 if (src->base_time == -1) {
4174 /* Take current running_time. This timestamp will be put on
4175 * the first buffer of each stream because we are a live source and so we
4176 * timestamp with the running_time. When we are dealing with TCP, we also
4177 * only timestamp the first buffer (using the DISCONT flag) because a server
4178 * typically bursts data, for which we don't want to compensate by speeding
4179 * up the media. The other timestamps will be interpollated from this one
4180 * using the RTP timestamps. */
4181 GST_OBJECT_LOCK (src);
4182 if (GST_ELEMENT_CLOCK (src)) {
4184 GstClockTime base_time;
4186 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
4187 base_time = GST_ELEMENT_CAST (src)->base_time;
4189 src->base_time = now - base_time;
4191 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
4192 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
4194 GST_OBJECT_UNLOCK (src);
4197 if (stream->discont && !is_rtcp) {
4198 /* mark first RTP buffer as discont */
4199 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
4200 stream->discont = FALSE;
4201 /* first buffer gets the timestamp, other buffers are not timestamped and
4202 * their presentation time will be interpollated from the rtp timestamps. */
4203 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
4204 GST_TIME_ARGS (src->base_time));
4206 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
4209 /* chain to the peer pad */
4210 if (GST_PAD_IS_SINK (outpad))
4211 ret = gst_pad_chain (outpad, buf);
4213 ret = gst_pad_push (outpad, buf);
4216 /* combine all stream flows for the data transport */
4217 ret = gst_rtspsrc_combine_flows (src, stream, ret);
4224 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
4225 gst_rtsp_message_unset (message);
4230 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4231 ("Short message received, ignoring."));
4232 gst_rtsp_message_unset (message);
4237 static GstFlowReturn
4238 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
4240 GstRTSPMessage message = { 0 };
4242 GstFlowReturn ret = GST_FLOW_OK;
4243 GTimeVal tv_timeout;
4246 /* get the next timeout interval */
4247 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4249 /* see if the timeout period expired */
4250 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
4251 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
4252 /* send keep-alive, only act on interrupt, a warning will be posted for
4254 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4256 /* get new timeout */
4257 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4260 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
4261 tv_timeout.tv_sec, tv_timeout.tv_usec);
4263 /* protect the connection with the connection lock so that we can see when
4264 * we are finished doing server communication */
4266 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4267 &message, src->ptcp_timeout);
4271 GST_DEBUG_OBJECT (src, "we received a server message");
4273 case GST_RTSP_EINTR:
4274 /* we got interrupted this means we need to stop */
4276 case GST_RTSP_ETIMEOUT:
4277 /* no reply, send keep alive */
4278 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4279 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4283 /* go EOS when the server closed the connection */
4289 switch (message.type) {
4290 case GST_RTSP_MESSAGE_REQUEST:
4291 /* server sends us a request message, handle it */
4293 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4295 if (res == GST_RTSP_EEOF)
4298 goto handle_request_failed;
4300 case GST_RTSP_MESSAGE_RESPONSE:
4301 /* we ignore response messages */
4302 GST_DEBUG_OBJECT (src, "ignoring response message");
4304 gst_rtsp_message_dump (&message);
4306 case GST_RTSP_MESSAGE_DATA:
4307 GST_DEBUG_OBJECT (src, "got data message");
4308 ret = gst_rtspsrc_handle_data (src, &message);
4309 if (ret != GST_FLOW_OK)
4310 goto handle_data_failed;
4313 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4318 g_assert_not_reached ();
4323 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4324 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4325 ("The server closed the connection."));
4326 src->conninfo.connected = FALSE;
4327 gst_rtsp_message_unset (&message);
4328 return GST_FLOW_EOS;
4332 gst_rtsp_message_unset (&message);
4333 GST_DEBUG_OBJECT (src, "got interrupted");
4334 return GST_FLOW_FLUSHING;
4338 gchar *str = gst_rtsp_strresult (res);
4340 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4341 ("Could not receive message. (%s)", str));
4344 gst_rtsp_message_unset (&message);
4345 return GST_FLOW_ERROR;
4347 handle_request_failed:
4349 gchar *str = gst_rtsp_strresult (res);
4351 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4352 ("Could not handle server message. (%s)", str));
4354 gst_rtsp_message_unset (&message);
4355 return GST_FLOW_ERROR;
4359 GST_DEBUG_OBJECT (src, "could no handle data message");
4364 static GstFlowReturn
4365 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
4368 GstRTSPMessage message = { 0 };
4372 GTimeVal tv_timeout;
4374 /* get the next timeout interval */
4375 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4377 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
4378 (gint) tv_timeout.tv_sec);
4380 gst_rtsp_message_unset (&message);
4382 /* we should continue reading the TCP socket because the server might
4383 * send us requests. When the session timeout expires, we need to send a
4384 * keep-alive request to keep the session open. */
4385 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4386 &message, &tv_timeout);
4390 GST_DEBUG_OBJECT (src, "we received a server message");
4392 case GST_RTSP_EINTR:
4393 /* we got interrupted, see what we have to do */
4395 case GST_RTSP_ETIMEOUT:
4396 /* send keep-alive, ignore the result, a warning will be posted. */
4397 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4398 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4402 /* server closed the connection. not very fatal for UDP, reconnect and
4403 * see what happens. */
4404 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4405 ("The server closed the connection."));
4406 if (src->udp_reconnect) {
4408 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
4415 GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
4417 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4418 ("Unhandled return value %d.", res));
4422 switch (message.type) {
4423 case GST_RTSP_MESSAGE_REQUEST:
4424 /* server sends us a request message, handle it */
4426 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4428 if (res == GST_RTSP_EEOF)
4431 goto handle_request_failed;
4433 case GST_RTSP_MESSAGE_RESPONSE:
4434 /* we ignore response and data messages */
4435 GST_DEBUG_OBJECT (src, "ignoring response message");
4437 gst_rtsp_message_dump (&message);
4438 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
4439 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
4440 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
4441 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
4442 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4449 case GST_RTSP_MESSAGE_DATA:
4450 /* we ignore response and data messages */
4451 GST_DEBUG_OBJECT (src, "ignoring data message");
4454 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4459 g_assert_not_reached ();
4461 /* we get here when the connection got interrupted */
4464 gst_rtsp_message_unset (&message);
4465 GST_DEBUG_OBJECT (src, "got interrupted");
4466 return GST_FLOW_FLUSHING;
4470 gchar *str = gst_rtsp_strresult (res);
4473 src->conninfo.connected = FALSE;
4474 if (res != GST_RTSP_EINTR) {
4475 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
4476 ("Could not connect to server. (%s)", str));
4478 ret = GST_FLOW_ERROR;
4480 ret = GST_FLOW_FLUSHING;
4486 gchar *str = gst_rtsp_strresult (res);
4488 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4489 ("Could not receive message. (%s)", str));
4491 return GST_FLOW_ERROR;
4493 handle_request_failed:
4495 gchar *str = gst_rtsp_strresult (res);
4498 gst_rtsp_message_unset (&message);
4499 if (res != GST_RTSP_EINTR) {
4500 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4501 ("Could not handle server message. (%s)", str));
4503 ret = GST_FLOW_ERROR;
4505 ret = GST_FLOW_FLUSHING;
4511 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4512 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4513 ("The server closed the connection."));
4514 src->conninfo.connected = FALSE;
4515 gst_rtsp_message_unset (&message);
4516 return GST_FLOW_EOS;
4520 static GstRTSPResult
4521 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
4523 GstRTSPResult res = GST_RTSP_OK;
4526 GST_DEBUG_OBJECT (src, "doing reconnect");
4528 GST_OBJECT_LOCK (src);
4529 /* only restart when the pads were not yet activated, else we were
4530 * streaming over UDP */
4531 restart = src->need_activate;
4532 GST_OBJECT_UNLOCK (src);
4534 /* no need to restart, we're done */
4538 /* we can try only TCP now */
4539 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
4541 /* close and cleanup our state */
4542 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
4545 /* see if we have TCP left to try. Also don't try TCP when we were configured
4547 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
4550 /* We post a warning message now to inform the user
4551 * that nothing happened. It's most likely a firewall thing. */
4552 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4553 ("Could not receive any UDP packets for %.4f seconds, maybe your "
4554 "firewall is blocking it. Retrying using a TCP connection.",
4555 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
4557 /* open new connection using tcp */
4558 if (gst_rtspsrc_open (src, async) < 0)
4561 /* start playback */
4562 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
4571 src->cur_protocols = 0;
4572 /* no transport possible, post an error and stop */
4573 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4574 ("Could not receive any UDP packets for %.4f seconds, maybe your "
4575 "firewall is blocking it. No other protocols to try.",
4576 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
4577 return GST_RTSP_ERROR;
4581 GST_DEBUG_OBJECT (src, "open failed");
4586 GST_DEBUG_OBJECT (src, "play failed");
4592 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
4596 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
4599 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
4602 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
4605 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
4613 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
4617 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
4620 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
4623 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
4626 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
4634 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
4638 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
4641 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
4644 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
4647 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
4655 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
4659 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
4662 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
4665 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
4668 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
4676 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
4678 if (ret == GST_RTSP_OK)
4679 gst_rtspsrc_loop_complete_cmd (src, cmd);
4680 else if (ret == GST_RTSP_EINTR)
4681 gst_rtspsrc_loop_cancel_cmd (src, cmd);
4683 gst_rtspsrc_loop_error_cmd (src, cmd);
4687 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
4690 gboolean flushed = FALSE;
4692 /* start new request */
4693 gst_rtspsrc_loop_start_cmd (src, cmd);
4695 GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
4697 GST_OBJECT_LOCK (src);
4698 old = src->pending_cmd;
4699 if (old == CMD_RECONNECT) {
4700 GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
4701 cmd = CMD_RECONNECT;
4703 if (old != CMD_WAIT) {
4704 src->pending_cmd = CMD_WAIT;
4705 GST_OBJECT_UNLOCK (src);
4706 /* cancel previous request */
4707 GST_DEBUG_OBJECT (src, "cancel previous request %d", old);
4708 gst_rtspsrc_loop_cancel_cmd (src, old);
4709 GST_OBJECT_LOCK (src);
4711 src->pending_cmd = cmd;
4712 /* interrupt if allowed */
4713 if (src->busy_cmd & mask) {
4714 GST_DEBUG_OBJECT (src, "connection flush busy %d", src->busy_cmd);
4715 gst_rtspsrc_connection_flush (src, TRUE);
4718 GST_DEBUG_OBJECT (src, "not interrupting busy cmd %d", src->busy_cmd);
4721 gst_task_start (src->task);
4722 GST_OBJECT_UNLOCK (src);
4728 gst_rtspsrc_loop (GstRTSPSrc * src)
4732 if (!src->conninfo.connection || !src->conninfo.connected)
4735 if (src->interleaved)
4736 ret = gst_rtspsrc_loop_interleaved (src);
4738 ret = gst_rtspsrc_loop_udp (src);
4740 if (ret != GST_FLOW_OK)
4748 GST_WARNING_OBJECT (src, "we are not connected");
4749 ret = GST_FLOW_FLUSHING;
4754 const gchar *reason = gst_flow_get_name (ret);
4756 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
4757 src->running = FALSE;
4758 if (ret == GST_FLOW_EOS) {
4759 /* perform EOS logic */
4760 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
4761 gst_element_post_message (GST_ELEMENT_CAST (src),
4762 gst_message_new_segment_done (GST_OBJECT_CAST (src),
4763 src->segment.format, src->segment.position));
4764 gst_rtspsrc_push_event (src,
4765 gst_event_new_segment_done (src->segment.format,
4766 src->segment.position));
4768 gst_rtspsrc_push_event (src, gst_event_new_eos ());
4770 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
4771 /* for fatal errors we post an error message, post the error before the
4772 * EOS so the app knows about the error first. */
4773 GST_ELEMENT_ERROR (src, STREAM, FAILED,
4774 ("Internal data flow error."),
4775 ("streaming task paused, reason %s (%d)", reason, ret));
4776 gst_rtspsrc_push_event (src, gst_event_new_eos ());
4778 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_LOOP);
4783 #ifndef GST_DISABLE_GST_DEBUG
4784 static const gchar *
4785 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
4789 while (method != 0) {
4806 static const gchar *
4807 gst_rtspsrc_skip_lws (const gchar * s)
4809 while (g_ascii_isspace (*s))
4814 static const gchar *
4815 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
4817 while (s > start && g_ascii_isspace (*(s - 1)))
4822 static const gchar *
4823 gst_rtspsrc_skip_commas (const gchar * s)
4825 /* The grammar allows for multiple commas */
4826 while (g_ascii_isspace (*s) || *s == ',')
4831 static const gchar *
4832 gst_rtspsrc_skip_item (const gchar * s)
4834 gboolean quoted = FALSE;
4835 const gchar *start = s;
4837 /* A list item ends at the last non-whitespace character
4838 * before a comma which is not inside a quoted-string. Or at
4839 * the end of the string.
4845 if (*s == '\\' && *(s + 1))
4854 return gst_rtspsrc_unskip_lws (s, start);
4858 gst_rtsp_decode_quoted_string (gchar * quoted_string)
4862 src = quoted_string + 1;
4863 dst = quoted_string;
4864 while (*src && *src != '"') {
4865 if (*src == '\\' && *(src + 1))
4872 /* Extract the authentication tokens that the server provided for each method
4873 * into an array of structures and give those to the connection object.
4876 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
4877 const gchar * header, gboolean * stale)
4879 GSList *list = NULL, *iter;
4881 gchar *item, *eq, *name_end, *value;
4883 g_return_if_fail (stale != NULL);
4885 gst_rtsp_connection_clear_auth_params (conn);
4888 /* Parse a header whose content is described by RFC2616 as
4889 * "#something", where "something" does not itself contain commas,
4890 * except as part of quoted-strings, into a list of allocated strings.
4892 header = gst_rtspsrc_skip_commas (header);
4894 end = gst_rtspsrc_skip_item (header);
4895 list = g_slist_prepend (list, g_strndup (header, end - header));
4896 header = gst_rtspsrc_skip_commas (end);
4901 list = g_slist_reverse (list);
4902 for (iter = list; iter; iter = iter->next) {
4905 eq = strchr (item, '=');
4907 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
4908 if (name_end == item) {
4909 /* That's no good... */
4916 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
4918 gst_rtsp_decode_quoted_string (value);
4922 if (item && (strcmp (item, "stale") == 0) &&
4923 value && (strcmp (value, "TRUE") == 0))
4925 gst_rtsp_connection_set_auth_param (conn, item, value);
4929 g_slist_free (list);
4932 /* Parse a WWW-Authenticate Response header and determine the
4933 * available authentication methods
4935 * This code should also cope with the fact that each WWW-Authenticate
4936 * header can contain multiple challenge methods + tokens
4938 * At the moment, for Basic auth, we just do a minimal check and don't
4939 * even parse out the realm */
4941 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
4942 GstRTSPConnection * conn, gboolean * stale)
4946 g_return_if_fail (hdr != NULL);
4947 g_return_if_fail (methods != NULL);
4948 g_return_if_fail (stale != NULL);
4950 /* Skip whitespace at the start of the string */
4951 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
4953 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
4954 *methods |= GST_RTSP_AUTH_BASIC;
4955 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
4956 *methods |= GST_RTSP_AUTH_DIGEST;
4957 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
4962 * gst_rtspsrc_setup_auth:
4963 * @src: the rtsp source
4965 * Configure a username and password and auth method on the
4966 * connection object based on a response we received from the
4969 * Currently, this requires that a username and password were supplied
4970 * in the uri. In the future, they may be requested on demand by sending
4971 * a message up the bus.
4973 * Returns: TRUE if authentication information could be set up correctly.
4976 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
4980 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
4981 GstRTSPAuthMethod method;
4982 GstRTSPResult auth_result;
4984 GstRTSPConnection *conn;
4986 gboolean stale = FALSE;
4988 conn = src->conninfo.connection;
4990 /* Identify the available auth methods and see if any are supported */
4991 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
4992 &hdr, 0) == GST_RTSP_OK) {
4993 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
4996 if (avail_methods == GST_RTSP_AUTH_NONE)
4997 goto no_auth_available;
4999 /* For digest auth, if the response indicates that the session
5000 * data are stale, we just update them in the connection object and
5001 * return TRUE to retry the request */
5003 src->tried_url_auth = FALSE;
5005 url = gst_rtsp_connection_get_url (conn);
5007 /* Do we have username and password available? */
5008 if (url != NULL && !src->tried_url_auth && url->user != NULL
5009 && url->passwd != NULL) {
5012 src->tried_url_auth = TRUE;
5013 GST_DEBUG_OBJECT (src,
5014 "Attempting authentication using credentials from the URL");
5016 user = src->user_id;
5017 pass = src->user_pw;
5018 GST_DEBUG_OBJECT (src,
5019 "Attempting authentication using credentials from the properties");
5022 /* FIXME: If the url didn't contain username and password or we tried them
5023 * already, request a username and passwd from the application via some kind
5024 * of credentials request message */
5026 /* If we don't have a username and passwd at this point, bail out. */
5027 if (user == NULL || pass == NULL)
5030 /* Try to configure for each available authentication method, strongest to
5032 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
5033 /* Check if this method is available on the server */
5034 if ((method & avail_methods) == 0)
5037 /* Pass the credentials to the connection to try on the next request */
5038 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
5039 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
5040 * ignore it and end up retrying later */
5041 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
5042 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
5043 gst_rtsp_auth_method_to_string (method));
5048 if (method == GST_RTSP_AUTH_NONE)
5049 goto no_auth_available;
5055 /* Output an error indicating that we couldn't connect because there were
5056 * no supported authentication protocols */
5057 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5058 ("No supported authentication protocol was found"));
5063 /* We don't fire an error message, we just return FALSE and let the
5064 * normal NOT_AUTHORIZED error be propagated */
5069 static GstRTSPResult
5070 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5071 GstRTSPMessage * request, GstRTSPMessage * response,
5072 GstRTSPStatusCode * code)
5075 GstRTSPStatusCode thecode;
5076 gchar *content_base = NULL;
5080 if (!src->short_header)
5081 gst_rtsp_ext_list_before_send (src->extensions, request);
5083 GST_DEBUG_OBJECT (src, "sending message");
5086 gst_rtsp_message_dump (request);
5088 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
5092 gst_rtsp_connection_reset_timeout (conn);
5095 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
5100 gst_rtsp_message_dump (response);
5102 switch (response->type) {
5103 case GST_RTSP_MESSAGE_REQUEST:
5104 res = gst_rtspsrc_handle_request (src, conn, response);
5105 if (res == GST_RTSP_EEOF)
5108 goto handle_request_failed;
5110 case GST_RTSP_MESSAGE_RESPONSE:
5111 /* ok, a response is good */
5112 GST_DEBUG_OBJECT (src, "received response message");
5114 case GST_RTSP_MESSAGE_DATA:
5115 /* get next response */
5116 GST_DEBUG_OBJECT (src, "handle data response message");
5117 gst_rtspsrc_handle_data (src, response);
5120 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
5125 thecode = response->type_data.response.code;
5127 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
5129 /* if the caller wanted the result code, we store it. */
5133 /* If the request didn't succeed, bail out before doing any more */
5134 if (thecode != GST_RTSP_STS_OK)
5137 /* store new content base if any */
5138 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
5141 g_free (src->content_base);
5142 src->content_base = g_strdup (content_base);
5144 gst_rtsp_ext_list_after_send (src->extensions, request, response);
5151 gchar *str = gst_rtsp_strresult (res);
5153 if (res != GST_RTSP_EINTR) {
5154 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5155 ("Could not send message. (%s)", str));
5157 GST_WARNING_OBJECT (src, "send interrupted");
5166 GST_WARNING_OBJECT (src, "server closed connection");
5167 if ((try == 0) && !src->interleaved && src->udp_reconnect) {
5169 /* if reconnect succeeds, try again */
5171 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
5175 /* only try once after reconnect, then fallthrough and error out */
5178 gchar *str = gst_rtsp_strresult (res);
5180 if (res != GST_RTSP_EINTR) {
5181 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5182 ("Could not receive message. (%s)", str));
5184 GST_WARNING_OBJECT (src, "receive interrupted");
5192 handle_request_failed:
5194 /* ERROR was posted */
5195 gst_rtsp_message_unset (response);
5200 GST_DEBUG_OBJECT (src, "we got an eof from the server");
5201 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5202 ("The server closed the connection."));
5203 gst_rtsp_message_unset (response);
5210 * @src: the rtsp source
5211 * @conn: the connection to send on
5212 * @request: must point to a valid request
5213 * @response: must point to an empty #GstRTSPMessage
5214 * @code: an optional code result
5216 * send @request and retrieve the response in @response. optionally @code can be
5217 * non-NULL in which case it will contain the status code of the response.
5219 * If This function returns #GST_RTSP_OK, @response will contain a valid response
5220 * message that should be cleaned with gst_rtsp_message_unset() after usage.
5222 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
5223 * @response message) if the response code was not 200 (OK).
5225 * If the attempt results in an authentication failure, then this will attempt
5226 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
5229 * Returns: #GST_RTSP_OK if the processing was successful.
5231 static GstRTSPResult
5232 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5233 GstRTSPMessage * request, GstRTSPMessage * response,
5234 GstRTSPStatusCode * code)
5236 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
5237 GstRTSPResult res = GST_RTSP_ERROR;
5240 GstRTSPMethod method = GST_RTSP_INVALID;
5246 /* make sure we don't loop forever */
5250 /* save method so we can disable it when the server complains */
5251 method = request->type_data.request.method;
5254 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
5258 case GST_RTSP_STS_UNAUTHORIZED:
5259 if (gst_rtspsrc_setup_auth (src, response)) {
5260 /* Try the request/response again after configuring the auth info
5268 } while (retry == TRUE);
5270 /* If the user requested the code, let them handle errors, otherwise
5271 * post an error below */
5274 else if (int_code != GST_RTSP_STS_OK)
5275 goto error_response;
5282 GST_DEBUG_OBJECT (src, "got error %d", res);
5287 res = GST_RTSP_ERROR;
5289 switch (response->type_data.response.code) {
5290 case GST_RTSP_STS_NOT_FOUND:
5291 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
5292 response->type_data.response.reason));
5294 case GST_RTSP_STS_MOVED_PERMANENTLY:
5295 case GST_RTSP_STS_MOVE_TEMPORARILY:
5297 gchar *new_location;
5298 GstRTSPLowerTrans transports;
5300 GST_DEBUG_OBJECT (src, "got redirection");
5301 /* if we don't have a Location Header, we must error */
5302 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
5303 &new_location, 0) < 0)
5306 /* When we receive a redirect result, we go back to the INIT state after
5307 * parsing the new URI. The caller should do the needed steps to issue
5308 * a new setup when it detects this state change. */
5309 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
5311 /* save current transports */
5312 if (src->conninfo.url)
5313 transports = src->conninfo.url->transports;
5315 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
5317 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
5319 /* set old transports */
5320 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
5321 src->conninfo.url->transports = transports;
5323 src->need_redirect = TRUE;
5324 src->state = GST_RTSP_STATE_INIT;
5328 case GST_RTSP_STS_NOT_ACCEPTABLE:
5329 case GST_RTSP_STS_NOT_IMPLEMENTED:
5330 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
5331 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
5332 gst_rtsp_method_as_text (method));
5333 src->methods &= ~method;
5337 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5338 ("Got error response: %d (%s).", response->type_data.response.code,
5339 response->type_data.response.reason));
5342 /* if we return ERROR we should unset the response ourselves */
5343 if (res == GST_RTSP_ERROR)
5344 gst_rtsp_message_unset (response);
5350 static GstRTSPResult
5351 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
5352 GstRTSPMessage * response, GstRTSPSrc * src)
5354 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
5359 /* parse the response and collect all the supported methods. We need this
5360 * information so that we don't try to send an unsupported request to the
5364 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
5366 GstRTSPHeaderField field;
5370 /* reset supported methods */
5373 /* Try Allow Header first */
5374 field = GST_RTSP_HDR_ALLOW;
5377 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5378 if (indx == 0 && !respoptions) {
5379 /* if no Allow header was found then try the Public header... */
5380 field = GST_RTSP_HDR_PUBLIC;
5381 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5386 src->methods |= gst_rtsp_options_from_text (respoptions);
5391 if (src->methods == 0) {
5392 /* neither Allow nor Public are required, assume the server supports
5393 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
5395 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
5396 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
5398 /* always assume PLAY, FIXME, extensions should be able to override
5400 src->methods |= GST_RTSP_PLAY;
5401 /* also assume it will support Range */
5402 src->seekable = TRUE;
5404 /* we need describe and setup */
5405 if (!(src->methods & GST_RTSP_DESCRIBE))
5407 if (!(src->methods & GST_RTSP_SETUP))
5415 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5416 ("Server does not support DESCRIBE."));
5421 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5422 ("Server does not support SETUP."));
5427 /* masks to be kept in sync with the hardcoded protocol order of preference
5429 static guint protocol_masks[] = {
5430 GST_RTSP_LOWER_TRANS_UDP,
5431 GST_RTSP_LOWER_TRANS_UDP_MCAST,
5432 GST_RTSP_LOWER_TRANS_TCP,
5436 static GstRTSPResult
5437 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
5438 GstRTSPLowerTrans protocols, GstRTSPProfile profile, gchar ** transports)
5442 gboolean add_udp_str;
5447 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
5452 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
5454 /* extension listed transports, use those */
5455 if (*transports != NULL)
5458 /* it's the default */
5459 add_udp_str = FALSE;
5461 /* the default RTSP transports */
5462 result = g_string_new ("RTP");
5465 case GST_RTSP_PROFILE_AVP:
5466 g_string_append (result, "/AVP");
5468 case GST_RTSP_PROFILE_SAVP:
5469 g_string_append (result, "/SAVP");
5471 case GST_RTSP_PROFILE_AVPF:
5472 g_string_append (result, "/AVPF");
5474 case GST_RTSP_PROFILE_SAVPF:
5475 g_string_append (result, "/SAVPF");
5481 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
5482 GST_DEBUG_OBJECT (src, "adding UDP unicast");
5484 g_string_append (result, "/UDP");
5485 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
5486 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
5487 GST_DEBUG_OBJECT (src, "adding UDP multicast");
5488 /* we don't have to allocate any UDP ports yet, if the selected transport
5489 * turns out to be multicast we can create them and join the multicast
5490 * group indicated in the transport reply */
5492 g_string_append (result, "/UDP");
5493 g_string_append (result, ";multicast");
5494 if (src->next_port_num != 0) {
5495 if (src->client_port_range.max > 0 &&
5496 src->next_port_num >= src->client_port_range.max)
5499 g_string_append_printf (result, ";client_port=%d-%d",
5500 src->next_port_num, src->next_port_num + 1);
5502 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
5503 GST_DEBUG_OBJECT (src, "adding TCP");
5505 g_string_append (result, "/TCP;unicast;interleaved=%%i1-%%i2");
5507 *transports = g_string_free (result, FALSE);
5509 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
5516 GST_ERROR ("extension gave error %d", res);
5521 GST_ERROR ("no more ports available");
5522 return GST_RTSP_ERROR;
5526 static GstRTSPResult
5527 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
5528 gint orig_rtpport, gint orig_rtcpport)
5531 gint nr_udp, nr_int;
5533 gint rtpport = 0, rtcpport = 0;
5536 src = stream->parent;
5538 /* find number of placeholders first */
5539 if (strstr (*transports, "%%i2"))
5541 else if (strstr (*transports, "%%i1"))
5546 if (strstr (*transports, "%%u2"))
5548 else if (strstr (*transports, "%%u1"))
5553 if (nr_udp == 0 && nr_int == 0)
5557 if (!orig_rtpport || !orig_rtcpport) {
5558 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
5561 rtpport = orig_rtpport;
5562 rtcpport = orig_rtcpport;
5566 str = g_string_new ("");
5568 while ((next = strstr (p, "%%"))) {
5569 g_string_append_len (str, p, next - p);
5570 if (next[2] == 'u') {
5572 g_string_append_printf (str, "%d", rtpport);
5573 else if (next[3] == '2')
5574 g_string_append_printf (str, "%d", rtcpport);
5576 if (next[2] == 'i') {
5578 g_string_append_printf (str, "%d", src->free_channel);
5579 else if (next[3] == '2')
5580 g_string_append_printf (str, "%d", src->free_channel + 1);
5585 /* append final part */
5586 g_string_append (str, p);
5588 g_free (*transports);
5589 *transports = g_string_free (str, FALSE);
5597 GST_ERROR ("failed to allocate udp ports");
5598 return GST_RTSP_ERROR;
5602 /* Perform the SETUP request for all the streams.
5604 * We ask the server for a specific transport, which initially includes all the
5605 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
5606 * two local UDP ports that we send to the server.
5608 * Once the server replied with a transport, we configure the other streams
5609 * with the same transport.
5611 * This function will also configure the stream for the selected transport,
5612 * which basically means creating the pipeline.
5614 static GstRTSPResult
5615 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
5618 GstRTSPResult res = GST_RTSP_ERROR;
5619 GstRTSPMessage request = { 0 };
5620 GstRTSPMessage response = { 0 };
5621 GstRTSPStream *stream = NULL;
5622 GstRTSPLowerTrans protocols;
5623 GstRTSPStatusCode code;
5624 gboolean unsupported_real = FALSE;
5625 gint rtpport, rtcpport;
5629 if (src->conninfo.connection) {
5630 url = gst_rtsp_connection_get_url (src->conninfo.connection);
5631 /* we initially allow all configured lower transports. based on the URL
5632 * transports and the replies from the server we narrow them down. */
5633 protocols = url->transports & src->cur_protocols;
5636 protocols = src->cur_protocols;
5642 /* reset some state */
5643 src->free_channel = 0;
5644 src->interleaved = FALSE;
5645 src->need_activate = FALSE;
5646 /* keep track of next port number, 0 is random */
5647 src->next_port_num = src->client_port_range.min;
5648 rtpport = rtcpport = 0;
5650 if (G_UNLIKELY (src->streams == NULL))
5653 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5654 GstRTSPConnection *conn;
5661 stream = (GstRTSPStream *) walk->data;
5663 caps = stream_get_caps_for_pt (stream, stream->default_pt);
5665 GST_DEBUG_OBJECT (src, "skipping stream %p, no caps", stream);
5669 if (stream->skipped) {
5670 GST_DEBUG_OBJECT (src, "skipping stream %p", stream);
5674 /* see if we need to configure this stream */
5675 if (!gst_rtsp_ext_list_configure_stream (src->extensions, caps)) {
5676 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
5681 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_SELECT_STREAM], 0,
5682 stream->id, caps, &selected);
5684 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by signal", stream);
5688 /* merge/overwrite global caps */
5693 s = gst_caps_get_structure (caps, 0);
5695 num = gst_structure_n_fields (src->props);
5696 for (j = 0; j < num; j++) {
5700 name = gst_structure_nth_field_name (src->props, j);
5701 val = gst_structure_get_value (src->props, name);
5702 gst_structure_set_value (s, name, val);
5704 GST_DEBUG_OBJECT (src, "copied %s", name);
5708 /* skip setup if we have no URL for it */
5709 if (stream->conninfo.location == NULL) {
5710 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
5714 if (src->conninfo.connection == NULL) {
5715 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
5716 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
5719 conn = stream->conninfo.connection;
5721 conn = src->conninfo.connection;
5723 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
5724 stream->conninfo.location);
5726 /* if we have a multicast connection, only suggest multicast from now on */
5727 if (stream->is_multicast)
5728 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
5731 /* first selectable protocol */
5732 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5734 if (!protocol_masks[mask])
5738 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
5739 protocol_masks[mask]);
5740 /* create a string with first transport in line */
5742 res = gst_rtspsrc_create_transports_string (src,
5743 protocols & protocol_masks[mask], stream->profile, &transports);
5744 if (res < 0 || transports == NULL)
5745 goto setup_transport_failed;
5747 if (strlen (transports) == 0) {
5748 g_free (transports);
5749 GST_DEBUG_OBJECT (src, "no transports found");
5754 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
5756 /* replace placeholders with real values, this function will optionally
5757 * allocate UDP ports and other info needed to execute the setup request */
5758 res = gst_rtspsrc_prepare_transports (stream, &transports,
5759 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
5761 g_free (transports);
5762 goto setup_transport_failed;
5765 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
5767 /* create SETUP request */
5769 gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
5770 stream->conninfo.location);
5772 g_free (transports);
5773 goto create_request_failed;
5776 /* select transport */
5777 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
5779 /* if the user wants a non default RTP packet size we add the blocksize
5781 if (src->rtp_blocksize > 0) {
5782 hval = g_strdup_printf ("%d", src->rtp_blocksize);
5783 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
5787 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
5790 /* handle the code ourselves */
5791 if ((res = gst_rtspsrc_send (src, conn, &request, &response, &code) < 0))
5795 case GST_RTSP_STS_OK:
5797 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
5798 gst_rtsp_message_unset (&request);
5799 gst_rtsp_message_unset (&response);
5800 /* cleanup of leftover transport */
5801 gst_rtspsrc_stream_free_udp (stream);
5802 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
5803 * we might be in this case */
5804 if (stream->container && rtpport && rtcpport && !retry) {
5805 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
5810 /* this transport did not go down well, but we may have others to try
5811 * that we did not send yet, try those and only give up then
5812 * but not without checking for lost cause/extension so we can
5813 * post a nicer/more useful error message later */
5814 if (!unsupported_real)
5815 unsupported_real = stream->is_real;
5816 /* select next available protocol, give up on this stream if none */
5818 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5820 if (!protocol_masks[mask] || unsupported_real)
5825 /* cleanup of leftover transport and move to the next stream */
5826 gst_rtspsrc_stream_free_udp (stream);
5827 goto response_error;
5830 /* parse response transport */
5832 gchar *resptrans = NULL;
5833 GstRTSPTransport transport = { 0 };
5835 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
5838 gst_rtspsrc_stream_free_udp (stream);
5842 /* parse transport, go to next stream on parse error */
5843 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
5844 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
5848 /* update allowed transports for other streams. once the transport of
5849 * one stream has been determined, we make sure that all other streams
5850 * are configured in the same way */
5851 switch (transport.lower_transport) {
5852 case GST_RTSP_LOWER_TRANS_TCP:
5853 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
5854 protocols = GST_RTSP_LOWER_TRANS_TCP;
5855 src->interleaved = TRUE;
5856 /* update free channels */
5858 MAX (transport.interleaved.min, src->free_channel);
5860 MAX (transport.interleaved.max, src->free_channel);
5861 src->free_channel++;
5863 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
5864 /* only allow multicast for other streams */
5865 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
5866 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
5867 /* if the server selected our ports, increment our counters so that
5868 * we select a new port later */
5869 if (src->next_port_num == transport.port.min &&
5870 src->next_port_num + 1 == transport.port.max) {
5871 src->next_port_num += 2;
5874 case GST_RTSP_LOWER_TRANS_UDP:
5875 /* only allow unicast for other streams */
5876 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
5877 protocols = GST_RTSP_LOWER_TRANS_UDP;
5880 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
5881 transport.lower_transport);
5885 if (!stream->container || (!src->interleaved && !retry)) {
5886 /* now configure the stream with the selected transport */
5887 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
5888 GST_DEBUG_OBJECT (src,
5889 "could not configure stream %p transport, skipping stream",
5892 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
5893 /* retain the first allocated UDP port pair */
5894 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
5895 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
5898 /* we need to activate at least one streams when we detect activity */
5899 src->need_activate = TRUE;
5901 /* stream is setup now */
5902 stream->setup = TRUE;
5907 GstRTSPStream *sskip;
5909 skip = g_list_next (skip);
5913 sskip = (GstRTSPStream *) skip->data;
5915 /* skip all streams with the same control url */
5916 if (g_str_equal (stream->conninfo.location, sskip->conninfo.location)) {
5917 GST_DEBUG_OBJECT (src, "found stream %p with same control %s",
5918 sskip, sskip->conninfo.location);
5919 sskip->skipped = TRUE;
5924 /* clean up our transport struct */
5925 gst_rtsp_transport_init (&transport);
5926 /* clean up used RTSP messages */
5927 gst_rtsp_message_unset (&request);
5928 gst_rtsp_message_unset (&response);
5932 /* store the transport protocol that was configured */
5933 src->cur_protocols = protocols;
5935 gst_rtsp_ext_list_stream_select (src->extensions, url);
5937 /* if there is nothing to activate, error out */
5938 if (!src->need_activate)
5939 goto nothing_to_activate;
5946 /* no transport possible, post an error and stop */
5947 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5948 ("Could not connect to server, no protocols left"));
5949 return GST_RTSP_ERROR;
5953 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5954 ("SDP contains no streams"));
5955 return GST_RTSP_ERROR;
5957 create_request_failed:
5959 gchar *str = gst_rtsp_strresult (res);
5961 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5962 ("Could not create request. (%s)", str));
5966 setup_transport_failed:
5968 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5969 ("Could not setup transport."));
5970 res = GST_RTSP_ERROR;
5975 const gchar *str = gst_rtsp_status_as_text (code);
5977 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5978 ("Error (%d): %s", code, GST_STR_NULL (str)));
5979 res = GST_RTSP_ERROR;
5984 gchar *str = gst_rtsp_strresult (res);
5986 if (res != GST_RTSP_EINTR) {
5987 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5988 ("Could not send message. (%s)", str));
5990 GST_WARNING_OBJECT (src, "send interrupted");
5997 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5998 ("Server did not select transport."));
5999 res = GST_RTSP_ERROR;
6002 nothing_to_activate:
6004 /* none of the available error codes is really right .. */
6005 if (unsupported_real) {
6006 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6007 (_("No supported stream was found. You might need to install a "
6008 "GStreamer RTSP extension plugin for Real media streams.")),
6011 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6012 (_("No supported stream was found. You might need to allow "
6013 "more transport protocols or may otherwise be missing "
6014 "the right GStreamer RTSP extension plugin.")), (NULL));
6016 return GST_RTSP_ERROR;
6020 gst_rtsp_message_unset (&request);
6021 gst_rtsp_message_unset (&response);
6027 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
6028 GstSegment * segment)
6031 GstRTSPTimeRange *therange;
6034 gst_rtsp_range_free (src->range);
6036 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
6037 GST_DEBUG_OBJECT (src, "parsed range %s", range);
6038 src->range = therange;
6040 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
6042 gst_segment_init (segment, GST_FORMAT_TIME);
6046 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
6047 therange->min.type, therange->min.seconds, therange->max.type,
6048 therange->max.seconds);
6050 if (therange->min.type == GST_RTSP_TIME_NOW)
6052 else if (therange->min.type == GST_RTSP_TIME_END)
6055 seconds = therange->min.seconds * GST_SECOND;
6057 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
6058 GST_TIME_ARGS (seconds));
6060 /* we need to start playback without clipping from the position reported by
6062 segment->start = seconds;
6063 segment->position = seconds;
6065 if (therange->max.type == GST_RTSP_TIME_NOW)
6067 else if (therange->max.type == GST_RTSP_TIME_END)
6070 seconds = therange->max.seconds * GST_SECOND;
6072 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
6073 GST_TIME_ARGS (seconds));
6075 /* live (WMS) server might send overflowed large max as its idea of infinity,
6076 * compensate to prevent problems later on */
6077 if (seconds != -1 && seconds < 0) {
6079 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
6082 /* live (WMS) might send min == max, which is not worth recording */
6083 if (segment->duration == -1 && seconds == segment->start)
6086 /* don't change duration with unknown value, we might have a valid value
6087 * there that we want to keep. */
6089 segment->duration = seconds;
6094 /* Parse clock profived by the server with following syntax:
6096 * "GstNetTimeProvider <wrapped-clock> <server-IP:port> <clock-time>"
6099 gst_rtspsrc_parse_gst_clock (GstRTSPSrc * src, const gchar * gstclock)
6101 gboolean res = FALSE;
6103 if (g_str_has_prefix (gstclock, "GstNetTimeProvider ")) {
6104 gchar **fields = NULL, **parts = NULL;
6105 gchar *remote_ip, *str;
6107 GstClockTime base_time;
6110 fields = g_strsplit (gstclock, " ", 0);
6112 /* wrapped clock, not very interesting for now */
6113 if (fields[1] == NULL)
6116 /* remote IP address and port */
6117 if ((str = fields[2]) == NULL)
6120 parts = g_strsplit (str, ":", 0);
6122 if ((remote_ip = parts[0]) == NULL)
6125 if ((str = parts[1]) == NULL)
6133 if ((str = fields[3]) == NULL)
6136 base_time = g_ascii_strtoull (str, NULL, 10);
6139 gst_net_client_clock_new ((gchar *) "GstRTSPClock", remote_ip, port,
6142 if (src->provided_clock)
6143 gst_object_unref (src->provided_clock);
6144 src->provided_clock = netclock;
6146 gst_element_post_message (GST_ELEMENT_CAST (src),
6147 gst_message_new_clock_provide (GST_OBJECT_CAST (src),
6148 src->provided_clock, TRUE));
6152 g_strfreev (fields);
6158 /* must be called with the RTSP state lock */
6159 static GstRTSPResult
6160 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
6166 /* prepare global stream caps properties */
6168 gst_structure_remove_all_fields (src->props);
6170 src->props = gst_structure_new_empty ("RTSPProperties");
6173 gst_sdp_message_dump (sdp);
6175 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
6177 /* let the app inspect and change the SDP */
6178 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_ON_SDP], 0, sdp);
6180 gst_segment_init (&src->segment, GST_FORMAT_TIME);
6182 /* parse range for duration reporting. */
6187 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
6191 /* keep track of the range and configure it in the segment */
6192 if (gst_rtspsrc_parse_range (src, range, &src->segment))
6196 /* parse clock information. This is GStreamer specific, a server can tell the
6197 * client what clock it is using and wrap that in a network clock. The
6198 * advantage of that is that we can slave to it. */
6200 const gchar *gstclock;
6203 gstclock = gst_sdp_message_get_attribute_val_n (sdp, "x-gst-clock", i);
6204 if (gstclock == NULL)
6207 /* parse the clock and expose it in the provide_clock method */
6208 if (gst_rtspsrc_parse_gst_clock (src, gstclock))
6212 /* try to find a global control attribute. Note that a '*' means that we should
6213 * do aggregate control with the current url (so we don't do anything and
6214 * leave the current connection as is) */
6216 const gchar *control;
6219 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
6220 if (control == NULL)
6223 /* only take fully qualified urls */
6224 if (g_str_has_prefix (control, "rtsp://"))
6228 g_free (src->conninfo.location);
6229 src->conninfo.location = g_strdup (control);
6230 /* make a connection for this, if there was a connection already, nothing
6232 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
6233 GST_ERROR_OBJECT (src, "could not connect");
6236 /* we need to keep the control url separate from the connection url because
6237 * the rules for constructing the media control url need it */
6238 g_free (src->control);
6239 src->control = g_strdup (control);
6242 /* create streams */
6243 n_streams = gst_sdp_message_medias_len (sdp);
6244 for (i = 0; i < n_streams; i++) {
6245 gst_rtspsrc_create_stream (src, sdp, i);
6248 src->state = GST_RTSP_STATE_INIT;
6251 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
6254 /* reset our state */
6255 src->need_range = TRUE;
6258 src->state = GST_RTSP_STATE_READY;
6265 GST_ERROR_OBJECT (src, "setup failed");
6266 gst_rtspsrc_cleanup (src);
6271 static GstRTSPResult
6272 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
6276 GstRTSPMessage request = { 0 };
6277 GstRTSPMessage response = { 0 };
6280 gchar *respcont = NULL;
6283 src->need_redirect = FALSE;
6285 /* can't continue without a valid url */
6286 if (G_UNLIKELY (src->conninfo.url == NULL)) {
6287 res = GST_RTSP_EINVAL;
6290 src->tried_url_auth = FALSE;
6292 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
6293 goto connect_failed;
6295 /* create OPTIONS */
6296 GST_DEBUG_OBJECT (src, "create options...");
6298 gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
6299 src->conninfo.url_str);
6301 goto create_request_failed;
6304 GST_DEBUG_OBJECT (src, "send options...");
6307 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
6310 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6315 if (!gst_rtspsrc_parse_methods (src, &response))
6318 /* create DESCRIBE */
6319 GST_DEBUG_OBJECT (src, "create describe...");
6321 gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
6322 src->conninfo.url_str);
6324 goto create_request_failed;
6326 /* we only accept SDP for now */
6327 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
6331 GST_DEBUG_OBJECT (src, "send describe...");
6334 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
6337 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6341 /* we only perform redirect for the describe, currently */
6342 if (src->need_redirect) {
6343 /* close connection, we don't have to send a TEARDOWN yet, ignore the
6345 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6347 gst_rtsp_message_unset (&request);
6348 gst_rtsp_message_unset (&response);
6354 /* it could be that the DESCRIBE method was not implemented */
6355 if (!src->methods & GST_RTSP_DESCRIBE)
6358 /* check if reply is SDP */
6359 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
6361 /* could not be set but since the request returned OK, we assume it
6362 * was SDP, else check it. */
6364 if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
6365 goto wrong_content_type;
6368 /* get message body and parse as SDP */
6369 gst_rtsp_message_get_body (&response, &data, &size);
6370 if (data == NULL || size == 0)
6373 GST_DEBUG_OBJECT (src, "parse SDP...");
6374 gst_sdp_message_new (sdp);
6375 gst_sdp_message_parse_buffer (data, size, *sdp);
6377 /* clean up any messages */
6378 gst_rtsp_message_unset (&request);
6379 gst_rtsp_message_unset (&response);
6386 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
6387 ("No valid RTSP URL was provided"));
6392 gchar *str = gst_rtsp_strresult (res);
6394 if (res != GST_RTSP_EINTR) {
6395 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
6396 ("Failed to connect. (%s)", str));
6398 GST_WARNING_OBJECT (src, "connect interrupted");
6403 create_request_failed:
6405 gchar *str = gst_rtsp_strresult (res);
6407 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6408 ("Could not create request. (%s)", str));
6414 /* Don't post a message - the rtsp_send method will have
6415 * taken care of it because we passed NULL for the response code */
6420 /* error was posted */
6421 res = GST_RTSP_ERROR;
6426 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6427 ("Server does not support SDP, got %s.", respcont));
6428 res = GST_RTSP_ERROR;
6433 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6434 ("Server can not provide an SDP."));
6435 res = GST_RTSP_ERROR;
6440 if (src->conninfo.connection) {
6441 GST_DEBUG_OBJECT (src, "free connection");
6442 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6444 gst_rtsp_message_unset (&request);
6445 gst_rtsp_message_unset (&response);
6450 static GstRTSPResult
6451 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
6456 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
6458 if (src->sdp == NULL) {
6459 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
6463 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
6468 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
6475 GST_WARNING_OBJECT (src, "can't get sdp");
6476 src->open_error = TRUE;
6481 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
6482 src->open_error = TRUE;
6487 static GstRTSPResult
6488 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
6490 GstRTSPMessage request = { 0 };
6491 GstRTSPMessage response = { 0 };
6492 GstRTSPResult res = GST_RTSP_OK;
6494 const gchar *control;
6496 GST_DEBUG_OBJECT (src, "TEARDOWN...");
6498 gst_rtspsrc_set_state (src, GST_STATE_READY);
6500 if (src->state < GST_RTSP_STATE_READY) {
6501 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
6508 /* construct a control url */
6509 control = get_aggregate_control (src);
6511 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
6514 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6515 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6516 const gchar *setup_url;
6517 GstRTSPConnInfo *info;
6519 /* try aggregate control first but do non-aggregate control otherwise */
6521 setup_url = control;
6522 else if ((setup_url = stream->conninfo.location) == NULL)
6525 if (src->conninfo.connection) {
6526 info = &src->conninfo;
6527 } else if (stream->conninfo.connection) {
6528 info = &stream->conninfo;
6532 if (!info->connected)
6537 gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
6539 goto create_request_failed;
6542 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
6545 gst_rtspsrc_send (src, info->connection, &request, &response,
6549 /* FIXME, parse result? */
6550 gst_rtsp_message_unset (&request);
6551 gst_rtsp_message_unset (&response);
6554 /* early exit when we did aggregate control */
6560 /* close connections */
6561 GST_DEBUG_OBJECT (src, "closing connection...");
6562 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6563 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6564 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6565 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
6569 gst_rtspsrc_cleanup (src);
6571 src->state = GST_RTSP_STATE_INVALID;
6574 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
6579 create_request_failed:
6581 gchar *str = gst_rtsp_strresult (res);
6583 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6584 ("Could not create request. (%s)", str));
6590 gchar *str = gst_rtsp_strresult (res);
6592 gst_rtsp_message_unset (&request);
6593 if (res != GST_RTSP_EINTR) {
6594 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6595 ("Could not send message. (%s)", str));
6597 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
6604 GST_DEBUG_OBJECT (src,
6605 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
6610 /* RTP-Info is of the format:
6612 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
6614 * rtptime corresponds to the timestamp for the NPT time given in the header
6615 * seqbase corresponds to the next sequence number we received. This number
6616 * indicates the first seqnum after the seek and should be used to discard
6617 * packets that are from before the seek.
6620 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
6625 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
6627 infos = g_strsplit (rtpinfo, ",", 0);
6628 for (i = 0; infos[i]; i++) {
6630 GstRTSPStream *stream;
6634 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
6636 /* init values, types of seqbase and timebase are bigger than needed so we
6637 * can store -1 as uninitialized values */
6642 /* parse url, find stream for url.
6643 * parse seq and rtptime. The seq number should be configured in the rtp
6644 * depayloader or session manager to detect gaps. Same for the rtptime, it
6645 * should be used to create an initial time newsegment. */
6646 fields = g_strsplit (infos[i], ";", 0);
6647 for (j = 0; fields[j]; j++) {
6648 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
6649 /* remove leading whitespace */
6650 fields[j] = g_strchug (fields[j]);
6651 if (g_str_has_prefix (fields[j], "url=")) {
6652 /* get the url and the stream */
6654 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
6655 } else if (g_str_has_prefix (fields[j], "seq=")) {
6656 seqbase = atoi (fields[j] + 4);
6657 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
6658 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
6661 g_strfreev (fields);
6662 /* now we need to store the values for the caps of the stream */
6663 if (stream != NULL) {
6664 GST_DEBUG_OBJECT (src,
6665 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
6666 stream, seqbase, timebase);
6668 /* we have a stream, configure detected params */
6669 stream->seqbase = seqbase;
6670 stream->timebase = timebase;
6679 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
6684 interval = strtoul (rtcp, NULL, 10);
6685 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
6690 interval *= GST_MSECOND;
6692 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6693 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6695 /* already (optionally) retrieved this when configuring manager */
6696 if (stream->session) {
6697 GObject *rtpsession = stream->session;
6699 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
6701 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
6705 /* now it happens that (Xenon) server sending this may also provide bogus
6706 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
6707 * and just use RTP-Info to sync */
6709 GObjectClass *klass;
6711 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
6712 if (g_object_class_find_property (klass, "rtcp-sync")) {
6713 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
6714 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
6720 gst_rtspsrc_get_float (const gchar * dstr)
6722 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
6724 /* canonicalise floating point string so we can handle float strings
6725 * in the form "24.930" or "24,930" irrespective of the current locale */
6726 g_strlcpy (s, dstr, sizeof (s));
6727 g_strdelimit (s, ",", '.');
6728 return g_ascii_strtod (s, NULL);
6732 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
6734 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
6736 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
6737 g_strlcpy (val_str, "now", sizeof (val_str));
6739 if (segment->position == 0) {
6740 g_strlcpy (val_str, "0", sizeof (val_str));
6742 g_ascii_dtostr (val_str, sizeof (val_str),
6743 ((gdouble) segment->position) / GST_SECOND);
6746 return g_strdup_printf ("npt=%s-", val_str);
6750 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
6754 stream->timebase = -1;
6755 stream->seqbase = -1;
6757 len = stream->ptmap->len;
6758 for (i = 0; i < len; i++) {
6759 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
6762 if (item->caps == NULL)
6765 item->caps = gst_caps_make_writable (item->caps);
6766 s = gst_caps_get_structure (item->caps, 0);
6767 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
6771 static GstRTSPResult
6772 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
6774 GstRTSPResult res = GST_RTSP_OK;
6776 if (src->state < GST_RTSP_STATE_READY) {
6777 res = GST_RTSP_ERROR;
6778 if (src->open_error) {
6779 GST_DEBUG_OBJECT (src, "the stream was in error");
6783 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
6785 if ((res = gst_rtspsrc_open (src, async)) < 0) {
6786 GST_DEBUG_OBJECT (src, "failed to open stream");
6795 static GstRTSPResult
6796 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
6798 GstRTSPMessage request = { 0 };
6799 GstRTSPMessage response = { 0 };
6800 GstRTSPResult res = GST_RTSP_OK;
6804 const gchar *control;
6806 GST_DEBUG_OBJECT (src, "PLAY...");
6808 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6811 if (!(src->methods & GST_RTSP_PLAY))
6814 if (src->state == GST_RTSP_STATE_PLAYING)
6817 if (!src->conninfo.connection || !src->conninfo.connected)
6820 /* send some dummy packets before we activate the receive in the
6822 gst_rtspsrc_send_dummy_packets (src);
6824 /* require new SR packets */
6826 g_signal_emit_by_name (src->manager, "reset-sync", NULL);
6828 gst_rtspsrc_set_state (src, GST_STATE_PLAYING);
6830 /* construct a control url */
6831 control = get_aggregate_control (src);
6833 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6834 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6835 const gchar *setup_url;
6836 GstRTSPConnection *conn;
6838 /* try aggregate control first but do non-aggregate control otherwise */
6840 setup_url = control;
6841 else if ((setup_url = stream->conninfo.location) == NULL)
6844 if (src->conninfo.connection) {
6845 conn = src->conninfo.connection;
6846 } else if (stream->conninfo.connection) {
6847 conn = stream->conninfo.connection;
6853 res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
6855 goto create_request_failed;
6857 if (src->need_range) {
6858 hval = gen_range_header (src, segment);
6860 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
6862 /* store the newsegment event so it can be sent from the streaming thread. */
6863 if (src->start_segment)
6864 gst_event_unref (src->start_segment);
6865 src->start_segment = gst_event_new_segment (&src->segment);
6868 if (segment->rate != 1.0) {
6869 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
6871 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
6873 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
6875 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
6879 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
6881 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
6884 /* seek may have silently failed as it is not supported */
6885 if (!(src->methods & GST_RTSP_PLAY)) {
6886 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
6887 /* obviously it is supported as we made it here */
6888 src->methods |= GST_RTSP_PLAY;
6889 src->seekable = FALSE;
6890 /* but there is nothing to parse in the response,
6891 * so convey we have no idea and not to expect anything particular */
6892 clear_rtp_base (src, stream);
6896 /* need to do for all streams */
6897 for (run = src->streams; run; run = g_list_next (run))
6898 clear_rtp_base (src, (GstRTSPStream *) run->data);
6900 /* NOTE the above also disables npt based eos detection */
6901 /* and below forces position to 0,
6902 * which is visible feedback we lost the plot */
6903 segment->start = segment->position = src->last_pos;
6906 gst_rtsp_message_unset (&request);
6908 /* parse RTP npt field. This is the current position in the stream (Normal
6909 * Play Time) and should be put in the NEWSEGMENT position field. */
6910 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
6912 gst_rtspsrc_parse_range (src, hval, segment);
6914 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
6915 segment->rate = 1.0;
6917 /* parse Speed header. This is the intended playback rate of the stream
6918 * and should be put in the NEWSEGMENT rate field. */
6919 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
6920 0) == GST_RTSP_OK) {
6921 segment->rate = gst_rtspsrc_get_float (hval);
6922 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
6923 &hval, 0) == GST_RTSP_OK) {
6924 segment->rate = gst_rtspsrc_get_float (hval);
6927 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
6928 * for the RTP packets. If this is not present, we assume all starts from 0...
6929 * This is info for the RTP session manager that we pass to it in caps. */
6931 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
6932 &hval, hval_idx++) == GST_RTSP_OK)
6933 gst_rtspsrc_parse_rtpinfo (src, hval);
6935 /* some servers indicate RTCP parameters in PLAY response,
6936 * rather than properly in SDP */
6937 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
6938 &hval, 0) == GST_RTSP_OK)
6939 gst_rtspsrc_handle_rtcp_interval (src, hval);
6941 gst_rtsp_message_unset (&response);
6943 /* early exit when we did aggregate control */
6947 /* configure the caps of the streams after we parsed all headers. Only reset
6948 * the manager object when we set a new Range header (we did a seek) */
6949 gst_rtspsrc_configure_caps (src, segment, src->need_range);
6951 /* set again when needed */
6952 src->need_range = FALSE;
6954 src->running = TRUE;
6955 src->base_time = -1;
6956 src->state = GST_RTSP_STATE_PLAYING;
6959 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
6960 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6961 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6962 stream->discont = TRUE;
6967 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
6974 GST_DEBUG_OBJECT (src, "failed to open stream");
6979 GST_DEBUG_OBJECT (src, "PLAY is not supported");
6984 GST_DEBUG_OBJECT (src, "we were already PLAYING");
6987 create_request_failed:
6989 gchar *str = gst_rtsp_strresult (res);
6991 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6992 ("Could not create request. (%s)", str));
6998 gchar *str = gst_rtsp_strresult (res);
7000 gst_rtsp_message_unset (&request);
7001 if (res != GST_RTSP_EINTR) {
7002 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7003 ("Could not send message. (%s)", str));
7005 GST_WARNING_OBJECT (src, "PLAY interrupted");
7012 static GstRTSPResult
7013 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
7015 GstRTSPResult res = GST_RTSP_OK;
7016 GstRTSPMessage request = { 0 };
7017 GstRTSPMessage response = { 0 };
7019 const gchar *control;
7021 GST_DEBUG_OBJECT (src, "PAUSE...");
7023 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
7026 if (!(src->methods & GST_RTSP_PAUSE))
7029 if (src->state == GST_RTSP_STATE_READY)
7032 if (!src->conninfo.connection || !src->conninfo.connected)
7035 /* construct a control url */
7036 control = get_aggregate_control (src);
7038 /* loop over the streams. We might exit the loop early when we could do an
7039 * aggregate control */
7040 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7041 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7042 GstRTSPConnection *conn;
7043 const gchar *setup_url;
7045 /* try aggregate control first but do non-aggregate control otherwise */
7047 setup_url = control;
7048 else if ((setup_url = stream->conninfo.location) == NULL)
7051 if (src->conninfo.connection) {
7052 conn = src->conninfo.connection;
7053 } else if (stream->conninfo.connection) {
7054 conn = stream->conninfo.connection;
7060 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
7061 ("Sending PAUSE request"));
7064 gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
7066 goto create_request_failed;
7068 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7071 gst_rtsp_message_unset (&request);
7072 gst_rtsp_message_unset (&response);
7074 /* exit early when we did agregate control */
7079 /* change element states now */
7080 gst_rtspsrc_set_state (src, GST_STATE_PAUSED);
7083 src->state = GST_RTSP_STATE_READY;
7087 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
7094 GST_DEBUG_OBJECT (src, "failed to open stream");
7099 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
7104 GST_DEBUG_OBJECT (src, "we were already PAUSED");
7107 create_request_failed:
7109 gchar *str = gst_rtsp_strresult (res);
7111 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7112 ("Could not create request. (%s)", str));
7118 gchar *str = gst_rtsp_strresult (res);
7120 gst_rtsp_message_unset (&request);
7121 if (res != GST_RTSP_EINTR) {
7122 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7123 ("Could not send message. (%s)", str));
7125 GST_WARNING_OBJECT (src, "PAUSE interrupted");
7133 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
7135 GstRTSPSrc *rtspsrc;
7137 rtspsrc = GST_RTSPSRC (bin);
7139 switch (GST_MESSAGE_TYPE (message)) {
7140 case GST_MESSAGE_EOS:
7141 gst_message_unref (message);
7143 case GST_MESSAGE_ELEMENT:
7145 const GstStructure *s = gst_message_get_structure (message);
7147 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
7148 gboolean ignore_timeout;
7150 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
7152 GST_OBJECT_LOCK (rtspsrc);
7153 ignore_timeout = rtspsrc->ignore_timeout;
7154 rtspsrc->ignore_timeout = TRUE;
7155 GST_OBJECT_UNLOCK (rtspsrc);
7157 /* we only act on the first udp timeout message, others are irrelevant
7158 * and can be ignored. */
7159 if (!ignore_timeout)
7160 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
7162 gst_message_unref (message);
7165 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7168 case GST_MESSAGE_ERROR:
7171 GstRTSPStream *stream;
7174 udpsrc = GST_MESSAGE_SRC (message);
7176 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
7177 GST_ELEMENT_NAME (udpsrc));
7179 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
7183 /* we ignore the RTCP udpsrc */
7184 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
7187 /* if we get error messages from the udp sources, that's not a problem as
7188 * long as not all of them error out. We also don't really know what the
7189 * problem is, the message does not give enough detail... */
7190 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
7191 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
7192 if (ret != GST_FLOW_OK)
7196 gst_message_unref (message);
7200 /* fatal but not our message, forward */
7201 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7206 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7212 /* the thread where everything happens */
7214 gst_rtspsrc_thread (GstRTSPSrc * src)
7218 GST_OBJECT_LOCK (src);
7219 cmd = src->pending_cmd;
7220 if (cmd == CMD_RECONNECT || cmd == CMD_PLAY || cmd == CMD_PAUSE
7221 || cmd == CMD_LOOP || cmd == CMD_OPEN)
7222 src->pending_cmd = CMD_LOOP;
7224 src->pending_cmd = CMD_WAIT;
7225 GST_DEBUG_OBJECT (src, "got command %d", cmd);
7227 /* we got the message command, so ensure communication is possible again */
7228 gst_rtspsrc_connection_flush (src, FALSE);
7230 src->busy_cmd = cmd;
7231 GST_OBJECT_UNLOCK (src);
7235 gst_rtspsrc_open (src, TRUE);
7238 gst_rtspsrc_play (src, &src->segment, TRUE);
7241 gst_rtspsrc_pause (src, TRUE);
7244 gst_rtspsrc_close (src, TRUE, FALSE);
7247 gst_rtspsrc_loop (src);
7250 gst_rtspsrc_reconnect (src, FALSE);
7256 GST_OBJECT_LOCK (src);
7257 /* and go back to sleep */
7258 if (src->pending_cmd == CMD_WAIT) {
7260 gst_task_pause (src->task);
7263 src->busy_cmd = CMD_WAIT;
7264 GST_OBJECT_UNLOCK (src);
7268 gst_rtspsrc_start (GstRTSPSrc * src)
7270 GST_DEBUG_OBJECT (src, "starting");
7272 GST_OBJECT_LOCK (src);
7274 src->pending_cmd = CMD_WAIT;
7276 if (src->task == NULL) {
7277 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src, NULL);
7278 if (src->task == NULL)
7281 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
7283 GST_OBJECT_UNLOCK (src);
7290 GST_OBJECT_UNLOCK (src);
7291 GST_ERROR_OBJECT (src, "failed to create task");
7297 gst_rtspsrc_stop (GstRTSPSrc * src)
7301 GST_DEBUG_OBJECT (src, "stopping");
7303 /* also cancels pending task */
7304 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_ALL);
7306 GST_OBJECT_LOCK (src);
7307 if ((task = src->task)) {
7309 GST_OBJECT_UNLOCK (src);
7311 gst_task_stop (task);
7313 /* make sure it is not running */
7314 GST_RTSP_STREAM_LOCK (src);
7315 GST_RTSP_STREAM_UNLOCK (src);
7317 /* now wait for the task to finish */
7318 gst_task_join (task);
7320 /* and free the task */
7321 gst_object_unref (GST_OBJECT (task));
7323 GST_OBJECT_LOCK (src);
7325 GST_OBJECT_UNLOCK (src);
7327 /* ensure synchronously all is closed and clean */
7328 gst_rtspsrc_close (src, FALSE, TRUE);
7333 static GstStateChangeReturn
7334 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
7336 GstRTSPSrc *rtspsrc;
7337 GstStateChangeReturn ret;
7339 rtspsrc = GST_RTSPSRC (element);
7341 switch (transition) {
7342 case GST_STATE_CHANGE_NULL_TO_READY:
7343 if (!gst_rtspsrc_start (rtspsrc))
7346 case GST_STATE_CHANGE_READY_TO_PAUSED:
7347 /* init some state */
7348 rtspsrc->cur_protocols = rtspsrc->protocols;
7349 /* first attempt, don't ignore timeouts */
7350 rtspsrc->ignore_timeout = FALSE;
7351 rtspsrc->open_error = FALSE;
7352 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
7354 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
7355 set_manager_buffer_mode (rtspsrc);
7357 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
7358 /* unblock the tcp tasks and make the loop waiting */
7359 if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
7360 /* make sure it is waiting before we send PAUSE or PLAY below */
7361 GST_RTSP_STREAM_LOCK (rtspsrc);
7362 GST_RTSP_STREAM_UNLOCK (rtspsrc);
7365 case GST_STATE_CHANGE_PAUSED_TO_READY:
7371 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
7372 if (ret == GST_STATE_CHANGE_FAILURE)
7375 switch (transition) {
7376 case GST_STATE_CHANGE_NULL_TO_READY:
7377 ret = GST_STATE_CHANGE_SUCCESS;
7379 case GST_STATE_CHANGE_READY_TO_PAUSED:
7380 ret = GST_STATE_CHANGE_NO_PREROLL;
7382 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
7383 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
7384 ret = GST_STATE_CHANGE_SUCCESS;
7386 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
7387 /* send pause request and keep the idle task around */
7388 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
7389 ret = GST_STATE_CHANGE_NO_PREROLL;
7391 case GST_STATE_CHANGE_PAUSED_TO_READY:
7392 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
7393 ret = GST_STATE_CHANGE_SUCCESS;
7395 case GST_STATE_CHANGE_READY_TO_NULL:
7396 gst_rtspsrc_stop (rtspsrc);
7397 ret = GST_STATE_CHANGE_SUCCESS;
7408 GST_DEBUG_OBJECT (rtspsrc, "start failed");
7409 return GST_STATE_CHANGE_FAILURE;
7414 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
7417 GstRTSPSrc *rtspsrc;
7419 rtspsrc = GST_RTSPSRC (element);
7421 if (GST_EVENT_IS_DOWNSTREAM (event)) {
7422 res = gst_rtspsrc_push_event (rtspsrc, event);
7424 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
7431 /*** GSTURIHANDLER INTERFACE *************************************************/
7434 gst_rtspsrc_uri_get_type (GType type)
7439 static const gchar *const *
7440 gst_rtspsrc_uri_get_protocols (GType type)
7442 static const gchar *protocols[] =
7443 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
7444 "rtsps", "rtspsu", "rtspst", "rtspsh", NULL
7451 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
7453 GstRTSPSrc *src = GST_RTSPSRC (handler);
7455 /* FIXME: make thread-safe */
7456 return g_strdup (src->conninfo.location);
7460 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
7465 GstRTSPUrl *newurl = NULL;
7466 GstSDPMessage *sdp = NULL;
7468 src = GST_RTSPSRC (handler);
7470 /* same URI, we're fine */
7471 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
7474 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
7475 if ((res = gst_sdp_message_new (&sdp) < 0))
7478 GST_DEBUG_OBJECT (src, "parsing SDP message");
7479 if ((res = gst_sdp_message_parse_uri (uri, sdp) < 0))
7483 GST_DEBUG_OBJECT (src, "parsing URI");
7484 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
7488 /* if worked, free previous and store new url object along with the original
7490 GST_DEBUG_OBJECT (src, "configuring URI");
7491 g_free (src->conninfo.location);
7492 src->conninfo.location = g_strdup (uri);
7493 gst_rtsp_url_free (src->conninfo.url);
7494 src->conninfo.url = newurl;
7495 g_free (src->conninfo.url_str);
7497 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
7499 src->conninfo.url_str = NULL;
7502 gst_sdp_message_free (src->sdp);
7504 src->from_sdp = sdp != NULL;
7506 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
7507 GST_DEBUG_OBJECT (src, "request uri is: %s",
7508 GST_STR_NULL (src->conninfo.url_str));
7515 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
7520 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", res);
7521 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7522 "Could not create SDP");
7527 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", res,
7528 GST_STR_NULL (uri));
7529 gst_sdp_message_free (sdp);
7530 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7536 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
7537 GST_STR_NULL (uri), res);
7538 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
7539 "Invalid RTSP URI");
7545 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
7547 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
7549 iface->get_type = gst_rtspsrc_uri_get_type;
7550 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
7551 iface->get_uri = gst_rtspsrc_uri_get_uri;
7552 iface->set_uri = gst_rtspsrc_uri_set_uri;