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
84 #endif /* HAVE_UNISTD_H */
90 #include <gst/net/gstnet.h>
91 #include <gst/sdp/gstsdpmessage.h>
92 #include <gst/sdp/gstmikey.h>
93 #include <gst/rtp/rtp.h>
95 #include "gst/gst-i18n-plugin.h"
97 #include "gstrtspsrc.h"
99 GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
100 #define GST_CAT_DEFAULT (rtspsrc_debug)
102 static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
105 GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
107 /* templates used internally */
108 static GstStaticPadTemplate anysrctemplate =
109 GST_STATIC_PAD_TEMPLATE ("internalsrc_%u",
112 GST_STATIC_CAPS_ANY);
114 static GstStaticPadTemplate anysinktemplate =
115 GST_STATIC_PAD_TEMPLATE ("internalsink_%u",
118 GST_STATIC_CAPS_ANY);
122 SIGNAL_HANDLE_REQUEST,
124 SIGNAL_SELECT_STREAM,
126 SIGNAL_REQUEST_RTCP_KEY,
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 enum _GstRtspSrcNtpTimeSource
170 NTP_TIME_SOURCE_UNIX,
171 NTP_TIME_SOURCE_RUNNING_TIME,
172 NTP_TIME_SOURCE_CLOCK_TIME
175 #define GST_TYPE_RTSP_SRC_NTP_TIME_SOURCE (gst_rtsp_src_ntp_time_source_get_type())
177 gst_rtsp_src_ntp_time_source_get_type (void)
179 static GType ntp_time_source_type = 0;
180 static const GEnumValue ntp_time_source_values[] = {
181 {NTP_TIME_SOURCE_NTP, "NTP time based on realtime clock", "ntp"},
182 {NTP_TIME_SOURCE_UNIX, "UNIX time based on realtime clock", "unix"},
183 {NTP_TIME_SOURCE_RUNNING_TIME,
184 "Running time based on pipeline clock",
186 {NTP_TIME_SOURCE_CLOCK_TIME, "Pipeline clock time", "clock-time"},
190 if (!ntp_time_source_type) {
191 ntp_time_source_type =
192 g_enum_register_static ("GstRTSPSrcNtpTimeSource",
193 ntp_time_source_values);
195 return ntp_time_source_type;
198 #define AES_128_KEY_LEN 16
199 #define AES_256_KEY_LEN 32
201 #define HMAC_32_KEY_LEN 4
202 #define HMAC_80_KEY_LEN 10
204 #define DEFAULT_LOCATION NULL
205 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
206 #define DEFAULT_DEBUG FALSE
207 #define DEFAULT_RETRY 20
208 #define DEFAULT_TIMEOUT 5000000
209 #define DEFAULT_UDP_BUFFER_SIZE 0x80000
210 #define DEFAULT_TCP_TIMEOUT 20000000
211 #define DEFAULT_LATENCY_MS 2000
212 #define DEFAULT_DROP_ON_LATENCY FALSE
213 #define DEFAULT_CONNECTION_SPEED 0
214 #define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
215 #define DEFAULT_DO_RTCP TRUE
216 #define DEFAULT_DO_RTSP_KEEP_ALIVE TRUE
217 #define DEFAULT_PROXY NULL
218 #define DEFAULT_RTP_BLOCKSIZE 0
219 #define DEFAULT_USER_ID NULL
220 #define DEFAULT_USER_PW NULL
221 #define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO
222 #define DEFAULT_PORT_RANGE NULL
223 #define DEFAULT_SHORT_HEADER FALSE
224 #define DEFAULT_PROBATION 2
225 #define DEFAULT_UDP_RECONNECT TRUE
226 #define DEFAULT_MULTICAST_IFACE NULL
227 #define DEFAULT_NTP_SYNC FALSE
228 #define DEFAULT_USE_PIPELINE_CLOCK FALSE
229 #define DEFAULT_TLS_VALIDATION_FLAGS G_TLS_CERTIFICATE_VALIDATE_ALL
230 #define DEFAULT_TLS_DATABASE NULL
231 #define DEFAULT_TLS_INTERACTION NULL
232 #define DEFAULT_DO_RETRANSMISSION TRUE
233 #define DEFAULT_NTP_TIME_SOURCE NTP_TIME_SOURCE_NTP
234 #define DEFAULT_USER_AGENT "GStreamer/" PACKAGE_VERSION
235 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
247 PROP_DROP_ON_LATENCY,
248 PROP_CONNECTION_SPEED,
251 PROP_DO_RTSP_KEEP_ALIVE,
260 PROP_UDP_BUFFER_SIZE,
264 PROP_MULTICAST_IFACE,
266 PROP_USE_PIPELINE_CLOCK,
268 PROP_TLS_VALIDATION_FLAGS,
270 PROP_TLS_INTERACTION,
271 PROP_DO_RETRANSMISSION,
272 PROP_NTP_TIME_SOURCE,
274 PROP_MAX_RTCP_RTP_TIME_DIFF
277 #define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
279 gst_rtsp_nat_method_get_type (void)
281 static GType rtsp_nat_method_type = 0;
282 static const GEnumValue rtsp_nat_method[] = {
283 {GST_RTSP_NAT_NONE, "None", "none"},
284 {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
288 if (!rtsp_nat_method_type) {
289 rtsp_nat_method_type =
290 g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
292 return rtsp_nat_method_type;
295 static void gst_rtspsrc_finalize (GObject * object);
297 static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
298 const GValue * value, GParamSpec * pspec);
299 static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
300 GValue * value, GParamSpec * pspec);
302 static GstClock *gst_rtspsrc_provide_clock (GstElement * element);
304 static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
305 gpointer iface_data);
307 static void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
310 static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
311 static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
313 static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
315 static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
316 GstStateChange transition);
317 static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
318 static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
320 static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
321 GstRTSPMessage * response);
323 static gboolean gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd,
325 static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
326 GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
328 static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
329 static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
331 static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
332 static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
333 gboolean only_close);
335 static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
336 const gchar * uri, GError ** error);
337 static gchar *gst_rtspsrc_uri_get_uri (GstURIHandler * handler);
339 static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
340 static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
341 static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
342 GstRTSPStream * stream, GstEvent * event);
343 static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
344 static void gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush);
352 /* commands we send to out loop to notify it of events */
353 #define CMD_OPEN (1 << 0)
354 #define CMD_PLAY (1 << 1)
355 #define CMD_PAUSE (1 << 2)
356 #define CMD_CLOSE (1 << 3)
357 #define CMD_WAIT (1 << 4)
358 #define CMD_RECONNECT (1 << 5)
359 #define CMD_LOOP (1 << 6)
361 /* mask for all commands */
362 #define CMD_ALL ((CMD_LOOP << 1) - 1)
364 #define GST_ELEMENT_PROGRESS(el, type, code, text) \
366 gchar *__txt = _gst_element_error_printf text; \
367 gst_element_post_message (GST_ELEMENT_CAST (el), \
368 gst_message_new_progress (GST_OBJECT_CAST (el), \
369 GST_PROGRESS_TYPE_ ##type, code, __txt)); \
373 static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 };
375 #define gst_rtspsrc_parent_class parent_class
376 G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
377 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
379 #ifndef GST_DISABLE_GST_DEBUG
380 static inline const char *
381 cmd_to_string (guint cmd)
405 default_select_stream (GstRTSPSrc * src, guint id, GstCaps * caps)
407 GST_DEBUG_OBJECT (src, "default handler");
412 select_stream_accum (GSignalInvocationHint * ihint,
413 GValue * return_accu, const GValue * handler_return, gpointer data)
417 myboolean = g_value_get_boolean (handler_return);
418 GST_DEBUG ("accum %d", myboolean);
419 g_value_set_boolean (return_accu, myboolean);
421 /* stop emission if FALSE */
426 gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
428 GObjectClass *gobject_class;
429 GstElementClass *gstelement_class;
430 GstBinClass *gstbin_class;
432 gobject_class = (GObjectClass *) klass;
433 gstelement_class = (GstElementClass *) klass;
434 gstbin_class = (GstBinClass *) klass;
436 GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
438 gobject_class->set_property = gst_rtspsrc_set_property;
439 gobject_class->get_property = gst_rtspsrc_get_property;
441 gobject_class->finalize = gst_rtspsrc_finalize;
443 g_object_class_install_property (gobject_class, PROP_LOCATION,
444 g_param_spec_string ("location", "RTSP Location",
445 "Location of the RTSP url to read",
446 DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
448 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
449 g_param_spec_flags ("protocols", "Protocols",
450 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
451 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
453 g_object_class_install_property (gobject_class, PROP_DEBUG,
454 g_param_spec_boolean ("debug", "Debug",
455 "Dump request and response messages to stdout",
456 DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
458 g_object_class_install_property (gobject_class, PROP_RETRY,
459 g_param_spec_uint ("retry", "Retry",
460 "Max number of retries when allocating RTP ports.",
461 0, G_MAXUINT16, DEFAULT_RETRY,
462 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
464 g_object_class_install_property (gobject_class, PROP_TIMEOUT,
465 g_param_spec_uint64 ("timeout", "Timeout",
466 "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
467 0, G_MAXUINT64, DEFAULT_TIMEOUT,
468 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
470 g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
471 g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
472 "Fail after timeout microseconds on TCP connections (0 = disabled)",
473 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
474 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
476 g_object_class_install_property (gobject_class, PROP_LATENCY,
477 g_param_spec_uint ("latency", "Buffer latency in ms",
478 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
479 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
481 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
482 g_param_spec_boolean ("drop-on-latency",
483 "Drop buffers when maximum latency is reached",
484 "Tells the jitterbuffer to never exceed the given latency in size",
485 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
487 g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
488 g_param_spec_uint64 ("connection-speed", "Connection Speed",
489 "Network connection speed in kbps (0 = unknown)",
490 0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
491 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
493 g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
494 g_param_spec_enum ("nat-method", "NAT Method",
495 "Method to use for traversing firewalls and NAT",
496 GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
497 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
500 * GstRTSPSrc:do-rtcp:
502 * Enable RTCP support. Some old server don't like RTCP and then this property
503 * needs to be set to FALSE.
505 g_object_class_install_property (gobject_class, PROP_DO_RTCP,
506 g_param_spec_boolean ("do-rtcp", "Do RTCP",
507 "Send RTCP packets, disable for old incompatible server.",
508 DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
511 * GstRTSPSrc:do-rtsp-keep-alive:
513 * Enable RTSP keep alive support. Some old server don't like RTSP
514 * keep alive and then this property needs to be set to FALSE.
516 g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
517 g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
518 "Send RTSP keep alive packets, disable for old incompatible server.",
519 DEFAULT_DO_RTSP_KEEP_ALIVE,
520 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
525 * Set the proxy parameters. This has to be a string of the format
526 * [http://][user:passwd@]host[:port].
528 g_object_class_install_property (gobject_class, PROP_PROXY,
529 g_param_spec_string ("proxy", "Proxy",
530 "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
531 DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
533 * GstRTSPSrc:proxy-id:
535 * Sets the proxy URI user id for authentication. If the URI set via the
536 * "proxy" property contains a user-id already, that will take precedence.
540 g_object_class_install_property (gobject_class, PROP_PROXY_ID,
541 g_param_spec_string ("proxy-id", "proxy-id",
542 "HTTP proxy URI user id for authentication", "",
543 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
545 * GstRTSPSrc:proxy-pw:
547 * Sets the proxy URI password for authentication. If the URI set via the
548 * "proxy" property contains a password already, that will take precedence.
552 g_object_class_install_property (gobject_class, PROP_PROXY_PW,
553 g_param_spec_string ("proxy-pw", "proxy-pw",
554 "HTTP proxy URI user password for authentication", "",
555 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
558 * GstRTSPSrc:rtp-blocksize:
560 * RTP package size to suggest to server.
562 g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
563 g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
564 "RTP package size to suggest to server (0 = disabled)",
565 0, 65536, DEFAULT_RTP_BLOCKSIZE,
566 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
568 g_object_class_install_property (gobject_class,
570 g_param_spec_string ("user-id", "user-id",
571 "RTSP location URI user id for authentication", DEFAULT_USER_ID,
572 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
573 g_object_class_install_property (gobject_class, PROP_USER_PW,
574 g_param_spec_string ("user-pw", "user-pw",
575 "RTSP location URI user password for authentication", DEFAULT_USER_PW,
576 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
579 * GstRTSPSrc:buffer-mode:
581 * Control the buffering and timestamping mode used by the jitterbuffer.
583 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
584 g_param_spec_enum ("buffer-mode", "Buffer Mode",
585 "Control the buffering algorithm in use",
586 GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
587 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
590 * GstRTSPSrc:port-range:
592 * Configure the client port numbers that can be used to recieve RTP and
595 g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
596 g_param_spec_string ("port-range", "Port range",
597 "Client port range that can be used to receive RTP and RTCP data, "
598 "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
599 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
602 * GstRTSPSrc:udp-buffer-size:
604 * Size of the kernel UDP receive buffer in bytes.
606 g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
607 g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
608 "Size of the kernel UDP receive buffer in bytes, 0=default",
609 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
610 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
613 * GstRTSPSrc:short-header:
615 * Only send the basic RTSP headers for broken encoders.
617 g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
618 g_param_spec_boolean ("short-header", "Short Header",
619 "Only send the basic RTSP headers for broken encoders",
620 DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
622 g_object_class_install_property (gobject_class, PROP_PROBATION,
623 g_param_spec_uint ("probation", "Number of probations",
624 "Consecutive packet sequence numbers to accept the source",
625 0, G_MAXUINT, DEFAULT_PROBATION,
626 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
628 g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
629 g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
630 "Reconnect to the server if RTSP connection is closed when doing UDP",
631 DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
633 g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
634 g_param_spec_string ("multicast-iface", "Multicast Interface",
635 "The network interface on which to join the multicast group",
636 DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
638 g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
639 g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
640 "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
641 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
643 g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
644 g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
645 "Use the pipeline running-time to set the NTP time in the RTCP SR messages"
646 "(DEPRECATED: Use ntp-time-source property)",
647 DEFAULT_USE_PIPELINE_CLOCK,
648 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
650 g_object_class_install_property (gobject_class, PROP_SDES,
651 g_param_spec_boxed ("sdes", "SDES",
652 "The SDES items of this session",
653 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
656 * GstRTSPSrc::tls-validation-flags:
658 * TLS certificate validation flags used to validate server
663 g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
664 g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
665 "TLS certificate validation flags used to validate the server certificate",
666 G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS,
667 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
670 * GstRTSPSrc::tls-database:
672 * TLS database with anchor certificate authorities used to validate
673 * the server certificate.
677 g_object_class_install_property (gobject_class, PROP_TLS_DATABASE,
678 g_param_spec_object ("tls-database", "TLS database",
679 "TLS database with anchor certificate authorities used to validate the server certificate",
680 G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
683 * GstRTSPSrc::tls-interaction:
685 * A #GTlsInteraction object to be used when the connection or certificate
686 * database need to interact with the user. This will be used to prompt the
687 * user for passwords where necessary.
691 g_object_class_install_property (gobject_class, PROP_TLS_INTERACTION,
692 g_param_spec_object ("tls-interaction", "TLS interaction",
693 "A GTlsInteraction object to promt the user for password or certificate",
694 G_TYPE_TLS_INTERACTION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
697 * GstRTSPSrc::do-retransmission:
699 * Attempt to ask the server to retransmit lost packets according to RFC4588.
701 * Note: currently only works with SSRC-multiplexed retransmission streams
705 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
706 g_param_spec_boolean ("do-retransmission", "Retransmission",
707 "Ask the server to retransmit lost packets",
708 DEFAULT_DO_RETRANSMISSION,
709 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
712 * GstRTSPSrc::ntp-time-source:
714 * allows to select the time source that should be used
715 * for the NTP time in RTCP packets
719 g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
720 g_param_spec_enum ("ntp-time-source", "NTP Time Source",
721 "NTP time source for RTCP packets",
722 GST_TYPE_RTSP_SRC_NTP_TIME_SOURCE, DEFAULT_NTP_TIME_SOURCE,
723 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
726 * GstRTSPSrc::user-agent:
728 * The string to set in the User-Agent header.
732 g_object_class_install_property (gobject_class, PROP_USER_AGENT,
733 g_param_spec_string ("user-agent", "User Agent",
734 "The User-Agent string to send to the server",
735 DEFAULT_USER_AGENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
737 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
738 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
739 "Maximum amount of time in ms that the RTP time in RTCP SRs "
740 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
741 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
742 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
745 * GstRTSPSrc::handle-request:
746 * @rtspsrc: a #GstRTSPSrc
747 * @request: a #GstRTSPMessage
748 * @response: a #GstRTSPMessage
750 * Handle a server request in @request and prepare @response.
752 * This signal is called from the streaming thread, you should therefore not
753 * do any state changes on @rtspsrc because this might deadlock. If you want
754 * to modify the state as a result of this signal, post a
755 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
760 gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST] =
761 g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
762 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2,
763 G_TYPE_POINTER, G_TYPE_POINTER);
766 * GstRTSPSrc::on-sdp:
767 * @rtspsrc: a #GstRTSPSrc
768 * @sdp: a #GstSDPMessage
770 * Emited when the client has retrieved the SDP and before it configures the
771 * streams in the SDP. @sdp can be inspected and modified.
773 * This signal is called from the streaming thread, you should therefore not
774 * do any state changes on @rtspsrc because this might deadlock. If you want
775 * to modify the state as a result of this signal, post a
776 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
781 gst_rtspsrc_signals[SIGNAL_ON_SDP] =
782 g_signal_new ("on-sdp", G_TYPE_FROM_CLASS (klass), 0,
783 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
784 GST_TYPE_SDP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
787 * GstRTSPSrc::select-stream:
788 * @rtspsrc: a #GstRTSPSrc
789 * @num: the stream number
790 * @caps: the stream caps
792 * Emited before the client decides to configure the stream @num with
795 * Returns: %TRUE when the stream should be selected, %FALSE when the stream
800 gst_rtspsrc_signals[SIGNAL_SELECT_STREAM] =
801 g_signal_new_class_handler ("select-stream", G_TYPE_FROM_CLASS (klass),
802 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP,
803 (GCallback) default_select_stream, select_stream_accum, NULL,
804 g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2, G_TYPE_UINT,
807 * GstRTSPSrc::new-manager:
808 * @rtspsrc: a #GstRTSPSrc
809 * @manager: a #GstElement
811 * Emited after a new manager (like rtpbin) was created and the default
812 * properties were configured.
816 gst_rtspsrc_signals[SIGNAL_NEW_MANAGER] =
817 g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass),
818 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
819 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
822 * GstRTSPSrc::request-rtcp-key:
823 * @rtspsrc: a #GstRTSPSrc
824 * @num: the stream number
826 * Signal emited to get the crypto parameters relevant to the RTCP
827 * stream. User should provide the key and the RTCP encryption ciphers
828 * and authentication, and return them wrapped in a GstCaps.
832 gst_rtspsrc_signals[SIGNAL_REQUEST_RTCP_KEY] =
833 g_signal_new ("request-rtcp-key", G_TYPE_FROM_CLASS (klass),
834 G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
836 gstelement_class->send_event = gst_rtspsrc_send_event;
837 gstelement_class->provide_clock = gst_rtspsrc_provide_clock;
838 gstelement_class->change_state = gst_rtspsrc_change_state;
840 gst_element_class_add_pad_template (gstelement_class,
841 gst_static_pad_template_get (&rtptemplate));
843 gst_element_class_set_static_metadata (gstelement_class,
844 "RTSP packet receiver", "Source/Network",
845 "Receive data over the network via RTSP (RFC 2326)",
846 "Wim Taymans <wim@fluendo.com>, "
847 "Thijs Vermeir <thijs.vermeir@barco.com>, "
848 "Lutz Mueller <lutz@topfrose.de>");
850 gstbin_class->handle_message = gst_rtspsrc_handle_message;
852 gst_rtsp_ext_list_init ();
856 gst_rtspsrc_init (GstRTSPSrc * src)
858 src->conninfo.location = g_strdup (DEFAULT_LOCATION);
859 src->protocols = DEFAULT_PROTOCOLS;
860 src->debug = DEFAULT_DEBUG;
861 src->retry = DEFAULT_RETRY;
862 src->udp_timeout = DEFAULT_TIMEOUT;
863 gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
864 src->latency = DEFAULT_LATENCY_MS;
865 src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
866 src->connection_speed = DEFAULT_CONNECTION_SPEED;
867 src->nat_method = DEFAULT_NAT_METHOD;
868 src->do_rtcp = DEFAULT_DO_RTCP;
869 src->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
870 gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
871 src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
872 src->user_id = g_strdup (DEFAULT_USER_ID);
873 src->user_pw = g_strdup (DEFAULT_USER_PW);
874 src->buffer_mode = DEFAULT_BUFFER_MODE;
875 src->client_port_range.min = 0;
876 src->client_port_range.max = 0;
877 src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
878 src->short_header = DEFAULT_SHORT_HEADER;
879 src->probation = DEFAULT_PROBATION;
880 src->udp_reconnect = DEFAULT_UDP_RECONNECT;
881 src->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
882 src->ntp_sync = DEFAULT_NTP_SYNC;
883 src->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
885 src->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
886 src->tls_database = DEFAULT_TLS_DATABASE;
887 src->tls_interaction = DEFAULT_TLS_INTERACTION;
888 src->do_retransmission = DEFAULT_DO_RETRANSMISSION;
889 src->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
890 src->user_agent = g_strdup (DEFAULT_USER_AGENT);
891 src->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
893 /* get a list of all extensions */
894 src->extensions = gst_rtsp_ext_list_get ();
896 /* connect to send signal */
897 gst_rtsp_ext_list_connect (src->extensions, "send",
898 (GCallback) gst_rtspsrc_send_cb, src);
900 /* protects the streaming thread in interleaved mode or the polling
901 * thread in UDP mode. */
902 g_rec_mutex_init (&src->stream_rec_lock);
904 /* protects our state changes from multiple invocations */
905 g_rec_mutex_init (&src->state_rec_lock);
907 src->state = GST_RTSP_STATE_INVALID;
909 GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
913 gst_rtspsrc_finalize (GObject * object)
917 rtspsrc = GST_RTSPSRC (object);
919 gst_rtsp_ext_list_free (rtspsrc->extensions);
920 g_free (rtspsrc->conninfo.location);
921 gst_rtsp_url_free (rtspsrc->conninfo.url);
922 g_free (rtspsrc->conninfo.url_str);
923 g_free (rtspsrc->user_id);
924 g_free (rtspsrc->user_pw);
925 g_free (rtspsrc->multi_iface);
926 g_free (rtspsrc->user_agent);
929 gst_sdp_message_free (rtspsrc->sdp);
932 if (rtspsrc->provided_clock)
933 gst_object_unref (rtspsrc->provided_clock);
936 gst_structure_free (rtspsrc->sdes);
938 if (rtspsrc->tls_database)
939 g_object_unref (rtspsrc->tls_database);
941 if (rtspsrc->tls_interaction)
942 g_object_unref (rtspsrc->tls_interaction);
945 g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
946 g_rec_mutex_clear (&rtspsrc->state_rec_lock);
948 G_OBJECT_CLASS (parent_class)->finalize (object);
952 gst_rtspsrc_provide_clock (GstElement * element)
954 GstRTSPSrc *src = GST_RTSPSRC (element);
957 if ((clock = src->provided_clock) != NULL)
958 gst_object_ref (clock);
963 /* a proxy string of the format [user:passwd@]host[:port] */
965 gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
969 g_free (rtsp->proxy_user);
970 rtsp->proxy_user = NULL;
971 g_free (rtsp->proxy_passwd);
972 rtsp->proxy_passwd = NULL;
973 g_free (rtsp->proxy_host);
974 rtsp->proxy_host = NULL;
975 rtsp->proxy_port = 0;
982 /* we allow http:// in front but ignore it */
983 if (g_str_has_prefix (p, "http://"))
986 at = strchr (p, '@');
988 /* look for user:passwd */
989 col = strchr (proxy, ':');
990 if (col == NULL || col > at)
993 rtsp->proxy_user = g_strndup (p, col - p);
995 rtsp->proxy_passwd = g_strndup (col, at - col);
1000 if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
1001 rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
1002 if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
1003 rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
1004 if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
1005 GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
1006 GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
1009 col = strchr (p, ':');
1012 /* everything before the colon is the hostname */
1013 rtsp->proxy_host = g_strndup (p, col - p);
1015 rtsp->proxy_port = strtoul (p, (char **) &p, 10);
1017 rtsp->proxy_host = g_strdup (p);
1018 rtsp->proxy_port = 8080;
1024 gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
1026 rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
1027 rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
1030 rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
1032 rtspsrc->ptcp_timeout = NULL;
1036 gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
1039 GstRTSPSrc *rtspsrc;
1041 rtspsrc = GST_RTSPSRC (object);
1045 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
1046 g_value_get_string (value), NULL);
1048 case PROP_PROTOCOLS:
1049 rtspsrc->protocols = g_value_get_flags (value);
1052 rtspsrc->debug = g_value_get_boolean (value);
1055 rtspsrc->retry = g_value_get_uint (value);
1058 rtspsrc->udp_timeout = g_value_get_uint64 (value);
1060 case PROP_TCP_TIMEOUT:
1061 gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
1064 rtspsrc->latency = g_value_get_uint (value);
1066 case PROP_DROP_ON_LATENCY:
1067 rtspsrc->drop_on_latency = g_value_get_boolean (value);
1069 case PROP_CONNECTION_SPEED:
1070 rtspsrc->connection_speed = g_value_get_uint64 (value);
1072 case PROP_NAT_METHOD:
1073 rtspsrc->nat_method = g_value_get_enum (value);
1076 rtspsrc->do_rtcp = g_value_get_boolean (value);
1078 case PROP_DO_RTSP_KEEP_ALIVE:
1079 rtspsrc->do_rtsp_keep_alive = g_value_get_boolean (value);
1082 gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
1085 g_free (rtspsrc->prop_proxy_id);
1086 rtspsrc->prop_proxy_id = g_value_dup_string (value);
1089 g_free (rtspsrc->prop_proxy_pw);
1090 rtspsrc->prop_proxy_pw = g_value_dup_string (value);
1092 case PROP_RTP_BLOCKSIZE:
1093 rtspsrc->rtp_blocksize = g_value_get_uint (value);
1096 g_free (rtspsrc->user_id);
1097 rtspsrc->user_id = g_value_dup_string (value);
1100 g_free (rtspsrc->user_pw);
1101 rtspsrc->user_pw = g_value_dup_string (value);
1103 case PROP_BUFFER_MODE:
1104 rtspsrc->buffer_mode = g_value_get_enum (value);
1106 case PROP_PORT_RANGE:
1110 str = g_value_get_string (value);
1111 if (sscanf (str, "%u-%u", &rtspsrc->client_port_range.min,
1112 &rtspsrc->client_port_range.max) != 2) {
1113 rtspsrc->client_port_range.min = 0;
1114 rtspsrc->client_port_range.max = 0;
1118 case PROP_UDP_BUFFER_SIZE:
1119 rtspsrc->udp_buffer_size = g_value_get_int (value);
1121 case PROP_SHORT_HEADER:
1122 rtspsrc->short_header = g_value_get_boolean (value);
1124 case PROP_PROBATION:
1125 rtspsrc->probation = g_value_get_uint (value);
1127 case PROP_UDP_RECONNECT:
1128 rtspsrc->udp_reconnect = g_value_get_boolean (value);
1130 case PROP_MULTICAST_IFACE:
1131 g_free (rtspsrc->multi_iface);
1133 if (g_value_get_string (value) == NULL)
1134 rtspsrc->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
1136 rtspsrc->multi_iface = g_value_dup_string (value);
1139 rtspsrc->ntp_sync = g_value_get_boolean (value);
1141 case PROP_USE_PIPELINE_CLOCK:
1142 rtspsrc->use_pipeline_clock = g_value_get_boolean (value);
1145 rtspsrc->sdes = g_value_dup_boxed (value);
1147 case PROP_TLS_VALIDATION_FLAGS:
1148 rtspsrc->tls_validation_flags = g_value_get_flags (value);
1150 case PROP_TLS_DATABASE:
1151 g_clear_object (&rtspsrc->tls_database);
1152 rtspsrc->tls_database = g_value_dup_object (value);
1154 case PROP_TLS_INTERACTION:
1155 g_clear_object (&rtspsrc->tls_interaction);
1156 rtspsrc->tls_interaction = g_value_dup_object (value);
1158 case PROP_DO_RETRANSMISSION:
1159 rtspsrc->do_retransmission = g_value_get_boolean (value);
1161 case PROP_NTP_TIME_SOURCE:
1162 rtspsrc->ntp_time_source = g_value_get_enum (value);
1164 case PROP_USER_AGENT:
1165 g_free (rtspsrc->user_agent);
1166 rtspsrc->user_agent = g_value_dup_string (value);
1168 case PROP_MAX_RTCP_RTP_TIME_DIFF:
1169 rtspsrc->max_rtcp_rtp_time_diff = g_value_get_int (value);
1172 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1178 gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
1181 GstRTSPSrc *rtspsrc;
1183 rtspsrc = GST_RTSPSRC (object);
1187 g_value_set_string (value, rtspsrc->conninfo.location);
1189 case PROP_PROTOCOLS:
1190 g_value_set_flags (value, rtspsrc->protocols);
1193 g_value_set_boolean (value, rtspsrc->debug);
1196 g_value_set_uint (value, rtspsrc->retry);
1199 g_value_set_uint64 (value, rtspsrc->udp_timeout);
1201 case PROP_TCP_TIMEOUT:
1205 timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
1206 rtspsrc->tcp_timeout.tv_usec;
1207 g_value_set_uint64 (value, timeout);
1211 g_value_set_uint (value, rtspsrc->latency);
1213 case PROP_DROP_ON_LATENCY:
1214 g_value_set_boolean (value, rtspsrc->drop_on_latency);
1216 case PROP_CONNECTION_SPEED:
1217 g_value_set_uint64 (value, rtspsrc->connection_speed);
1219 case PROP_NAT_METHOD:
1220 g_value_set_enum (value, rtspsrc->nat_method);
1223 g_value_set_boolean (value, rtspsrc->do_rtcp);
1225 case PROP_DO_RTSP_KEEP_ALIVE:
1226 g_value_set_boolean (value, rtspsrc->do_rtsp_keep_alive);
1232 if (rtspsrc->proxy_host) {
1234 g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
1238 g_value_take_string (value, str);
1242 g_value_set_string (value, rtspsrc->prop_proxy_id);
1245 g_value_set_string (value, rtspsrc->prop_proxy_pw);
1247 case PROP_RTP_BLOCKSIZE:
1248 g_value_set_uint (value, rtspsrc->rtp_blocksize);
1251 g_value_set_string (value, rtspsrc->user_id);
1254 g_value_set_string (value, rtspsrc->user_pw);
1256 case PROP_BUFFER_MODE:
1257 g_value_set_enum (value, rtspsrc->buffer_mode);
1259 case PROP_PORT_RANGE:
1263 if (rtspsrc->client_port_range.min != 0) {
1264 str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
1265 rtspsrc->client_port_range.max);
1269 g_value_take_string (value, str);
1272 case PROP_UDP_BUFFER_SIZE:
1273 g_value_set_int (value, rtspsrc->udp_buffer_size);
1275 case PROP_SHORT_HEADER:
1276 g_value_set_boolean (value, rtspsrc->short_header);
1278 case PROP_PROBATION:
1279 g_value_set_uint (value, rtspsrc->probation);
1281 case PROP_UDP_RECONNECT:
1282 g_value_set_boolean (value, rtspsrc->udp_reconnect);
1284 case PROP_MULTICAST_IFACE:
1285 g_value_set_string (value, rtspsrc->multi_iface);
1288 g_value_set_boolean (value, rtspsrc->ntp_sync);
1290 case PROP_USE_PIPELINE_CLOCK:
1291 g_value_set_boolean (value, rtspsrc->use_pipeline_clock);
1294 g_value_set_boxed (value, rtspsrc->sdes);
1296 case PROP_TLS_VALIDATION_FLAGS:
1297 g_value_set_flags (value, rtspsrc->tls_validation_flags);
1299 case PROP_TLS_DATABASE:
1300 g_value_set_object (value, rtspsrc->tls_database);
1302 case PROP_TLS_INTERACTION:
1303 g_value_set_object (value, rtspsrc->tls_interaction);
1305 case PROP_DO_RETRANSMISSION:
1306 g_value_set_boolean (value, rtspsrc->do_retransmission);
1308 case PROP_NTP_TIME_SOURCE:
1309 g_value_set_enum (value, rtspsrc->ntp_time_source);
1311 case PROP_USER_AGENT:
1312 g_value_set_string (value, rtspsrc->user_agent);
1314 case PROP_MAX_RTCP_RTP_TIME_DIFF:
1315 g_value_set_int (value, rtspsrc->max_rtcp_rtp_time_diff);
1318 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1324 find_stream_by_id (GstRTSPStream * stream, gint * id)
1326 if (stream->id == *id)
1333 find_stream_by_channel (GstRTSPStream * stream, gint * channel)
1335 if (stream->channel[0] == *channel || stream->channel[1] == *channel)
1342 find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
1344 GstElement *src = (GstElement *) a;
1346 if (stream->udpsrc[0] == src)
1348 if (stream->udpsrc[1] == src)
1355 find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
1357 if (stream->conninfo.location) {
1358 /* check qualified setup_url */
1359 if (!strcmp (stream->conninfo.location, (gchar *) a))
1362 if (stream->control_url) {
1363 /* check original control_url */
1364 if (!strcmp (stream->control_url, (gchar *) a))
1367 /* check if qualified setup_url ends with string */
1368 if (g_str_has_suffix (stream->control_url, (gchar *) a))
1375 static GstRTSPStream *
1376 find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
1380 /* find and get stream */
1381 if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
1382 return (GstRTSPStream *) lstream->data;
1387 static const GstSDPBandwidth *
1388 gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1389 const GstSDPMedia * media, const gchar * type)
1393 /* first look in the media specific section */
1394 len = gst_sdp_media_bandwidths_len (media);
1395 for (i = 0; i < len; i++) {
1396 const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
1398 if (strcmp (bw->bwtype, type) == 0)
1401 /* then look in the message specific section */
1402 len = gst_sdp_message_bandwidths_len (sdp);
1403 for (i = 0; i < len; i++) {
1404 const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
1406 if (strcmp (bw->bwtype, type) == 0)
1413 gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1414 const GstSDPMedia * media, GstRTSPStream * stream)
1416 const GstSDPBandwidth *bw;
1418 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
1419 stream->as_bandwidth = bw->bandwidth;
1421 stream->as_bandwidth = -1;
1423 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
1424 stream->rr_bandwidth = bw->bandwidth;
1426 stream->rr_bandwidth = -1;
1428 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
1429 stream->rs_bandwidth = bw->bandwidth;
1431 stream->rs_bandwidth = -1;
1435 gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
1436 const GstSDPConnection * conn)
1438 if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
1441 if (conn->addrtype == NULL)
1444 /* check for IPV6 */
1445 if (strcmp (conn->addrtype, "IP4") == 0)
1446 stream->is_ipv6 = FALSE;
1447 else if (strcmp (conn->addrtype, "IP6") == 0)
1448 stream->is_ipv6 = TRUE;
1453 g_free (stream->destination);
1454 stream->destination = g_strdup (conn->address);
1456 /* check for multicast */
1457 stream->is_multicast =
1458 gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
1460 stream->ttl = conn->ttl;
1463 /* Go over the connections for a stream.
1464 * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
1466 * - If we are dealing with a localhost address, we disable multicast
1469 gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
1470 const GstSDPMedia * media, GstRTSPStream * stream)
1472 const GstSDPConnection *conn;
1475 /* first look in the media specific section */
1476 len = gst_sdp_media_connections_len (media);
1477 for (i = 0; i < len; i++) {
1478 conn = gst_sdp_media_get_connection (media, i);
1480 gst_rtspsrc_do_stream_connection (src, stream, conn);
1482 /* then look in the message specific section */
1483 if ((conn = gst_sdp_message_get_connection (sdp))) {
1484 gst_rtspsrc_do_stream_connection (src, stream, conn);
1488 /* m=<media> <UDP port> RTP/AVP <payload>
1491 gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
1492 const GstSDPMedia * media, GstRTSPStream * stream)
1496 GstCaps *global_caps;
1499 proto = gst_sdp_media_get_proto (media);
1503 if (g_str_equal (proto, "RTP/AVP"))
1504 stream->profile = GST_RTSP_PROFILE_AVP;
1505 else if (g_str_equal (proto, "RTP/SAVP"))
1506 stream->profile = GST_RTSP_PROFILE_SAVP;
1507 else if (g_str_equal (proto, "RTP/AVPF"))
1508 stream->profile = GST_RTSP_PROFILE_AVPF;
1509 else if (g_str_equal (proto, "RTP/SAVPF"))
1510 stream->profile = GST_RTSP_PROFILE_SAVPF;
1514 /* Parse global SDP attributes once */
1515 global_caps = gst_caps_new_empty_simple ("application/x-unknown");
1516 GST_DEBUG ("mapping sdp session level attributes to caps");
1517 gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, global_caps);
1518 GST_DEBUG ("mapping sdp media level attributes to caps");
1519 gst_rtspsrc_sdp_attributes_to_caps (media->attributes, global_caps);
1521 len = gst_sdp_media_formats_len (media);
1522 for (i = 0; i < len; i++) {
1524 GstCaps *caps, *outcaps;
1529 pt = atoi (gst_sdp_media_get_format (media, i));
1531 GST_DEBUG_OBJECT (src, " looking at %d pt: %d", i, pt);
1534 caps = gst_rtspsrc_media_to_caps (pt, media);
1536 GST_WARNING_OBJECT (src, " skipping pt %d without caps", pt);
1540 /* do some tweaks */
1541 s = gst_caps_get_structure (caps, 0);
1542 if ((enc = gst_structure_get_string (s, "encoding-name"))) {
1543 stream->is_real = (strstr (enc, "-REAL") != NULL);
1544 if (strcmp (enc, "X-ASF-PF") == 0)
1545 stream->container = TRUE;
1548 /* Merge in global caps */
1549 /* Intersect will merge in missing fields to the current caps */
1550 outcaps = gst_caps_intersect (caps, global_caps);
1551 gst_caps_unref (caps);
1553 /* the first pt will be the default */
1554 if (stream->ptmap->len == 0)
1555 stream->default_pt = pt;
1558 item.caps = outcaps;
1560 g_array_append_val (stream->ptmap, item);
1563 gst_caps_unref (global_caps);
1568 GST_ERROR_OBJECT (src, "can't find proto in media");
1573 GST_ERROR_OBJECT (src, "unknown proto in media %s", proto);
1578 static const gchar *
1579 get_aggregate_control (GstRTSPSrc * src)
1584 base = src->control;
1585 else if (src->content_base)
1586 base = src->content_base;
1587 else if (src->conninfo.url_str)
1588 base = src->conninfo.url_str;
1596 clear_ptmap_item (PtMapItem * item)
1599 gst_caps_unref (item->caps);
1602 static GstRTSPStream *
1603 gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
1605 GstRTSPStream *stream;
1606 const gchar *control_url;
1607 const GstSDPMedia *media;
1609 /* get media, should not return NULL */
1610 media = gst_sdp_message_get_media (sdp, idx);
1614 stream = g_new0 (GstRTSPStream, 1);
1615 stream->parent = src;
1616 /* we mark the pad as not linked, we will mark it as OK when we add the pad to
1618 stream->last_ret = GST_FLOW_NOT_LINKED;
1619 stream->added = FALSE;
1620 stream->setup = FALSE;
1621 stream->skipped = FALSE;
1623 stream->eos = FALSE;
1624 stream->discont = TRUE;
1625 stream->seqbase = -1;
1626 stream->timebase = -1;
1627 stream->send_ssrc = g_random_int ();
1628 stream->profile = GST_RTSP_PROFILE_AVP;
1629 stream->ptmap = g_array_new (FALSE, FALSE, sizeof (PtMapItem));
1630 g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);
1632 /* collect bandwidth information for this steam. FIXME, configure in the RTP
1633 * session manager to scale RTCP. */
1634 gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
1636 /* collect connection info */
1637 gst_rtspsrc_collect_connections (src, sdp, media, stream);
1639 /* make the payload type map */
1640 gst_rtspsrc_collect_payloads (src, sdp, media, stream);
1642 /* collect port number */
1643 stream->port = gst_sdp_media_get_port (media);
1645 /* get control url to construct the setup url. The setup url is used to
1646 * configure the transport of the stream and is used to identity the stream in
1647 * the RTP-Info header field returned from PLAY. */
1648 control_url = gst_sdp_media_get_attribute_val (media, "control");
1649 if (control_url == NULL)
1650 control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
1652 GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
1653 GST_DEBUG_OBJECT (src, " port: %d", stream->port);
1654 GST_DEBUG_OBJECT (src, " container: %d", stream->container);
1655 GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
1657 if (control_url != NULL) {
1658 stream->control_url = g_strdup (control_url);
1659 /* Build a fully qualified url using the content_base if any or by prefixing
1660 * the original request.
1661 * If the control_url starts with a '/' or a non rtsp: protocol we will most
1662 * likely build a URL that the server will fail to understand, this is ok,
1663 * we will fail then. */
1664 if (g_str_has_prefix (control_url, "rtsp://"))
1665 stream->conninfo.location = g_strdup (control_url);
1670 if (g_strcmp0 (control_url, "*") == 0)
1673 base = get_aggregate_control (src);
1675 /* check if the base ends or control starts with / */
1676 has_slash = g_str_has_prefix (control_url, "/");
1677 has_slash = has_slash || g_str_has_suffix (base, "/");
1679 /* concatenate the two strings, insert / when not present */
1680 stream->conninfo.location =
1681 g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
1684 GST_DEBUG_OBJECT (src, " setup: %s",
1685 GST_STR_NULL (stream->conninfo.location));
1687 /* we keep track of all streams */
1688 src->streams = g_list_append (src->streams, stream);
1696 gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
1700 GST_DEBUG_OBJECT (src, "free stream %p", stream);
1702 g_array_free (stream->ptmap, TRUE);
1704 g_free (stream->destination);
1705 g_free (stream->control_url);
1706 g_free (stream->conninfo.location);
1708 for (i = 0; i < 2; i++) {
1709 if (stream->udpsrc[i]) {
1710 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
1711 gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
1712 gst_object_unref (stream->udpsrc[i]);
1714 if (stream->channelpad[i])
1715 gst_object_unref (stream->channelpad[i]);
1717 if (stream->udpsink[i]) {
1718 gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
1719 gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
1720 gst_object_unref (stream->udpsink[i]);
1723 if (stream->fakesrc) {
1724 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
1725 gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
1726 gst_object_unref (stream->fakesrc);
1728 if (stream->srcpad) {
1729 gst_pad_set_active (stream->srcpad, FALSE);
1731 gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
1733 if (stream->srtpenc)
1734 gst_object_unref (stream->srtpenc);
1735 if (stream->srtpdec)
1736 gst_object_unref (stream->srtpdec);
1737 if (stream->srtcpparams)
1738 gst_caps_unref (stream->srtcpparams);
1739 if (stream->rtcppad)
1740 gst_object_unref (stream->rtcppad);
1741 if (stream->session)
1742 g_object_unref (stream->session);
1743 if (stream->rtx_pt_map)
1744 gst_structure_free (stream->rtx_pt_map);
1749 gst_rtspsrc_cleanup (GstRTSPSrc * src)
1753 GST_DEBUG_OBJECT (src, "cleanup");
1755 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1756 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1758 gst_rtspsrc_stream_free (src, stream);
1760 g_list_free (src->streams);
1761 src->streams = NULL;
1763 if (src->manager_sig_id) {
1764 g_signal_handler_disconnect (src->manager, src->manager_sig_id);
1765 src->manager_sig_id = 0;
1767 gst_element_set_state (src->manager, GST_STATE_NULL);
1768 gst_bin_remove (GST_BIN_CAST (src), src->manager);
1769 src->manager = NULL;
1772 gst_structure_free (src->props);
1775 g_free (src->content_base);
1776 src->content_base = NULL;
1778 g_free (src->control);
1779 src->control = NULL;
1782 gst_rtsp_range_free (src->range);
1785 /* don't clear the SDP when it was used in the url */
1786 if (src->sdp && !src->from_sdp) {
1787 gst_sdp_message_free (src->sdp);
1791 src->need_segment = FALSE;
1793 if (src->provided_clock) {
1794 gst_object_unref (src->provided_clock);
1795 src->provided_clock = NULL;
1799 #define PARSE_INT(p, del, res) \
1802 p = strstr (p, del); \
1812 #define PARSE_STRING(p, del, res) \
1815 p = strstr (p, del); \
1827 #define SKIP_SPACES(p) \
1828 while (*p && g_ascii_isspace (*p)) \
1833 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1836 gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
1837 gint * rate, gchar ** params)
1841 p = (gchar *) rtpmap;
1843 PARSE_INT (p, " ", *payload);
1851 PARSE_STRING (p, "/", *name);
1852 if (*name == NULL) {
1853 GST_DEBUG ("no rate, name %s", p);
1854 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
1855 * streams seem to omit the rate. */
1862 p = strstr (p, "/");
1880 parse_keymgmt (const gchar * keymgmt, GstCaps * caps)
1882 gboolean res = FALSE;
1885 GstMIKEYMessage *msg;
1886 const GstMIKEYPayload *payload;
1887 const gchar *srtp_cipher;
1888 const gchar *srtp_auth;
1894 p = orig_value = g_strdup (keymgmt);
1898 g_free (orig_value);
1902 PARSE_STRING (p, " ", kmpid);
1903 if (kmpid == NULL || !g_str_equal (kmpid, "mikey")) {
1904 g_free (orig_value);
1907 data = g_base64_decode (p, &size);
1909 g_free (orig_value); /* Don't need this any more */
1915 msg = gst_mikey_message_new_from_data (data, size, NULL, NULL);
1920 srtp_cipher = "aes-128-icm";
1921 srtp_auth = "hmac-sha1-80";
1923 /* check the Security policy if any */
1924 if ((payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, 0))) {
1925 GstMIKEYPayloadSP *p = (GstMIKEYPayloadSP *) payload;
1928 if (p->proto != GST_MIKEY_SEC_PROTO_SRTP)
1931 len = gst_mikey_payload_sp_get_n_params (payload);
1932 for (i = 0; i < len; i++) {
1933 const GstMIKEYPayloadSPParam *param =
1934 gst_mikey_payload_sp_get_param (payload, i);
1936 switch (param->type) {
1937 case GST_MIKEY_SP_SRTP_ENC_ALG:
1938 switch (param->val[0]) {
1940 srtp_cipher = "null";
1944 srtp_cipher = "aes-128-icm";
1950 case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
1951 switch (param->val[0]) {
1952 case AES_128_KEY_LEN:
1953 srtp_cipher = "aes-128-icm";
1955 case AES_256_KEY_LEN:
1956 srtp_cipher = "aes-256-icm";
1962 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1963 switch (param->val[0]) {
1969 srtp_auth = "hmac-sha1-80";
1975 case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
1976 switch (param->val[0]) {
1977 case HMAC_32_KEY_LEN:
1978 srtp_auth = "hmac-sha1-32";
1980 case HMAC_80_KEY_LEN:
1981 srtp_auth = "hmac-sha1-80";
1987 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1989 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1997 if (!(payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_KEMAC, 0)))
2000 GstMIKEYPayloadKEMAC *p = (GstMIKEYPayloadKEMAC *) payload;
2001 const GstMIKEYPayload *sub;
2002 GstMIKEYPayloadKeyData *pkd;
2005 if (p->enc_alg != GST_MIKEY_ENC_NULL || p->mac_alg != GST_MIKEY_MAC_NULL)
2008 if (!(sub = gst_mikey_payload_kemac_get_sub (payload, 0)))
2011 if (sub->type != GST_MIKEY_PT_KEY_DATA)
2014 pkd = (GstMIKEYPayloadKeyData *) sub;
2016 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
2018 gst_caps_set_simple (caps, "srtp-key", GST_TYPE_BUFFER, buf, NULL);
2019 gst_buffer_unref (buf);
2022 gst_caps_set_simple (caps,
2023 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
2024 "srtp-auth", G_TYPE_STRING, srtp_auth,
2025 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
2026 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
2030 gst_mikey_message_unref (msg);
2036 * Mapping SDP attributes to caps
2038 * prepend 'a-' to IANA registered sdp attributes names
2039 * (ie: not prefixed with 'x-') in order to avoid
2040 * collision with gstreamer standard caps properties names
2043 gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
2045 if (attributes->len > 0) {
2049 s = gst_caps_get_structure (caps, 0);
2051 for (i = 0; i < attributes->len; i++) {
2052 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
2053 gchar *tofree, *key;
2057 /* skip some of the attribute we already handle */
2058 if (!strcmp (key, "fmtp"))
2060 if (!strcmp (key, "rtpmap"))
2062 if (!strcmp (key, "control"))
2064 if (!strcmp (key, "range"))
2066 if (!strcmp (key, "framesize"))
2068 if (g_str_equal (key, "key-mgmt")) {
2069 parse_keymgmt (attr->value, caps);
2073 /* string must be valid UTF8 */
2074 if (!g_utf8_validate (attr->value, -1, NULL))
2077 if (!g_str_has_prefix (key, "x-"))
2078 tofree = key = g_strdup_printf ("a-%s", key);
2082 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
2083 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
2089 static const gchar *
2090 rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
2099 if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
2102 if (sscanf (attr, "%d ", &val) != 1)
2112 * Mapping of caps to and from SDP fields:
2114 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
2115 * a=framesize:<payload> <width>-<height>
2116 * a=fmtp:<payload> <param>[=<value>];...
2119 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
2122 const gchar *rtpmap;
2124 const gchar *framesize;
2127 gchar *params = NULL;
2133 /* get and parse rtpmap */
2134 rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
2137 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
2139 g_warning ("error parsing rtpmap, ignoring");
2143 /* dynamic payloads need rtpmap or we fail */
2144 if (rtpmap == NULL && pt >= 96)
2147 /* check if we have a rate, if not, we need to look up the rate from the
2148 * default rates based on the payload types. */
2150 const GstRTPPayloadInfo *info;
2152 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
2153 /* dynamic types, use media and encoding_name */
2154 tmp = g_ascii_strdown (media->media, -1);
2155 info = gst_rtp_payload_info_for_name (tmp, name);
2158 /* static types, use payload type */
2159 info = gst_rtp_payload_info_for_pt (pt);
2163 if ((rate = info->clock_rate) == 0)
2166 /* we fail if we cannot find one */
2171 tmp = g_ascii_strdown (media->media, -1);
2172 caps = gst_caps_new_simple ("application/x-unknown",
2173 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
2175 s = gst_caps_get_structure (caps, 0);
2177 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
2179 /* encoding name must be upper case */
2181 tmp = g_ascii_strup (name, -1);
2182 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
2186 /* params must be lower case */
2187 if (params != NULL) {
2188 tmp = g_ascii_strdown (params, -1);
2189 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
2193 /* parse optional fmtp: field */
2194 if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
2200 /* p is now of the format <payload> <param>[=<value>];... */
2201 PARSE_INT (p, " ", payload);
2202 if (payload != -1 && payload == pt) {
2206 /* <param>[=<value>] are separated with ';' */
2207 pairs = g_strsplit (p, ";", 0);
2208 for (i = 0; pairs[i]; i++) {
2210 const gchar *val, *key;
2212 const gchar *reserved_keys[] =
2213 { "media", "payload", "clock-rate", "encoding-name",
2217 /* the key may not have a '=', the value can have other '='s */
2218 valpos = strstr (pairs[i], "=");
2220 /* we have a '=' and thus a value, remove the '=' with \0 */
2222 /* value is everything between '=' and ';'. We split the pairs at ;
2223 * boundaries so we can take the remainder of the value. Some servers
2224 * put spaces around the value which we strip off here. Alternatively
2225 * we could strip those spaces in the depayloaders should these spaces
2226 * actually carry any meaning in the future. */
2227 val = g_strstrip (valpos + 1);
2229 /* simple <param>;.. is translated into <param>=1;... */
2232 /* strip the key of spaces, convert key to lowercase but not the value. */
2233 key = g_strstrip (pairs[i]);
2235 /* skip keys from the fmtp, which we already use ourselves for the
2236 * caps. Some software is adding random things like clock-rate into
2237 * the fmtp, and we would otherwise here set a string-typed clock-rate
2238 * in the caps... and thus fail to create valid RTP caps
2240 for (j = 0; j < G_N_ELEMENTS (reserved_keys); j++) {
2241 if (g_ascii_strcasecmp (reserved_keys[j], key) == 0) {
2247 if (strlen (key) > 1) {
2248 tmp = g_ascii_strdown (key, -1);
2249 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
2257 /* parse framesize: field */
2258 if ((framesize = gst_sdp_media_get_attribute_val (media, "framesize"))) {
2261 /* p is now of the format <payload> <width>-<height> */
2262 p = (gchar *) framesize;
2264 PARSE_INT (p, " ", payload);
2265 if (payload != -1 && payload == pt) {
2266 gst_structure_set (s, "a-framesize", G_TYPE_STRING, p, NULL);
2274 g_warning ("rtpmap type not given for dynamic payload %d", pt);
2279 g_warning ("rate unknown for payload type %d", pt);
2285 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
2286 gint * rtpport, gint * rtcpport)
2289 GstStateChangeReturn ret;
2290 GstElement *udpsrc0, *udpsrc1;
2291 gint tmp_rtp, tmp_rtcp;
2295 src = stream->parent;
2301 /* Start at next port */
2302 tmp_rtp = src->next_port_num;
2304 if (stream->is_ipv6)
2305 host = "udp://[::0]";
2307 host = "udp://0.0.0.0";
2309 /* try to allocate 2 UDP ports, the RTP port should be an even
2310 * number and the RTCP port should be the next (uneven) port */
2313 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
2314 tmp_rtp >= src->client_port_range.max)
2317 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2318 if (udpsrc0 == NULL)
2319 goto no_udp_protocol;
2320 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
2322 if (src->udp_buffer_size != 0)
2323 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
2326 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
2327 if (ret == GST_STATE_CHANGE_FAILURE) {
2329 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
2332 if (++count > src->retry)
2335 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2336 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2337 gst_object_unref (udpsrc0);
2340 GST_DEBUG_OBJECT (src, "retry %d", count);
2343 goto no_udp_protocol;
2346 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
2347 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
2349 /* check if port is even */
2350 if ((tmp_rtp & 0x01) != 0) {
2351 /* port not even, close and allocate another */
2352 if (++count > src->retry)
2355 GST_DEBUG_OBJECT (src, "RTP port not even");
2357 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2358 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2359 gst_object_unref (udpsrc0);
2362 GST_DEBUG_OBJECT (src, "retry %d", count);
2367 /* allocate port+1 for RTCP now */
2368 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2369 if (udpsrc1 == NULL)
2370 goto no_udp_rtcp_protocol;
2373 tmp_rtcp = tmp_rtp + 1;
2374 if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
2377 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
2379 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
2380 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
2381 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
2382 if (ret == GST_STATE_CHANGE_FAILURE) {
2383 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
2385 if (++count > src->retry)
2388 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2389 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2390 gst_object_unref (udpsrc0);
2393 GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
2394 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2395 gst_object_unref (udpsrc1);
2399 GST_DEBUG_OBJECT (src, "retry %d", count);
2403 /* all fine, do port check */
2404 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
2405 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
2407 /* this should not happen... */
2408 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
2411 /* we keep these elements, we configure all in configure_transport when the
2412 * server told us to really use the UDP ports. */
2413 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
2414 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
2415 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
2416 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
2418 /* keep track of next available port number when we have a range
2420 if (src->next_port_num != 0)
2421 src->next_port_num = tmp_rtcp + 1;
2428 GST_DEBUG_OBJECT (src, "could not get UDP source");
2433 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
2437 no_udp_rtcp_protocol:
2439 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
2444 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
2445 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
2451 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2452 gst_object_unref (udpsrc0);
2455 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2456 gst_object_unref (udpsrc1);
2463 gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
2468 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
2470 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2471 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2474 for (i = 0; i < 2; i++) {
2475 if (stream->udpsrc[i])
2476 gst_element_set_state (stream->udpsrc[i], state);
2482 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
2489 event = gst_event_new_flush_start ();
2490 GST_DEBUG_OBJECT (src, "start flush");
2492 state = GST_STATE_PAUSED;
2494 event = gst_event_new_flush_stop (FALSE);
2495 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
2498 state = GST_STATE_PLAYING;
2500 state = GST_STATE_PAUSED;
2502 gst_rtspsrc_push_event (src, event);
2503 gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
2504 gst_rtspsrc_set_state (src, state);
2507 static GstRTSPResult
2508 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
2509 GstRTSPMessage * message, GTimeVal * timeout)
2514 ret = gst_rtsp_connection_send (conn, message, timeout);
2516 ret = GST_RTSP_ERROR;
2521 static GstRTSPResult
2522 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
2523 GstRTSPMessage * message, GTimeVal * timeout)
2528 ret = gst_rtsp_connection_receive (conn, message, timeout);
2530 ret = GST_RTSP_ERROR;
2536 gst_rtspsrc_get_position (GstRTSPSrc * src)
2541 query = gst_query_new_position (GST_FORMAT_TIME);
2542 /* should be known somewhere down the stream (e.g. jitterbuffer) */
2543 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2544 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2548 if (stream->srcpad) {
2549 if (gst_pad_query (stream->srcpad, query)) {
2550 gst_query_parse_position (query, &fmt, &pos);
2551 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
2552 GST_TIME_ARGS (pos));
2553 src->last_pos = pos;
2563 gst_query_unref (query);
2567 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
2572 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
2574 gboolean flush, skip;
2577 GstSegment seeksegment = { 0, };
2581 GST_DEBUG_OBJECT (src, "doing seek with event");
2583 gst_event_parse_seek (event, &rate, &format, &flags,
2584 &cur_type, &cur, &stop_type, &stop);
2586 /* no negative rates yet */
2590 /* we need TIME format */
2591 if (format != src->segment.format)
2594 GST_DEBUG_OBJECT (src, "doing seek without event");
2596 cur_type = GST_SEEK_TYPE_SET;
2597 stop_type = GST_SEEK_TYPE_SET;
2600 /* get flush flag */
2601 flush = flags & GST_SEEK_FLAG_FLUSH;
2602 skip = flags & GST_SEEK_FLAG_SKIP;
2604 /* now we need to make sure the streaming thread is stopped. We do this by
2605 * either sending a FLUSH_START event downstream which will cause the
2606 * streaming thread to stop with a WRONG_STATE.
2607 * For a non-flushing seek we simply pause the task, which will happen as soon
2608 * as it completes one iteration (and thus might block when the sink is
2609 * blocking in preroll). */
2611 GST_DEBUG_OBJECT (src, "starting flush");
2612 gst_rtspsrc_flush (src, TRUE, FALSE);
2615 gst_task_pause (src->task);
2619 /* we should now be able to grab the streaming thread because we stopped it
2620 * with the above flush/pause code */
2621 GST_RTSP_STREAM_LOCK (src);
2623 GST_DEBUG_OBJECT (src, "stopped streaming");
2625 /* stop flushing the rtsp connection so we can send PAUSE/PLAY below */
2626 gst_rtspsrc_connection_flush (src, FALSE);
2628 /* copy segment, we need this because we still need the old
2629 * segment when we close the current segment. */
2630 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
2632 /* configure the seek parameters in the seeksegment. We will then have the
2633 * right values in the segment to perform the seek */
2635 GST_DEBUG_OBJECT (src, "configuring seek");
2636 gst_segment_do_seek (&seeksegment, rate, format, flags,
2637 cur_type, cur, stop_type, stop, &update);
2640 /* figure out the last position we need to play. If it's configured (stop !=
2641 * -1), use that, else we play until the total duration of the file */
2642 if ((stop = seeksegment.stop) == -1)
2643 stop = seeksegment.duration;
2645 playing = (src->state == GST_RTSP_STATE_PLAYING);
2647 /* if we were playing, pause first */
2649 /* obtain current position in case seek fails */
2650 gst_rtspsrc_get_position (src);
2651 gst_rtspsrc_pause (src, FALSE);
2655 src->state = GST_RTSP_STATE_SEEKING;
2657 /* PLAY will add the range header now. */
2658 src->need_range = TRUE;
2660 /* and continue playing */
2662 gst_rtspsrc_play (src, &seeksegment, FALSE);
2664 /* prepare for streaming again */
2666 /* if we started flush, we stop now */
2667 GST_DEBUG_OBJECT (src, "stopping flush");
2668 gst_rtspsrc_flush (src, FALSE, playing);
2671 /* now we did the seek and can activate the new segment values */
2672 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
2674 /* if we're doing a segment seek, post a SEGMENT_START message */
2675 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2676 gst_element_post_message (GST_ELEMENT_CAST (src),
2677 gst_message_new_segment_start (GST_OBJECT_CAST (src),
2678 src->segment.format, src->segment.position));
2681 /* now create the newsegment */
2682 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
2683 " to %" G_GINT64_FORMAT, src->segment.position, stop);
2686 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
2687 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2688 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2689 stream->discont = TRUE;
2692 GST_RTSP_STREAM_UNLOCK (src);
2699 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
2704 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
2710 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
2714 gboolean res = TRUE;
2717 src = GST_RTSPSRC_CAST (parent);
2719 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
2720 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
2722 switch (GST_EVENT_TYPE (event)) {
2723 case GST_EVENT_SEEK:
2724 res = gst_rtspsrc_perform_seek (src, event);
2728 case GST_EVENT_NAVIGATION:
2729 case GST_EVENT_LATENCY:
2737 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
2738 res = gst_pad_send_event (target, event);
2739 gst_object_unref (target);
2741 gst_event_unref (event);
2744 gst_event_unref (event);
2750 /* this is the final event function we receive on the internal source pad when
2751 * we deal with TCP connections */
2753 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
2758 GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
2760 switch (GST_EVENT_TYPE (event)) {
2761 case GST_EVENT_SEEK:
2763 case GST_EVENT_NAVIGATION:
2764 case GST_EVENT_LATENCY:
2766 gst_event_unref (event);
2773 /* this is the final query function we receive on the internal source pad when
2774 * we deal with TCP connections */
2776 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
2780 gboolean res = TRUE;
2782 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
2784 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2785 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2787 switch (GST_QUERY_TYPE (query)) {
2788 case GST_QUERY_POSITION:
2793 case GST_QUERY_DURATION:
2797 gst_query_parse_duration (query, &format, NULL);
2800 case GST_FORMAT_TIME:
2801 gst_query_set_duration (query, format, src->segment.duration);
2809 case GST_QUERY_LATENCY:
2811 /* we are live with a min latency of 0 and unlimited max latency, this
2812 * result will be updated by the session manager if there is any. */
2813 gst_query_set_latency (query, TRUE, 0, -1);
2823 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2825 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2829 gboolean res = FALSE;
2831 src = GST_RTSPSRC_CAST (parent);
2833 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2834 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2836 switch (GST_QUERY_TYPE (query)) {
2837 case GST_QUERY_DURATION:
2841 gst_query_parse_duration (query, &format, NULL);
2844 case GST_FORMAT_TIME:
2845 gst_query_set_duration (query, format, src->segment.duration);
2853 case GST_QUERY_SEEKING:
2857 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2858 if (format == GST_FORMAT_TIME) {
2860 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2862 /* seeking without duration is unlikely */
2863 seekable = seekable && src->seekable && src->segment.duration &&
2864 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2866 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, 0,
2867 src->segment.duration);
2876 uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
2878 gst_query_set_uri (query, uri);
2886 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2888 /* forward the query to the proxy target pad */
2890 res = gst_pad_query (target, query);
2891 gst_object_unref (target);
2900 /* callback for RTCP messages to be sent to the server when operating in TCP
2902 static GstFlowReturn
2903 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2906 GstRTSPStream *stream;
2907 GstFlowReturn res = GST_FLOW_OK;
2912 GstRTSPMessage message = { 0 };
2913 GstRTSPConnection *conn;
2915 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2916 src = stream->parent;
2918 gst_buffer_map (buffer, &map, GST_MAP_READ);
2922 gst_rtsp_message_init_data (&message, stream->channel[1]);
2924 /* lend the body data to the message */
2925 gst_rtsp_message_take_body (&message, data, size);
2927 if (stream->conninfo.connection)
2928 conn = stream->conninfo.connection;
2930 conn = src->conninfo.connection;
2932 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2933 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2934 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2936 /* and steal it away again because we will free it when unreffing the
2938 gst_rtsp_message_steal_body (&message, &data, &size);
2939 gst_rtsp_message_unset (&message);
2941 gst_buffer_unmap (buffer, &map);
2942 gst_buffer_unref (buffer);
2947 static GstPadProbeReturn
2948 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2950 GstRTSPSrc *src = user_data;
2952 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2953 GST_DEBUG_PAD_NAME (pad));
2955 /* activate the streams */
2956 GST_OBJECT_LOCK (src);
2957 if (!src->need_activate)
2960 src->need_activate = FALSE;
2961 GST_OBJECT_UNLOCK (src);
2963 gst_rtspsrc_activate_streams (src);
2965 return GST_PAD_PROBE_OK;
2969 GST_OBJECT_UNLOCK (src);
2970 return GST_PAD_PROBE_OK;
2975 copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
2977 GstPad *gpad = GST_PAD_CAST (user_data);
2979 GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
2980 gst_pad_store_sticky_event (gpad, *event);
2985 /* this callback is called when the session manager generated a new src pad with
2986 * payloaded RTP packets. We simply ghost the pad here. */
2988 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2991 GstPadTemplate *template;
2994 GstRTSPStream *stream;
2997 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2999 GST_RTSP_STATE_LOCK (src);
3001 name = gst_object_get_name (GST_OBJECT_CAST (pad));
3002 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
3003 goto unknown_stream;
3005 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);
3007 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
3009 goto unknown_stream;
3012 stream->ssrc = ssrc;
3014 /* we'll add it later see below */
3015 stream->added = TRUE;
3017 /* check if we added all streams */
3019 for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
3020 GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;
3022 GST_DEBUG_OBJECT (src, "stream %p, container %d, added %d, setup %d",
3023 ostream, ostream->container, ostream->added, ostream->setup);
3025 /* if we find a stream for which we did a setup that is not added, we
3026 * need to wait some more */
3027 if (ostream->setup && !ostream->added) {
3032 GST_RTSP_STATE_UNLOCK (src);
3034 /* create a new pad we will use to stream to */
3035 template = gst_static_pad_template_get (&rtptemplate);
3036 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
3037 gst_object_unref (template);
3040 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
3041 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
3042 gst_pad_set_active (stream->srcpad, TRUE);
3043 gst_pad_sticky_events_foreach (pad, copy_sticky_events, stream->srcpad);
3044 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
3047 GST_DEBUG_OBJECT (src, "We added all streams");
3048 /* when we get here, all stream are added and we can fire the no-more-pads
3050 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
3058 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
3059 GST_RTSP_STATE_UNLOCK (src);
3066 stream_get_caps_for_pt (GstRTSPStream * stream, guint pt)
3070 len = stream->ptmap->len;
3071 for (i = 0; i < len; i++) {
3072 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
3080 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
3082 GstRTSPStream *stream;
3085 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
3087 GST_RTSP_STATE_LOCK (src);
3088 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
3090 goto unknown_stream;
3092 if ((caps = stream_get_caps_for_pt (stream, pt)))
3093 gst_caps_ref (caps);
3094 GST_RTSP_STATE_UNLOCK (src);
3100 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
3101 GST_RTSP_STATE_UNLOCK (src);
3107 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
3109 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
3115 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
3121 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
3127 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
3129 GstRTSPSrc *src = stream->parent;
3132 g_object_get (source, "ssrc", &ssrc, NULL);
3134 GST_DEBUG_OBJECT (src, "source %08x, stream %08x, session %u received BYE",
3135 ssrc, stream->ssrc, stream->id);
3137 if (ssrc == stream->ssrc)
3138 gst_rtspsrc_do_stream_eos (src, stream);
3142 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
3144 GstRTSPSrc *src = stream->parent;
3147 g_object_get (source, "ssrc", &ssrc, NULL);
3149 GST_WARNING_OBJECT (src, "source %08x, stream %08x in session %u timed out",
3150 ssrc, stream->ssrc, stream->id);
3152 if (ssrc == stream->ssrc)
3153 gst_rtspsrc_do_stream_eos (src, stream);
3157 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
3159 GstRTSPStream *stream;
3161 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
3163 /* get stream for session */
3164 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
3166 gst_rtspsrc_do_stream_eos (src, stream);
3171 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
3173 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
3178 set_manager_buffer_mode (GstRTSPSrc * src)
3180 GObjectClass *klass;
3182 if (src->manager == NULL)
3185 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
3187 if (!g_object_class_find_property (klass, "buffer-mode"))
3190 if (src->buffer_mode != BUFFER_MODE_AUTO) {
3191 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
3196 GST_DEBUG_OBJECT (src,
3197 "auto buffering mode, have clock %" GST_PTR_FORMAT, src->provided_clock);
3199 if (src->provided_clock) {
3200 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
3202 if (clock == src->provided_clock) {
3203 GST_DEBUG_OBJECT (src, "selected synced");
3204 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SYNCED, NULL);
3207 gst_object_unref (clock);
3212 /* Otherwise fall-through and use another buffer mode */
3214 gst_object_unref (clock);
3217 GST_DEBUG_OBJECT (src, "auto buffering mode");
3218 if (src->use_buffering) {
3219 GST_DEBUG_OBJECT (src, "selected buffer");
3220 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER, NULL);
3222 GST_DEBUG_OBJECT (src, "selected slave");
3223 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
3228 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
3230 GST_DEBUG ("request key %u", ssrc);
3231 return gst_caps_ref (stream_get_caps_for_pt (stream, stream->default_pt));
3235 request_rtp_decoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
3237 GST_DEBUG ("decoder session %u, stream %p, %d", session, stream, stream->id);
3238 if (stream->id != session)
3241 if (stream->profile != GST_RTSP_PROFILE_SAVP &&
3242 stream->profile != GST_RTSP_PROFILE_SAVPF)
3245 if (stream->srtpdec == NULL) {
3248 name = g_strdup_printf ("srtpdec_%u", session);
3249 stream->srtpdec = gst_element_factory_make ("srtpdec", name);
3252 g_signal_connect (stream->srtpdec, "request-key",
3253 (GCallback) request_key, stream);
3255 return gst_object_ref (stream->srtpdec);
3259 request_rtcp_encoder (GstElement * rtpbin, guint session,
3260 GstRTSPStream * stream)
3265 GST_DEBUG ("decoder session %u, stream %p, %d", session, stream, stream->id);
3266 if (stream->id != session)
3269 if (stream->profile != GST_RTSP_PROFILE_SAVP &&
3270 stream->profile != GST_RTSP_PROFILE_SAVPF)
3273 if (stream->srtpenc == NULL) {
3276 name = g_strdup_printf ("srtpenc_%u", session);
3277 stream->srtpenc = gst_element_factory_make ("srtpenc", name);
3280 /* get RTCP crypto parameters from caps */
3281 s = gst_caps_get_structure (stream->srtcpparams, 0);
3285 GType ciphertype, authtype;
3286 GValue rtcp_cipher = G_VALUE_INIT, rtcp_auth = G_VALUE_INIT;
3288 ciphertype = g_type_from_name ("GstSrtpCipherType");
3289 authtype = g_type_from_name ("GstSrtpAuthType");
3290 g_value_init (&rtcp_cipher, ciphertype);
3291 g_value_init (&rtcp_auth, authtype);
3293 str = gst_structure_get_string (s, "srtcp-cipher");
3294 gst_value_deserialize (&rtcp_cipher, str);
3295 str = gst_structure_get_string (s, "srtcp-auth");
3296 gst_value_deserialize (&rtcp_auth, str);
3297 gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL);
3299 g_object_set_property (G_OBJECT (stream->srtpenc), "rtcp-cipher",
3301 g_object_set_property (G_OBJECT (stream->srtpenc), "rtcp-auth",
3303 g_object_set (stream->srtpenc, "key", buf, NULL);
3305 g_value_unset (&rtcp_cipher);
3306 g_value_unset (&rtcp_auth);
3307 gst_buffer_unref (buf);
3310 name = g_strdup_printf ("rtcp_sink_%d", session);
3311 pad = gst_element_get_request_pad (stream->srtpenc, name);
3313 gst_object_unref (pad);
3315 return gst_object_ref (stream->srtpenc);
3319 request_aux_receiver (GstElement * rtpbin, guint sessid, GstRTSPSrc * src)
3321 GstElement *rtx, *bin;
3324 GstRTSPStream *stream;
3326 stream = find_stream (src, &sessid, (gpointer) find_stream_by_id);
3328 GST_WARNING_OBJECT (src, "Stream %u not found", sessid);
3332 GST_INFO_OBJECT (src, "creating retransmision receiver for session %u "
3333 "with map %" GST_PTR_FORMAT, sessid, stream->rtx_pt_map);
3334 bin = gst_bin_new (NULL);
3335 rtx = gst_element_factory_make ("rtprtxreceive", NULL);
3336 g_object_set (rtx, "payload-type-map", stream->rtx_pt_map, NULL);
3337 gst_bin_add (GST_BIN (bin), rtx);
3339 pad = gst_element_get_static_pad (rtx, "src");
3340 name = g_strdup_printf ("src_%u", sessid);
3341 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
3343 gst_object_unref (pad);
3345 pad = gst_element_get_static_pad (rtx, "sink");
3346 name = g_strdup_printf ("sink_%u", sessid);
3347 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
3349 gst_object_unref (pad);
3355 add_retransmission (GstRTSPSrc * src, GstRTSPTransport * transport)
3359 gboolean do_retransmission = FALSE;
3361 if (transport->trans != GST_RTSP_TRANS_RTP)
3363 if (transport->profile != GST_RTSP_PROFILE_AVPF &&
3364 transport->profile != GST_RTSP_PROFILE_SAVPF)
3367 signal_id = g_signal_lookup ("request-aux-receiver",
3368 G_OBJECT_TYPE (src->manager));
3369 /* there's already something connected */
3370 if (g_signal_handler_find (src->manager, G_SIGNAL_MATCH_ID, signal_id, 0,
3371 NULL, NULL, NULL) != 0) {
3372 GST_DEBUG_OBJECT (src, "Not adding RTX AUX element as "
3373 "\"request-aux-receiver\" signal is "
3374 "already used by the application");
3378 /* build the retransmission payload type map */
3379 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3380 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3381 gboolean do_retransmission_stream = FALSE;
3384 if (stream->rtx_pt_map)
3385 gst_structure_free (stream->rtx_pt_map);
3386 stream->rtx_pt_map = gst_structure_new_empty ("application/x-rtp-pt-map");
3388 for (i = 0; i < stream->ptmap->len; i++) {
3389 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
3390 GstStructure *s = gst_caps_get_structure (item->caps, 0);
3391 const gchar *encoding;
3393 /* we only care about RTX streams */
3394 if ((encoding = gst_structure_get_string (s, "encoding-name"))
3395 && g_strcmp0 (encoding, "RTX") == 0) {
3396 const gchar *stream_pt_s;
3399 if (gst_structure_get_int (s, "payload", &rtx_pt)
3400 && (stream_pt_s = gst_structure_get_string (s, "apt"))) {
3403 gst_structure_set (stream->rtx_pt_map, stream_pt_s, G_TYPE_UINT,
3405 do_retransmission_stream = TRUE;
3411 if (do_retransmission_stream) {
3412 GST_DEBUG_OBJECT (src, "built retransmission payload map for stream "
3413 "id %i: %" GST_PTR_FORMAT, stream->id, stream->rtx_pt_map);
3414 do_retransmission = TRUE;
3416 GST_DEBUG_OBJECT (src, "no retransmission payload map for stream "
3417 "id %i", stream->id);
3418 gst_structure_free (stream->rtx_pt_map);
3419 stream->rtx_pt_map = NULL;
3423 if (do_retransmission) {
3424 GST_DEBUG_OBJECT (src, "Enabling retransmissions");
3426 g_object_set (src->manager, "do-retransmission", TRUE, NULL);
3428 /* enable RFC4588 retransmission handling by setting rtprtxreceive
3429 * as the "aux" element of rtpbin */
3430 g_signal_connect (src->manager, "request-aux-receiver",
3431 (GCallback) request_aux_receiver, src);
3433 GST_DEBUG_OBJECT (src,
3434 "Not enabling retransmissions as no stream had a retransmission payload map");
3438 /* try to get and configure a manager */
3440 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
3441 GstRTSPTransport * transport)
3443 const gchar *manager;
3445 GstStateChangeReturn ret;
3447 /* find a manager */
3448 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
3452 GST_DEBUG_OBJECT (src, "using manager %s", manager);
3454 /* configure the manager */
3455 if (src->manager == NULL) {
3456 GObjectClass *klass;
3458 if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
3460 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
3464 goto use_no_manager;
3466 if (!(src->manager = gst_element_factory_make (manager, "manager")))
3467 goto manager_failed;
3470 /* we manage this element */
3471 gst_element_set_locked_state (src->manager, TRUE);
3472 gst_bin_add (GST_BIN_CAST (src), src->manager);
3474 ret = gst_element_set_state (src->manager, GST_STATE_PAUSED);
3475 if (ret == GST_STATE_CHANGE_FAILURE)
3476 goto start_manager_failure;
3478 g_object_set (src->manager, "latency", src->latency, NULL);
3480 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
3482 if (g_object_class_find_property (klass, "ntp-sync")) {
3483 g_object_set (src->manager, "ntp-sync", src->ntp_sync, NULL);
3486 if (src->use_pipeline_clock) {
3487 if (g_object_class_find_property (klass, "use-pipeline-clock")) {
3488 g_object_set (src->manager, "use-pipeline-clock", TRUE, NULL);
3491 if (g_object_class_find_property (klass, "ntp-time-source")) {
3492 g_object_set (src->manager, "ntp-time-source", src->ntp_time_source,
3497 if (src->sdes && g_object_class_find_property (klass, "sdes")) {
3498 g_object_set (src->manager, "sdes", src->sdes, NULL);
3501 if (g_object_class_find_property (klass, "drop-on-latency")) {
3502 g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
3506 if (g_object_class_find_property (klass, "max-rtcp-rtp-time-diff")) {
3507 g_object_set (src->manager, "max-rtcp-rtp-time-diff",
3508 src->max_rtcp_rtp_time_diff, NULL);
3511 /* buffer mode pauses are handled by adding offsets to buffer times,
3512 * but some depayloaders may have a hard time syncing output times
3513 * with such input times, e.g. container ones, most notably ASF */
3514 /* TODO alternatives are having an event that indicates these shifts,
3515 * or having rtsp extensions provide suggestion on buffer mode */
3516 /* valid duration implies not likely live pipeline,
3517 * so slaving in jitterbuffer does not make much sense
3518 * (and might mess things up due to bursts) */
3519 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
3520 src->segment.duration && stream->container) {
3521 src->use_buffering = TRUE;
3523 src->use_buffering = FALSE;
3526 set_manager_buffer_mode (src);
3528 /* connect to signals */
3529 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
3531 src->manager_sig_id =
3532 g_signal_connect (src->manager, "pad-added",
3533 (GCallback) new_manager_pad, src);
3534 src->manager_ptmap_id =
3535 g_signal_connect (src->manager, "request-pt-map",
3536 (GCallback) request_pt_map, src);
3538 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
3541 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_NEW_MANAGER], 0,
3544 if (src->do_retransmission)
3545 add_retransmission (src, transport);
3547 g_signal_connect (src->manager, "request-rtp-decoder",
3548 (GCallback) request_rtp_decoder, stream);
3549 g_signal_connect (src->manager, "request-rtcp-decoder",
3550 (GCallback) request_rtp_decoder, stream);
3551 g_signal_connect (src->manager, "request-rtcp-encoder",
3552 (GCallback) request_rtcp_encoder, stream);
3554 /* we stream directly to the manager, get some pads. Each RTSP stream goes
3555 * into a separate RTP session. */
3556 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
3557 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
3559 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
3560 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
3563 /* now configure the bandwidth in the manager */
3564 if (g_signal_lookup ("get-internal-session",
3565 G_OBJECT_TYPE (src->manager)) != 0) {
3566 GObject *rtpsession;
3568 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
3571 GstRTPProfile rtp_profile;
3573 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
3575 stream->session = rtpsession;
3577 if (stream->as_bandwidth != -1) {
3578 GST_INFO_OBJECT (src, "setting AS: %f",
3579 (gdouble) (stream->as_bandwidth * 1000));
3580 g_object_set (rtpsession, "bandwidth",
3581 (gdouble) (stream->as_bandwidth * 1000), NULL);
3583 if (stream->rr_bandwidth != -1) {
3584 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
3585 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
3588 if (stream->rs_bandwidth != -1) {
3589 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
3590 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
3594 switch (stream->profile) {
3595 case GST_RTSP_PROFILE_AVPF:
3596 rtp_profile = GST_RTP_PROFILE_AVPF;
3598 case GST_RTSP_PROFILE_SAVP:
3599 rtp_profile = GST_RTP_PROFILE_SAVP;
3601 case GST_RTSP_PROFILE_SAVPF:
3602 rtp_profile = GST_RTP_PROFILE_SAVPF;
3604 case GST_RTSP_PROFILE_AVP:
3606 rtp_profile = GST_RTP_PROFILE_AVP;
3610 g_object_set (rtpsession, "rtp-profile", rtp_profile, NULL);
3612 g_object_set (rtpsession, "probation", src->probation, NULL);
3614 g_object_set (rtpsession, "internal-ssrc", stream->send_ssrc, NULL);
3616 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
3618 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
3620 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
3622 g_signal_connect (rtpsession, "on-ssrc-active",
3623 (GCallback) on_ssrc_active, stream);
3634 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3639 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
3642 start_manager_failure:
3644 GST_DEBUG_OBJECT (src, "could not start session manager");
3649 /* free the UDP sources allocated when negotiating a transport.
3650 * This function is called when the server negotiated to a transport where the
3651 * UDP sources are not needed anymore, such as TCP or multicast. */
3653 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
3657 for (i = 0; i < 2; i++) {
3658 if (stream->udpsrc[i]) {
3659 GST_DEBUG ("free UDP source %d for stream %p", i, stream);
3660 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
3661 gst_object_unref (stream->udpsrc[i]);
3662 stream->udpsrc[i] = NULL;
3667 /* for TCP, create pads to send and receive data to and from the manager and to
3668 * intercept various events and queries
3671 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
3672 GstRTSPTransport * transport, GstPad ** outpad)
3675 GstPadTemplate *template;
3676 GstPad *pad0, *pad1;
3678 /* configure for interleaved delivery, nothing needs to be done
3679 * here, the loop function will call the chain functions of the
3680 * session manager. */
3681 stream->channel[0] = transport->interleaved.min;
3682 stream->channel[1] = transport->interleaved.max;
3683 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
3684 stream->channel[0], stream->channel[1]);
3686 /* we can remove the allocated UDP ports now */
3687 gst_rtspsrc_stream_free_udp (stream);
3689 /* no session manager, send data to srcpad directly */
3690 if (!stream->channelpad[0]) {
3691 GST_DEBUG_OBJECT (src, "no manager, creating pad");
3693 /* create a new pad we will use to stream to */
3694 name = g_strdup_printf ("stream_%u", stream->id);
3695 template = gst_static_pad_template_get (&rtptemplate);
3696 stream->channelpad[0] = gst_pad_new_from_template (template, name);
3697 gst_object_unref (template);
3700 /* set caps and activate */
3701 gst_pad_use_fixed_caps (stream->channelpad[0]);
3702 gst_pad_set_active (stream->channelpad[0], TRUE);
3704 *outpad = gst_object_ref (stream->channelpad[0]);
3706 GST_DEBUG_OBJECT (src, "using manager source pad");
3708 template = gst_static_pad_template_get (&anysrctemplate);
3710 /* allocate pads for sending the channel data into the manager */
3711 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
3712 gst_pad_link_full (pad0, stream->channelpad[0], GST_PAD_LINK_CHECK_NOTHING);
3713 gst_object_unref (stream->channelpad[0]);
3714 stream->channelpad[0] = pad0;
3715 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
3716 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
3717 gst_pad_set_element_private (pad0, src);
3718 gst_pad_set_active (pad0, TRUE);
3720 if (stream->channelpad[1]) {
3721 /* if we have a sinkpad for the other channel, create a pad and link to the
3723 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
3724 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
3725 gst_pad_link_full (pad1, stream->channelpad[1],
3726 GST_PAD_LINK_CHECK_NOTHING);
3727 gst_object_unref (stream->channelpad[1]);
3728 stream->channelpad[1] = pad1;
3729 gst_pad_set_active (pad1, TRUE);
3731 gst_object_unref (template);
3733 /* setup RTCP transport back to the server if we have to. */
3734 if (src->manager && src->do_rtcp) {
3737 template = gst_static_pad_template_get (&anysinktemplate);
3739 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
3740 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
3741 gst_pad_set_element_private (stream->rtcppad, stream);
3742 gst_pad_set_active (stream->rtcppad, TRUE);
3744 /* get session RTCP pad */
3745 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3746 pad = gst_element_get_request_pad (src->manager, name);
3751 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3752 gst_object_unref (pad);
3755 gst_object_unref (template);
3761 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
3762 GstRTSPTransport * transport, const gchar ** destination, gint * min,
3763 gint * max, guint * ttl)
3765 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3767 if (!(*destination = transport->destination))
3768 *destination = stream->destination;
3771 /* transport first */
3772 *min = transport->port.min;
3773 *max = transport->port.max;
3774 if (*min == -1 && *max == -1) {
3775 /* then try from SDP */
3776 if (stream->port != 0) {
3777 *min = stream->port;
3778 *max = stream->port + 1;
3784 if (!(*ttl = transport->ttl))
3789 /* first take the source, then the endpoint to figure out where to send
3791 if (!(*destination = transport->source)) {
3792 if (src->conninfo.connection)
3793 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
3794 else if (stream->conninfo.connection)
3796 gst_rtsp_connection_get_ip (stream->conninfo.connection);
3800 /* for unicast we only expect the ports here */
3801 *min = transport->server_port.min;
3802 *max = transport->server_port.max;
3807 /* For multicast create UDP sources and join the multicast group. */
3809 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
3810 GstRTSPTransport * transport, GstPad ** outpad)
3813 const gchar *destination;
3816 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
3818 /* we can remove the allocated UDP ports now */
3819 gst_rtspsrc_stream_free_udp (stream);
3821 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
3824 /* we need a destination now */
3825 if (destination == NULL)
3826 goto no_destination;
3828 /* we really need ports now or we won't be able to receive anything at all */
3829 if (min == -1 && max == -1)
3832 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
3833 destination, min, max);
3835 /* creating UDP source for RTP */
3837 uri = g_strdup_printf ("udp://%s:%d", destination, min);
3839 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3841 if (stream->udpsrc[0] == NULL)
3844 /* take ownership */
3845 gst_object_ref_sink (stream->udpsrc[0]);
3847 if (src->udp_buffer_size != 0)
3848 g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
3849 src->udp_buffer_size, NULL);
3851 if (src->multi_iface != NULL)
3852 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3853 src->multi_iface, NULL);
3856 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3857 gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
3860 /* creating another UDP source for RTCP */
3864 uri = g_strdup_printf ("udp://%s:%d", destination, max);
3866 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3868 if (stream->udpsrc[1] == NULL)
3871 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3872 stream->profile == GST_RTSP_PROFILE_SAVPF)
3873 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3875 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3876 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3877 gst_caps_unref (caps);
3879 /* take ownership */
3880 gst_object_ref_sink (stream->udpsrc[1]);
3882 if (src->multi_iface != NULL)
3883 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3884 src->multi_iface, NULL);
3886 gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
3893 GST_DEBUG_OBJECT (src, "no UDP source element found");
3898 GST_DEBUG_OBJECT (src, "no destination found");
3903 GST_DEBUG_OBJECT (src, "no ports found");
3908 /* configure the remainder of the UDP ports */
3910 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
3911 GstRTSPTransport * transport, GstPad ** outpad)
3913 /* we manage the UDP elements now. For unicast, the UDP sources where
3914 * allocated in the stream when we suggested a transport. */
3915 if (stream->udpsrc[0]) {
3918 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3919 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
3921 GST_DEBUG_OBJECT (src, "setting up UDP source");
3923 /* configure a timeout on the UDP port. When the timeout message is
3924 * posted, we assume UDP transport is not possible. We reconnect using TCP
3926 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
3927 src->udp_timeout * 1000, NULL);
3929 if ((caps = stream_get_caps_for_pt (stream, stream->default_pt)))
3930 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
3932 /* get output pad of the UDP source. */
3933 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
3935 /* save it so we can unblock */
3936 stream->blockedpad = *outpad;
3938 /* configure pad block on the pad. As soon as there is dataflow on the
3939 * UDP source, we know that UDP is not blocked by a firewall and we can
3940 * configure all the streams to let the application autoplug decoders. */
3942 gst_pad_add_probe (stream->blockedpad,
3943 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3944 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocked, src, NULL);
3946 if (stream->channelpad[0]) {
3947 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
3948 /* configure for UDP delivery, we need to connect the UDP pads to
3949 * the session plugin. */
3950 gst_pad_link_full (*outpad, stream->channelpad[0],
3951 GST_PAD_LINK_CHECK_NOTHING);
3952 gst_object_unref (*outpad);
3954 /* we connected to pad-added signal to get pads from the manager */
3956 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
3961 if (stream->udpsrc[1]) {
3964 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
3965 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
3967 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3968 stream->profile == GST_RTSP_PROFILE_SAVPF)
3969 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3971 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3972 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3973 gst_caps_unref (caps);
3975 if (stream->channelpad[1]) {
3978 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
3980 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
3981 gst_pad_link_full (pad, stream->channelpad[1],
3982 GST_PAD_LINK_CHECK_NOTHING);
3983 gst_object_unref (pad);
3985 /* leave unlinked */
3991 /* configure the UDP sink back to the server for status reports */
3993 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
3994 GstRTSPStream * stream, GstRTSPTransport * transport)
3997 gint rtp_port, rtcp_port;
3998 gboolean do_rtp, do_rtcp;
3999 const gchar *destination;
4004 /* get transport info */
4005 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
4006 &rtp_port, &rtcp_port, &ttl);
4008 /* see what we need to do */
4009 do_rtp = (rtp_port != -1);
4010 /* it's possible that the server does not want us to send RTCP in which case
4012 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
4014 /* we need a destination when we have RTP or RTCP ports */
4015 if (destination == NULL && (do_rtp || do_rtcp))
4016 goto no_destination;
4018 /* try to construct the fakesrc to the RTP port of the server to open up any
4021 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
4024 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
4025 stream->udpsink[0] =
4026 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
4028 if (stream->udpsink[0] == NULL)
4029 goto no_sink_element;
4031 /* don't join multicast group, we will have the source socket do that */
4032 /* no sync or async state changes needed */
4033 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
4034 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
4036 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
4038 if (stream->udpsrc[0]) {
4039 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
4040 * so that NAT firewalls will open a hole for us */
4041 g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
4042 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
4043 /* configure socket and make sure udpsink does not close it when shutting
4044 * down, it belongs to udpsrc after all. */
4045 g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
4046 "close-socket", FALSE, NULL);
4047 g_object_unref (socket);
4050 /* the source for the dummy packets to open up NAT */
4051 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
4052 if (stream->fakesrc == NULL)
4053 goto no_fakesrc_element;
4055 /* random data in 5 buffers, a size of 200 bytes should be fine */
4056 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
4057 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
4059 /* we don't want to consider this a sink */
4060 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
4062 /* keep everything locked */
4063 gst_element_set_locked_state (stream->udpsink[0], TRUE);
4064 gst_element_set_locked_state (stream->fakesrc, TRUE);
4066 gst_object_ref (stream->udpsink[0]);
4067 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
4068 gst_object_ref (stream->fakesrc);
4069 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
4071 gst_element_link_pads_full (stream->fakesrc, "src", stream->udpsink[0],
4072 "sink", GST_PAD_LINK_CHECK_NOTHING);
4075 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
4078 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
4079 stream->udpsink[1] =
4080 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
4082 if (stream->udpsink[1] == NULL)
4083 goto no_sink_element;
4085 /* don't join multicast group, we will have the source socket do that */
4086 /* no sync or async state changes needed */
4087 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
4088 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
4090 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
4092 if (stream->udpsrc[1]) {
4093 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
4094 * because some servers check the port number of where it sends RTCP to identify
4095 * the RTCP packets it receives */
4096 g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
4097 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
4098 /* configure socket and make sure udpsink does not close it when shutting
4099 * down, it belongs to udpsrc after all. */
4100 g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
4101 "close-socket", FALSE, NULL);
4102 g_object_unref (socket);
4105 /* we don't want to consider this a sink */
4106 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
4108 /* we keep this playing always */
4109 gst_element_set_locked_state (stream->udpsink[1], TRUE);
4110 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
4112 gst_object_ref (stream->udpsink[1]);
4113 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
4115 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
4117 /* get session RTCP pad */
4118 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
4119 pad = gst_element_get_request_pad (src->manager, name);
4124 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
4125 gst_object_unref (pad);
4134 GST_DEBUG_OBJECT (src, "no destination address specified");
4139 GST_DEBUG_OBJECT (src, "no UDP sink element found");
4144 GST_DEBUG_OBJECT (src, "no fakesrc element found");
4149 /* sets up all elements needed for streaming over the specified transport.
4150 * Does not yet expose the element pads, this will be done when there is actuall
4151 * dataflow detected, which might never happen when UDP is blocked in a
4152 * firewall, for example.
4155 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
4156 GstRTSPTransport * transport)
4159 GstPad *outpad = NULL;
4160 GstPadTemplate *template;
4162 const gchar *media_type;
4165 src = stream->parent;
4167 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
4169 /* get the proper media type for this stream now */
4170 if (gst_rtsp_transport_get_media_type (transport, &media_type) < 0)
4171 goto unknown_transport;
4173 goto unknown_transport;
4175 /* configure the final media type */
4176 GST_DEBUG_OBJECT (src, "setting media type to %s", media_type);
4178 len = stream->ptmap->len;
4179 for (i = 0; i < len; i++) {
4181 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
4183 if (item->caps == NULL)
4186 s = gst_caps_get_structure (item->caps, 0);
4187 gst_structure_set_name (s, media_type);
4188 /* set ssrc if known */
4189 if (transport->ssrc)
4190 gst_structure_set (s, "ssrc", G_TYPE_UINT, transport->ssrc, NULL);
4193 /* try to get and configure a manager, channelpad[0-1] will be configured with
4194 * the pads for the manager, or NULL when no manager is needed. */
4195 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
4198 switch (transport->lower_transport) {
4199 case GST_RTSP_LOWER_TRANS_TCP:
4200 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
4201 goto transport_failed;
4203 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
4204 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
4205 goto transport_failed;
4206 /* fallthrough, the rest is the same for UDP and MCAST */
4207 case GST_RTSP_LOWER_TRANS_UDP:
4208 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
4209 goto transport_failed;
4210 /* configure udpsinks back to the server for RTCP messages and for the
4211 * dummy RTP messages to open NAT. */
4212 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
4213 goto transport_failed;
4216 goto unknown_transport;
4220 GST_DEBUG_OBJECT (src, "creating ghostpad");
4222 gst_pad_use_fixed_caps (outpad);
4224 /* create ghostpad, don't add just yet, this will be done when we activate
4226 name = g_strdup_printf ("stream_%u", stream->id);
4227 template = gst_static_pad_template_get (&rtptemplate);
4228 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
4229 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
4230 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
4231 gst_object_unref (template);
4234 gst_object_unref (outpad);
4236 /* mark pad as ok */
4237 stream->last_ret = GST_FLOW_OK;
4244 GST_DEBUG_OBJECT (src, "failed to configure transport");
4249 GST_DEBUG_OBJECT (src, "unknown transport");
4254 GST_DEBUG_OBJECT (src, "cannot get a session manager");
4259 /* send a couple of dummy random packets on the receiver RTP port to the server,
4260 * this should make a firewall think we initiated the data transfer and
4261 * hopefully allow packets to go from the sender port to our RTP receiver port */
4263 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
4267 if (src->nat_method != GST_RTSP_NAT_DUMMY)
4270 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4271 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4273 if (stream->fakesrc && stream->udpsink[0]) {
4274 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
4275 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
4276 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
4277 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
4278 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
4284 /* Adds the source pads of all configured streams to the element.
4285 * This code is performed when we detected dataflow.
4287 * We detect dataflow from either the _loop function or with pad probes on the
4291 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
4295 GST_DEBUG_OBJECT (src, "activating streams");
4297 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4298 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4300 if (stream->udpsrc[0]) {
4301 /* remove timeout, we are streaming now and timeouts will be handled by
4302 * the session manager and jitter buffer */
4303 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
4305 if (stream->srcpad) {
4306 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
4307 gst_pad_set_active (stream->srcpad, TRUE);
4309 /* if we don't have a session manager, set the caps now. If we have a
4310 * session, we will get a notification of the pad and the caps. */
4311 if (!src->manager) {
4314 caps = stream_get_caps_for_pt (stream, stream->default_pt);
4315 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
4316 gst_pad_set_caps (stream->srcpad, caps);
4319 if (!stream->added) {
4320 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
4321 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
4322 stream->added = TRUE;
4327 /* unblock all pads */
4328 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4329 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4331 if (stream->blockid) {
4332 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
4333 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
4334 stream->blockid = 0;
4342 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
4343 gboolean reset_manager)
4346 guint64 start, stop;
4347 gdouble play_speed, play_scale;
4349 GST_DEBUG_OBJECT (src, "configuring stream caps");
4351 start = segment->position;
4352 stop = segment->duration;
4353 play_speed = segment->rate;
4354 play_scale = segment->applied_rate;
4356 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4357 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4363 len = stream->ptmap->len;
4364 for (j = 0; j < len; j++) {
4366 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, j);
4368 if (item->caps == NULL)
4371 caps = gst_caps_make_writable (item->caps);
4373 if (stream->timebase != -1)
4374 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
4375 (guint) stream->timebase, NULL);
4376 if (stream->seqbase != -1)
4377 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
4378 (guint) stream->seqbase, NULL);
4379 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
4381 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
4382 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
4383 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
4386 GST_DEBUG_OBJECT (src, "stream %p, pt %d, caps %" GST_PTR_FORMAT, stream,
4389 if (item->pt == stream->default_pt && stream->udpsrc[0]) {
4390 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
4394 if (reset_manager && src->manager) {
4395 GST_DEBUG_OBJECT (src, "clear session");
4396 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
4400 static GstFlowReturn
4401 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
4406 /* store the value */
4407 stream->last_ret = ret;
4409 /* if it's success we can return the value right away */
4410 if (ret == GST_FLOW_OK)
4413 /* any other error that is not-linked can be returned right
4415 if (ret != GST_FLOW_NOT_LINKED)
4418 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
4419 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4420 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4422 ret = ostream->last_ret;
4423 /* some other return value (must be SUCCESS but we can return
4424 * other values as well) */
4425 if (ret != GST_FLOW_NOT_LINKED)
4428 /* if we get here, all other pads were unlinked and we return
4429 * NOT_LINKED then */
4435 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
4438 gboolean res = TRUE;
4440 /* only streams that have a connection to the outside world */
4444 if (stream->udpsrc[0]) {
4445 gst_event_ref (event);
4446 res = gst_element_send_event (stream->udpsrc[0], event);
4447 } else if (stream->channelpad[0]) {
4448 gst_event_ref (event);
4449 if (GST_PAD_IS_SRC (stream->channelpad[0]))
4450 res = gst_pad_push_event (stream->channelpad[0], event);
4452 res = gst_pad_send_event (stream->channelpad[0], event);
4455 if (stream->udpsrc[1]) {
4456 gst_event_ref (event);
4457 res &= gst_element_send_event (stream->udpsrc[1], event);
4458 } else if (stream->channelpad[1]) {
4459 gst_event_ref (event);
4460 if (GST_PAD_IS_SRC (stream->channelpad[1]))
4461 res &= gst_pad_push_event (stream->channelpad[1], event);
4463 res &= gst_pad_send_event (stream->channelpad[1], event);
4467 gst_event_unref (event);
4473 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
4476 gboolean res = TRUE;
4478 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4479 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4481 gst_event_ref (event);
4482 res &= gst_rtspsrc_stream_push_event (src, ostream, event);
4484 gst_event_unref (event);
4489 static GstRTSPResult
4490 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
4495 if (info->connection == NULL) {
4496 if (info->url == NULL) {
4497 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
4498 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
4502 /* create connection */
4503 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
4504 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
4505 goto could_not_create;
4507 g_free (info->url_str);
4508 info->url_str = gst_rtsp_url_get_request_uri (info->url);
4510 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
4512 if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
4513 if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
4514 src->tls_validation_flags))
4515 GST_WARNING_OBJECT (src, "Unable to set TLS validation flags");
4517 if (src->tls_database)
4518 gst_rtsp_connection_set_tls_database (info->connection,
4521 if (src->tls_interaction)
4522 gst_rtsp_connection_set_tls_interaction (info->connection,
4523 src->tls_interaction);
4526 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
4527 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
4529 if (src->proxy_host) {
4530 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
4532 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
4537 if (!info->connected) {
4540 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
4541 ("Connecting to %s", info->location));
4542 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
4544 gst_rtsp_connection_connect (info->connection,
4545 src->ptcp_timeout)) < 0)
4546 goto could_not_connect;
4548 info->connected = TRUE;
4555 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
4560 gchar *str = gst_rtsp_strresult (res);
4561 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
4567 gchar *str = gst_rtsp_strresult (res);
4568 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
4574 static GstRTSPResult
4575 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
4578 GST_RTSP_STATE_LOCK (src);
4579 if (info->connected) {
4580 GST_DEBUG_OBJECT (src, "closing connection...");
4581 gst_rtsp_connection_close (info->connection);
4582 info->connected = FALSE;
4584 if (free && info->connection) {
4585 /* free connection */
4586 GST_DEBUG_OBJECT (src, "freeing connection...");
4587 gst_rtsp_connection_free (info->connection);
4588 info->connection = NULL;
4590 GST_RTSP_STATE_UNLOCK (src);
4594 static GstRTSPResult
4595 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
4600 GST_DEBUG_OBJECT (src, "reconnecting connection...");
4601 gst_rtsp_conninfo_close (src, info, FALSE);
4602 res = gst_rtsp_conninfo_connect (src, info, async);
4608 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
4612 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
4613 GST_RTSP_STATE_LOCK (src);
4614 if (src->conninfo.connection && src->conninfo.flushing != flush) {
4615 GST_DEBUG_OBJECT (src, "connection flush");
4616 gst_rtsp_connection_flush (src->conninfo.connection, flush);
4617 src->conninfo.flushing = flush;
4619 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4620 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4621 if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
4622 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
4623 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
4624 stream->conninfo.flushing = flush;
4627 GST_RTSP_STATE_UNLOCK (src);
4630 static GstRTSPResult
4631 gst_rtspsrc_init_request (GstRTSPSrc * src, GstRTSPMessage * msg,
4632 GstRTSPMethod method, const gchar * uri)
4636 res = gst_rtsp_message_init_request (msg, method, uri);
4640 /* set user-agent */
4641 if (src->user_agent)
4642 gst_rtsp_message_add_header (msg, GST_RTSP_HDR_USER_AGENT, src->user_agent);
4647 /* FIXME, handle server request, reply with OK, for now */
4648 static GstRTSPResult
4649 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
4650 GstRTSPMessage * request)
4652 GstRTSPMessage response = { 0 };
4655 GST_DEBUG_OBJECT (src, "got server request message");
4658 gst_rtsp_message_dump (request);
4660 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
4662 if (res == GST_RTSP_ENOTIMPL) {
4663 /* default implementation, send OK */
4664 GST_DEBUG_OBJECT (src, "prepare OK reply");
4666 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
4671 /* let app parse and reply */
4672 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST],
4673 0, request, &response);
4676 gst_rtsp_message_dump (&response);
4678 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
4682 gst_rtsp_message_unset (&response);
4683 } else if (res == GST_RTSP_EEOF)
4691 gst_rtsp_message_unset (&response);
4696 /* send server keep-alive */
4697 static GstRTSPResult
4698 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
4700 GstRTSPMessage request = { 0 };
4702 GstRTSPMethod method;
4703 const gchar *control;
4705 if (src->do_rtsp_keep_alive == FALSE) {
4706 GST_DEBUG_OBJECT (src, "do-rtsp-keep-alive is FALSE, not sending.");
4707 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4711 GST_DEBUG_OBJECT (src, "creating server keep-alive");
4713 /* find a method to use for keep-alive */
4714 if (src->methods & GST_RTSP_GET_PARAMETER)
4715 method = GST_RTSP_GET_PARAMETER;
4717 method = GST_RTSP_OPTIONS;
4719 control = get_aggregate_control (src);
4720 if (control == NULL)
4723 res = gst_rtspsrc_init_request (src, &request, method, control);
4728 gst_rtsp_message_dump (&request);
4731 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
4736 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4737 gst_rtsp_message_unset (&request);
4744 GST_WARNING_OBJECT (src, "no control url to send keepalive");
4749 gchar *str = gst_rtsp_strresult (res);
4751 gst_rtsp_message_unset (&request);
4752 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
4753 ("Could not send keep-alive. (%s)", str));
4759 static GstFlowReturn
4760 gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
4762 GstFlowReturn ret = GST_FLOW_OK;
4764 GstRTSPStream *stream;
4765 GstPad *outpad = NULL;
4771 channel = message->type_data.data.channel;
4773 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
4775 goto unknown_stream;
4777 if (channel == stream->channel[0]) {
4778 outpad = stream->channelpad[0];
4780 } else if (channel == stream->channel[1]) {
4781 outpad = stream->channelpad[1];
4787 /* take a look at the body to figure out what we have */
4788 gst_rtsp_message_get_body (message, &data, &size);
4790 goto invalid_length;
4792 /* channels are not correct on some servers, do extra check */
4793 if (data[1] >= 200 && data[1] <= 204) {
4794 /* hmm RTCP message switch to the RTCP pad of the same stream. */
4795 outpad = stream->channelpad[1];
4799 /* we have no clue what this is, just ignore then. */
4801 goto unknown_stream;
4803 /* take the message body for further processing */
4804 gst_rtsp_message_steal_body (message, &data, &size);
4806 /* strip the trailing \0 */
4809 buf = gst_buffer_new ();
4810 gst_buffer_append_memory (buf,
4811 gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
4813 /* don't need message anymore */
4814 gst_rtsp_message_unset (message);
4816 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
4819 if (src->need_activate) {
4825 guint group_id = gst_util_group_id_next ();
4827 /* generate an SHA256 sum of the URI */
4828 cs = g_checksum_new (G_CHECKSUM_SHA256);
4829 uri = src->conninfo.location;
4830 g_checksum_update (cs, (const guchar *) uri, strlen (uri));
4832 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4833 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4837 g_strdup_printf ("%s/%d", g_checksum_get_string (cs), ostream->id);
4838 event = gst_event_new_stream_start (stream_id);
4839 gst_event_set_group_id (event, group_id);
4842 gst_rtspsrc_stream_push_event (src, ostream, event);
4844 if ((caps = stream_get_caps_for_pt (ostream, ostream->default_pt))) {
4845 /* only streams that have a connection to the outside world */
4846 if (ostream->setup) {
4847 if (ostream->udpsrc[0]) {
4848 gst_element_send_event (ostream->udpsrc[0],
4849 gst_event_new_caps (caps));
4850 } else if (ostream->channelpad[0]) {
4851 if (GST_PAD_IS_SRC (ostream->channelpad[0]))
4852 gst_pad_push_event (ostream->channelpad[0],
4853 gst_event_new_caps (caps));
4855 gst_pad_send_event (ostream->channelpad[0],
4856 gst_event_new_caps (caps));
4859 caps = gst_caps_new_empty_simple ("application/x-rtcp");
4861 if (ostream->udpsrc[1]) {
4862 gst_element_send_event (ostream->udpsrc[1],
4863 gst_event_new_caps (caps));
4864 } else if (ostream->channelpad[1]) {
4865 if (GST_PAD_IS_SRC (ostream->channelpad[1]))
4866 gst_pad_push_event (ostream->channelpad[1],
4867 gst_event_new_caps (caps));
4869 gst_pad_send_event (ostream->channelpad[1],
4870 gst_event_new_caps (caps));
4873 gst_caps_unref (caps);
4877 g_checksum_free (cs);
4879 gst_rtspsrc_activate_streams (src);
4880 src->need_activate = FALSE;
4881 src->need_segment = TRUE;
4884 if (src->base_time == -1) {
4885 /* Take current running_time. This timestamp will be put on
4886 * the first buffer of each stream because we are a live source and so we
4887 * timestamp with the running_time. When we are dealing with TCP, we also
4888 * only timestamp the first buffer (using the DISCONT flag) because a server
4889 * typically bursts data, for which we don't want to compensate by speeding
4890 * up the media. The other timestamps will be interpollated from this one
4891 * using the RTP timestamps. */
4892 GST_OBJECT_LOCK (src);
4893 if (GST_ELEMENT_CLOCK (src)) {
4895 GstClockTime base_time;
4897 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
4898 base_time = GST_ELEMENT_CAST (src)->base_time;
4900 src->base_time = now - base_time;
4902 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
4903 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
4905 GST_OBJECT_UNLOCK (src);
4908 /* If needed send a new segment, don't forget we are live and buffer are
4909 * timestamped with running time */
4910 if (src->need_segment) {
4912 src->need_segment = FALSE;
4913 gst_segment_init (&segment, GST_FORMAT_TIME);
4914 gst_rtspsrc_push_event (src, gst_event_new_segment (&segment));
4917 if (stream->discont && !is_rtcp) {
4918 /* mark first RTP buffer as discont */
4919 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
4920 stream->discont = FALSE;
4921 /* first buffer gets the timestamp, other buffers are not timestamped and
4922 * their presentation time will be interpollated from the rtp timestamps. */
4923 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
4924 GST_TIME_ARGS (src->base_time));
4926 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
4929 /* chain to the peer pad */
4930 if (GST_PAD_IS_SINK (outpad))
4931 ret = gst_pad_chain (outpad, buf);
4933 ret = gst_pad_push (outpad, buf);
4936 /* combine all stream flows for the data transport */
4937 ret = gst_rtspsrc_combine_flows (src, stream, ret);
4944 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
4945 gst_rtsp_message_unset (message);
4950 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4951 ("Short message received, ignoring."));
4952 gst_rtsp_message_unset (message);
4957 static GstFlowReturn
4958 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
4960 GstRTSPMessage message = { 0 };
4962 GstFlowReturn ret = GST_FLOW_OK;
4963 GTimeVal tv_timeout;
4966 /* get the next timeout interval */
4967 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4969 /* see if the timeout period expired */
4970 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
4971 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
4972 /* send keep-alive, only act on interrupt, a warning will be posted for
4974 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4976 /* get new timeout */
4977 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4980 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
4981 tv_timeout.tv_sec, tv_timeout.tv_usec);
4983 /* protect the connection with the connection lock so that we can see when
4984 * we are finished doing server communication */
4986 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4987 &message, src->ptcp_timeout);
4991 GST_DEBUG_OBJECT (src, "we received a server message");
4993 case GST_RTSP_EINTR:
4994 /* we got interrupted this means we need to stop */
4996 case GST_RTSP_ETIMEOUT:
4997 /* no reply, send keep alive */
4998 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4999 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
5003 /* go EOS when the server closed the connection */
5009 switch (message.type) {
5010 case GST_RTSP_MESSAGE_REQUEST:
5011 /* server sends us a request message, handle it */
5013 gst_rtspsrc_handle_request (src, src->conninfo.connection,
5015 if (res == GST_RTSP_EEOF)
5018 goto handle_request_failed;
5020 case GST_RTSP_MESSAGE_RESPONSE:
5021 /* we ignore response messages */
5022 GST_DEBUG_OBJECT (src, "ignoring response message");
5024 gst_rtsp_message_dump (&message);
5026 case GST_RTSP_MESSAGE_DATA:
5027 GST_DEBUG_OBJECT (src, "got data message");
5028 ret = gst_rtspsrc_handle_data (src, &message);
5029 if (ret != GST_FLOW_OK)
5030 goto handle_data_failed;
5033 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
5038 g_assert_not_reached ();
5043 GST_DEBUG_OBJECT (src, "we got an eof from the server");
5044 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5045 ("The server closed the connection."));
5046 src->conninfo.connected = FALSE;
5047 gst_rtsp_message_unset (&message);
5048 return GST_FLOW_EOS;
5052 gst_rtsp_message_unset (&message);
5053 GST_DEBUG_OBJECT (src, "got interrupted");
5054 return GST_FLOW_FLUSHING;
5058 gchar *str = gst_rtsp_strresult (res);
5060 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5061 ("Could not receive message. (%s)", str));
5064 gst_rtsp_message_unset (&message);
5065 return GST_FLOW_ERROR;
5067 handle_request_failed:
5069 gchar *str = gst_rtsp_strresult (res);
5071 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5072 ("Could not handle server message. (%s)", str));
5074 gst_rtsp_message_unset (&message);
5075 return GST_FLOW_ERROR;
5079 GST_DEBUG_OBJECT (src, "could no handle data message");
5084 static GstFlowReturn
5085 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
5088 GstRTSPMessage message = { 0 };
5092 GTimeVal tv_timeout;
5094 /* get the next timeout interval */
5095 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
5097 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
5098 (gint) tv_timeout.tv_sec);
5100 gst_rtsp_message_unset (&message);
5102 /* we should continue reading the TCP socket because the server might
5103 * send us requests. When the session timeout expires, we need to send a
5104 * keep-alive request to keep the session open. */
5105 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
5106 &message, &tv_timeout);
5110 GST_DEBUG_OBJECT (src, "we received a server message");
5112 case GST_RTSP_EINTR:
5113 /* we got interrupted, see what we have to do */
5115 case GST_RTSP_ETIMEOUT:
5116 /* send keep-alive, ignore the result, a warning will be posted. */
5117 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
5118 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
5122 /* server closed the connection. not very fatal for UDP, reconnect and
5123 * see what happens. */
5124 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5125 ("The server closed the connection."));
5126 if (src->udp_reconnect) {
5128 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
5135 GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
5137 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5138 ("Unhandled return value %d.", res));
5142 switch (message.type) {
5143 case GST_RTSP_MESSAGE_REQUEST:
5144 /* server sends us a request message, handle it */
5146 gst_rtspsrc_handle_request (src, src->conninfo.connection,
5148 if (res == GST_RTSP_EEOF)
5151 goto handle_request_failed;
5153 case GST_RTSP_MESSAGE_RESPONSE:
5154 /* we ignore response and data messages */
5155 GST_DEBUG_OBJECT (src, "ignoring response message");
5157 gst_rtsp_message_dump (&message);
5158 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
5159 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
5160 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
5161 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
5162 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
5169 case GST_RTSP_MESSAGE_DATA:
5170 /* we ignore response and data messages */
5171 GST_DEBUG_OBJECT (src, "ignoring data message");
5174 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
5179 g_assert_not_reached ();
5181 /* we get here when the connection got interrupted */
5184 gst_rtsp_message_unset (&message);
5185 GST_DEBUG_OBJECT (src, "got interrupted");
5186 return GST_FLOW_FLUSHING;
5190 gchar *str = gst_rtsp_strresult (res);
5193 src->conninfo.connected = FALSE;
5194 if (res != GST_RTSP_EINTR) {
5195 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
5196 ("Could not connect to server. (%s)", str));
5198 ret = GST_FLOW_ERROR;
5200 ret = GST_FLOW_FLUSHING;
5206 gchar *str = gst_rtsp_strresult (res);
5208 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5209 ("Could not receive message. (%s)", str));
5211 return GST_FLOW_ERROR;
5213 handle_request_failed:
5215 gchar *str = gst_rtsp_strresult (res);
5218 gst_rtsp_message_unset (&message);
5219 if (res != GST_RTSP_EINTR) {
5220 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5221 ("Could not handle server message. (%s)", str));
5223 ret = GST_FLOW_ERROR;
5225 ret = GST_FLOW_FLUSHING;
5231 GST_DEBUG_OBJECT (src, "we got an eof from the server");
5232 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5233 ("The server closed the connection."));
5234 src->conninfo.connected = FALSE;
5235 gst_rtsp_message_unset (&message);
5236 return GST_FLOW_EOS;
5240 static GstRTSPResult
5241 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
5243 GstRTSPResult res = GST_RTSP_OK;
5246 GST_DEBUG_OBJECT (src, "doing reconnect");
5248 GST_OBJECT_LOCK (src);
5249 /* only restart when the pads were not yet activated, else we were
5250 * streaming over UDP */
5251 restart = src->need_activate;
5252 GST_OBJECT_UNLOCK (src);
5254 /* no need to restart, we're done */
5258 /* we can try only TCP now */
5259 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
5261 /* close and cleanup our state */
5262 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
5265 /* see if we have TCP left to try. Also don't try TCP when we were configured
5267 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
5270 /* We post a warning message now to inform the user
5271 * that nothing happened. It's most likely a firewall thing. */
5272 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5273 ("Could not receive any UDP packets for %.4f seconds, maybe your "
5274 "firewall is blocking it. Retrying using a TCP connection.",
5275 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
5277 /* open new connection using tcp */
5278 if (gst_rtspsrc_open (src, async) < 0)
5281 /* start playback */
5282 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
5291 src->cur_protocols = 0;
5292 /* no transport possible, post an error and stop */
5293 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5294 ("Could not receive any UDP packets for %.4f seconds, maybe your "
5295 "firewall is blocking it. No other protocols to try.",
5296 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
5297 return GST_RTSP_ERROR;
5301 GST_DEBUG_OBJECT (src, "open failed");
5306 GST_DEBUG_OBJECT (src, "play failed");
5312 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
5316 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
5319 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
5322 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
5325 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
5333 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
5337 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
5340 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
5343 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
5346 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
5354 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
5358 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
5361 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
5364 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
5367 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
5375 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
5379 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
5382 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
5385 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
5388 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
5396 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
5398 if (ret == GST_RTSP_OK)
5399 gst_rtspsrc_loop_complete_cmd (src, cmd);
5400 else if (ret == GST_RTSP_EINTR)
5401 gst_rtspsrc_loop_cancel_cmd (src, cmd);
5403 gst_rtspsrc_loop_error_cmd (src, cmd);
5407 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
5410 gboolean flushed = FALSE;
5412 /* start new request */
5413 gst_rtspsrc_loop_start_cmd (src, cmd);
5415 GST_DEBUG_OBJECT (src, "sending cmd %s", cmd_to_string (cmd));
5417 GST_OBJECT_LOCK (src);
5418 old = src->pending_cmd;
5419 if (old == CMD_RECONNECT) {
5420 GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
5421 cmd = CMD_RECONNECT;
5423 if (old != CMD_WAIT) {
5424 src->pending_cmd = CMD_WAIT;
5425 GST_OBJECT_UNLOCK (src);
5426 /* cancel previous request */
5427 GST_DEBUG_OBJECT (src, "cancel previous request %s", cmd_to_string (old));
5428 gst_rtspsrc_loop_cancel_cmd (src, old);
5429 GST_OBJECT_LOCK (src);
5431 src->pending_cmd = cmd;
5432 /* interrupt if allowed */
5433 if (src->busy_cmd & mask) {
5434 GST_DEBUG_OBJECT (src, "connection flush busy %s",
5435 cmd_to_string (src->busy_cmd));
5436 gst_rtspsrc_connection_flush (src, TRUE);
5439 GST_DEBUG_OBJECT (src, "not interrupting busy cmd %s",
5440 cmd_to_string (src->busy_cmd));
5443 gst_task_start (src->task);
5444 GST_OBJECT_UNLOCK (src);
5450 gst_rtspsrc_loop (GstRTSPSrc * src)
5454 if (!src->conninfo.connection || !src->conninfo.connected)
5457 if (src->interleaved)
5458 ret = gst_rtspsrc_loop_interleaved (src);
5460 ret = gst_rtspsrc_loop_udp (src);
5462 if (ret != GST_FLOW_OK)
5470 GST_WARNING_OBJECT (src, "we are not connected");
5471 ret = GST_FLOW_FLUSHING;
5476 const gchar *reason = gst_flow_get_name (ret);
5478 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
5479 src->running = FALSE;
5480 if (ret == GST_FLOW_EOS) {
5481 /* perform EOS logic */
5482 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
5483 gst_element_post_message (GST_ELEMENT_CAST (src),
5484 gst_message_new_segment_done (GST_OBJECT_CAST (src),
5485 src->segment.format, src->segment.position));
5486 gst_rtspsrc_push_event (src,
5487 gst_event_new_segment_done (src->segment.format,
5488 src->segment.position));
5490 gst_rtspsrc_push_event (src, gst_event_new_eos ());
5492 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
5493 /* for fatal errors we post an error message, post the error before the
5494 * EOS so the app knows about the error first. */
5495 GST_ELEMENT_ERROR (src, STREAM, FAILED,
5496 ("Internal data flow error."),
5497 ("streaming task paused, reason %s (%d)", reason, ret));
5498 gst_rtspsrc_push_event (src, gst_event_new_eos ());
5500 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_LOOP);
5505 #ifndef GST_DISABLE_GST_DEBUG
5506 static const gchar *
5507 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
5511 while (method != 0) {
5528 static const gchar *
5529 gst_rtspsrc_skip_lws (const gchar * s)
5531 while (g_ascii_isspace (*s))
5536 static const gchar *
5537 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
5539 while (s > start && g_ascii_isspace (*(s - 1)))
5544 static const gchar *
5545 gst_rtspsrc_skip_commas (const gchar * s)
5547 /* The grammar allows for multiple commas */
5548 while (g_ascii_isspace (*s) || *s == ',')
5553 static const gchar *
5554 gst_rtspsrc_skip_item (const gchar * s)
5556 gboolean quoted = FALSE;
5557 const gchar *start = s;
5559 /* A list item ends at the last non-whitespace character
5560 * before a comma which is not inside a quoted-string. Or at
5561 * the end of the string.
5567 if (*s == '\\' && *(s + 1))
5576 return gst_rtspsrc_unskip_lws (s, start);
5580 gst_rtsp_decode_quoted_string (gchar * quoted_string)
5584 src = quoted_string + 1;
5585 dst = quoted_string;
5586 while (*src && *src != '"') {
5587 if (*src == '\\' && *(src + 1))
5594 /* Extract the authentication tokens that the server provided for each method
5595 * into an array of structures and give those to the connection object.
5598 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
5599 const gchar * header, gboolean * stale)
5601 GSList *list = NULL, *iter;
5603 gchar *item, *eq, *name_end, *value;
5605 g_return_if_fail (stale != NULL);
5607 gst_rtsp_connection_clear_auth_params (conn);
5610 /* Parse a header whose content is described by RFC2616 as
5611 * "#something", where "something" does not itself contain commas,
5612 * except as part of quoted-strings, into a list of allocated strings.
5614 header = gst_rtspsrc_skip_commas (header);
5616 end = gst_rtspsrc_skip_item (header);
5617 list = g_slist_prepend (list, g_strndup (header, end - header));
5618 header = gst_rtspsrc_skip_commas (end);
5623 list = g_slist_reverse (list);
5624 for (iter = list; iter; iter = iter->next) {
5627 eq = strchr (item, '=');
5629 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
5630 if (name_end == item) {
5631 /* That's no good... */
5638 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
5640 gst_rtsp_decode_quoted_string (value);
5644 if (value && strcmp (item, "stale") == 0 && strcmp (value, "TRUE") == 0)
5646 gst_rtsp_connection_set_auth_param (conn, item, value);
5650 g_slist_free (list);
5653 /* Parse a WWW-Authenticate Response header and determine the
5654 * available authentication methods
5656 * This code should also cope with the fact that each WWW-Authenticate
5657 * header can contain multiple challenge methods + tokens
5659 * At the moment, for Basic auth, we just do a minimal check and don't
5660 * even parse out the realm */
5662 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
5663 GstRTSPConnection * conn, gboolean * stale)
5667 g_return_if_fail (hdr != NULL);
5668 g_return_if_fail (methods != NULL);
5669 g_return_if_fail (stale != NULL);
5671 /* Skip whitespace at the start of the string */
5672 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
5674 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
5675 *methods |= GST_RTSP_AUTH_BASIC;
5676 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
5677 *methods |= GST_RTSP_AUTH_DIGEST;
5678 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
5683 * gst_rtspsrc_setup_auth:
5684 * @src: the rtsp source
5686 * Configure a username and password and auth method on the
5687 * connection object based on a response we received from the
5690 * Currently, this requires that a username and password were supplied
5691 * in the uri. In the future, they may be requested on demand by sending
5692 * a message up the bus.
5694 * Returns: TRUE if authentication information could be set up correctly.
5697 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
5701 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
5702 GstRTSPAuthMethod method;
5703 GstRTSPResult auth_result;
5705 GstRTSPConnection *conn;
5707 gboolean stale = FALSE;
5709 conn = src->conninfo.connection;
5711 /* Identify the available auth methods and see if any are supported */
5712 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
5713 &hdr, 0) == GST_RTSP_OK) {
5714 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
5717 if (avail_methods == GST_RTSP_AUTH_NONE)
5718 goto no_auth_available;
5720 /* For digest auth, if the response indicates that the session
5721 * data are stale, we just update them in the connection object and
5722 * return TRUE to retry the request */
5724 src->tried_url_auth = FALSE;
5726 url = gst_rtsp_connection_get_url (conn);
5728 /* Do we have username and password available? */
5729 if (url != NULL && !src->tried_url_auth && url->user != NULL
5730 && url->passwd != NULL) {
5733 src->tried_url_auth = TRUE;
5734 GST_DEBUG_OBJECT (src,
5735 "Attempting authentication using credentials from the URL");
5737 user = src->user_id;
5738 pass = src->user_pw;
5739 GST_DEBUG_OBJECT (src,
5740 "Attempting authentication using credentials from the properties");
5743 /* FIXME: If the url didn't contain username and password or we tried them
5744 * already, request a username and passwd from the application via some kind
5745 * of credentials request message */
5747 /* If we don't have a username and passwd at this point, bail out. */
5748 if (user == NULL || pass == NULL)
5751 /* Try to configure for each available authentication method, strongest to
5753 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
5754 /* Check if this method is available on the server */
5755 if ((method & avail_methods) == 0)
5758 /* Pass the credentials to the connection to try on the next request */
5759 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
5760 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
5761 * ignore it and end up retrying later */
5762 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
5763 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
5764 gst_rtsp_auth_method_to_string (method));
5769 if (method == GST_RTSP_AUTH_NONE)
5770 goto no_auth_available;
5776 /* Output an error indicating that we couldn't connect because there were
5777 * no supported authentication protocols */
5778 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5779 ("No supported authentication protocol was found"));
5784 /* We don't fire an error message, we just return FALSE and let the
5785 * normal NOT_AUTHORIZED error be propagated */
5790 static GstRTSPResult
5791 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5792 GstRTSPMessage * request, GstRTSPMessage * response,
5793 GstRTSPStatusCode * code)
5796 GstRTSPStatusCode thecode;
5797 gchar *content_base = NULL;
5801 if (!src->short_header)
5802 gst_rtsp_ext_list_before_send (src->extensions, request);
5804 GST_DEBUG_OBJECT (src, "sending message");
5807 gst_rtsp_message_dump (request);
5809 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
5813 gst_rtsp_connection_reset_timeout (conn);
5816 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
5821 gst_rtsp_message_dump (response);
5823 switch (response->type) {
5824 case GST_RTSP_MESSAGE_REQUEST:
5825 res = gst_rtspsrc_handle_request (src, conn, response);
5826 if (res == GST_RTSP_EEOF)
5829 goto handle_request_failed;
5831 case GST_RTSP_MESSAGE_RESPONSE:
5832 /* ok, a response is good */
5833 GST_DEBUG_OBJECT (src, "received response message");
5835 case GST_RTSP_MESSAGE_DATA:
5836 /* get next response */
5837 GST_DEBUG_OBJECT (src, "handle data response message");
5838 gst_rtspsrc_handle_data (src, response);
5841 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
5846 thecode = response->type_data.response.code;
5848 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
5850 /* if the caller wanted the result code, we store it. */
5854 /* If the request didn't succeed, bail out before doing any more */
5855 if (thecode != GST_RTSP_STS_OK)
5858 /* store new content base if any */
5859 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
5862 g_free (src->content_base);
5863 src->content_base = g_strdup (content_base);
5865 gst_rtsp_ext_list_after_send (src->extensions, request, response);
5872 gchar *str = gst_rtsp_strresult (res);
5874 if (res != GST_RTSP_EINTR) {
5875 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5876 ("Could not send message. (%s)", str));
5878 GST_WARNING_OBJECT (src, "send interrupted");
5887 GST_WARNING_OBJECT (src, "server closed connection");
5888 if ((try == 0) && !src->interleaved && src->udp_reconnect) {
5890 /* if reconnect succeeds, try again */
5892 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
5896 /* only try once after reconnect, then fallthrough and error out */
5899 gchar *str = gst_rtsp_strresult (res);
5901 if (res != GST_RTSP_EINTR) {
5902 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5903 ("Could not receive message. (%s)", str));
5905 GST_WARNING_OBJECT (src, "receive interrupted");
5913 handle_request_failed:
5915 /* ERROR was posted */
5916 gst_rtsp_message_unset (response);
5921 GST_DEBUG_OBJECT (src, "we got an eof from the server");
5922 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5923 ("The server closed the connection."));
5924 gst_rtsp_message_unset (response);
5931 * @src: the rtsp source
5932 * @conn: the connection to send on
5933 * @request: must point to a valid request
5934 * @response: must point to an empty #GstRTSPMessage
5935 * @code: an optional code result
5937 * send @request and retrieve the response in @response. optionally @code can be
5938 * non-NULL in which case it will contain the status code of the response.
5940 * If This function returns #GST_RTSP_OK, @response will contain a valid response
5941 * message that should be cleaned with gst_rtsp_message_unset() after usage.
5943 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
5944 * @response message) if the response code was not 200 (OK).
5946 * If the attempt results in an authentication failure, then this will attempt
5947 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
5950 * Returns: #GST_RTSP_OK if the processing was successful.
5952 static GstRTSPResult
5953 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5954 GstRTSPMessage * request, GstRTSPMessage * response,
5955 GstRTSPStatusCode * code)
5957 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
5958 GstRTSPResult res = GST_RTSP_ERROR;
5961 GstRTSPMethod method = GST_RTSP_INVALID;
5967 /* make sure we don't loop forever */
5971 /* save method so we can disable it when the server complains */
5972 method = request->type_data.request.method;
5975 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
5979 case GST_RTSP_STS_UNAUTHORIZED:
5980 case GST_RTSP_STS_NOT_FOUND:
5981 if (gst_rtspsrc_setup_auth (src, response)) {
5982 /* Try the request/response again after configuring the auth info
5990 } while (retry == TRUE);
5992 /* If the user requested the code, let them handle errors, otherwise
5993 * post an error below */
5996 else if (int_code != GST_RTSP_STS_OK)
5997 goto error_response;
6004 GST_DEBUG_OBJECT (src, "got error %d", res);
6009 res = GST_RTSP_ERROR;
6011 switch (response->type_data.response.code) {
6012 case GST_RTSP_STS_NOT_FOUND:
6013 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
6014 response->type_data.response.reason));
6016 case GST_RTSP_STS_UNAUTHORIZED:
6017 GST_ELEMENT_ERROR (src, RESOURCE, NOT_AUTHORIZED, (NULL), ("%s",
6018 response->type_data.response.reason));
6020 case GST_RTSP_STS_MOVED_PERMANENTLY:
6021 case GST_RTSP_STS_MOVE_TEMPORARILY:
6023 gchar *new_location;
6024 GstRTSPLowerTrans transports;
6026 GST_DEBUG_OBJECT (src, "got redirection");
6027 /* if we don't have a Location Header, we must error */
6028 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
6029 &new_location, 0) < 0)
6032 /* When we receive a redirect result, we go back to the INIT state after
6033 * parsing the new URI. The caller should do the needed steps to issue
6034 * a new setup when it detects this state change. */
6035 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
6037 /* save current transports */
6038 if (src->conninfo.url)
6039 transports = src->conninfo.url->transports;
6041 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
6043 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
6045 /* set old transports */
6046 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
6047 src->conninfo.url->transports = transports;
6049 src->need_redirect = TRUE;
6050 src->state = GST_RTSP_STATE_INIT;
6054 case GST_RTSP_STS_NOT_ACCEPTABLE:
6055 case GST_RTSP_STS_NOT_IMPLEMENTED:
6056 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
6057 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
6058 gst_rtsp_method_as_text (method));
6059 src->methods &= ~method;
6063 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
6064 ("Got error response: %d (%s).", response->type_data.response.code,
6065 response->type_data.response.reason));
6068 /* if we return ERROR we should unset the response ourselves */
6069 if (res == GST_RTSP_ERROR)
6070 gst_rtsp_message_unset (response);
6076 static GstRTSPResult
6077 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
6078 GstRTSPMessage * response, GstRTSPSrc * src)
6080 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
6085 /* parse the response and collect all the supported methods. We need this
6086 * information so that we don't try to send an unsupported request to the
6090 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
6092 GstRTSPHeaderField field;
6096 /* reset supported methods */
6099 /* Try Allow Header first */
6100 field = GST_RTSP_HDR_ALLOW;
6103 gst_rtsp_message_get_header (response, field, &respoptions, indx);
6104 if (indx == 0 && !respoptions) {
6105 /* if no Allow header was found then try the Public header... */
6106 field = GST_RTSP_HDR_PUBLIC;
6107 gst_rtsp_message_get_header (response, field, &respoptions, indx);
6112 src->methods |= gst_rtsp_options_from_text (respoptions);
6117 if (src->methods == 0) {
6118 /* neither Allow nor Public are required, assume the server supports
6119 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
6121 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
6122 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
6124 /* always assume PLAY, FIXME, extensions should be able to override
6126 src->methods |= GST_RTSP_PLAY;
6127 /* also assume it will support Range */
6128 src->seekable = TRUE;
6130 /* we need describe and setup */
6131 if (!(src->methods & GST_RTSP_DESCRIBE))
6133 if (!(src->methods & GST_RTSP_SETUP))
6141 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
6142 ("Server does not support DESCRIBE."));
6147 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
6148 ("Server does not support SETUP."));
6153 /* masks to be kept in sync with the hardcoded protocol order of preference
6155 static const guint protocol_masks[] = {
6156 GST_RTSP_LOWER_TRANS_UDP,
6157 GST_RTSP_LOWER_TRANS_UDP_MCAST,
6158 GST_RTSP_LOWER_TRANS_TCP,
6162 static GstRTSPResult
6163 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
6164 GstRTSPLowerTrans protocols, GstRTSPProfile profile, gchar ** transports)
6168 gboolean add_udp_str;
6173 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
6178 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
6180 /* extension listed transports, use those */
6181 if (*transports != NULL)
6184 /* it's the default */
6185 add_udp_str = FALSE;
6187 /* the default RTSP transports */
6188 result = g_string_new ("RTP");
6191 case GST_RTSP_PROFILE_AVP:
6192 g_string_append (result, "/AVP");
6194 case GST_RTSP_PROFILE_SAVP:
6195 g_string_append (result, "/SAVP");
6197 case GST_RTSP_PROFILE_AVPF:
6198 g_string_append (result, "/AVPF");
6200 case GST_RTSP_PROFILE_SAVPF:
6201 g_string_append (result, "/SAVPF");
6207 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
6208 GST_DEBUG_OBJECT (src, "adding UDP unicast");
6210 g_string_append (result, "/UDP");
6211 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
6212 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
6213 GST_DEBUG_OBJECT (src, "adding UDP multicast");
6214 /* we don't have to allocate any UDP ports yet, if the selected transport
6215 * turns out to be multicast we can create them and join the multicast
6216 * group indicated in the transport reply */
6218 g_string_append (result, "/UDP");
6219 g_string_append (result, ";multicast");
6220 if (src->next_port_num != 0) {
6221 if (src->client_port_range.max > 0 &&
6222 src->next_port_num >= src->client_port_range.max)
6225 g_string_append_printf (result, ";client_port=%d-%d",
6226 src->next_port_num, src->next_port_num + 1);
6228 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
6229 GST_DEBUG_OBJECT (src, "adding TCP");
6231 g_string_append (result, "/TCP;unicast;interleaved=%%i1-%%i2");
6233 *transports = g_string_free (result, FALSE);
6235 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
6242 GST_ERROR ("extension gave error %d", res);
6247 GST_ERROR ("no more ports available");
6248 return GST_RTSP_ERROR;
6252 static GstRTSPResult
6253 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
6254 gint orig_rtpport, gint orig_rtcpport)
6257 gint nr_udp, nr_int;
6259 gint rtpport = 0, rtcpport = 0;
6262 src = stream->parent;
6264 /* find number of placeholders first */
6265 if (strstr (*transports, "%%i2"))
6267 else if (strstr (*transports, "%%i1"))
6272 if (strstr (*transports, "%%u2"))
6274 else if (strstr (*transports, "%%u1"))
6279 if (nr_udp == 0 && nr_int == 0)
6283 if (!orig_rtpport || !orig_rtcpport) {
6284 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
6287 rtpport = orig_rtpport;
6288 rtcpport = orig_rtcpport;
6292 str = g_string_new ("");
6294 while ((next = strstr (p, "%%"))) {
6295 g_string_append_len (str, p, next - p);
6296 if (next[2] == 'u') {
6298 g_string_append_printf (str, "%d", rtpport);
6299 else if (next[3] == '2')
6300 g_string_append_printf (str, "%d", rtcpport);
6302 if (next[2] == 'i') {
6304 g_string_append_printf (str, "%d", src->free_channel);
6305 else if (next[3] == '2')
6306 g_string_append_printf (str, "%d", src->free_channel + 1);
6311 /* append final part */
6312 g_string_append (str, p);
6314 g_free (*transports);
6315 *transports = g_string_free (str, FALSE);
6323 GST_ERROR ("failed to allocate udp ports");
6324 return GST_RTSP_ERROR;
6329 enc_key_length_from_cipher_name (const gchar * cipher)
6331 if (g_strcmp0 (cipher, "aes-128-icm") == 0)
6332 return AES_128_KEY_LEN;
6333 else if (g_strcmp0 (cipher, "aes-256-icm") == 0)
6334 return AES_256_KEY_LEN;
6336 GST_ERROR ("encryption algorithm '%s' not supported", cipher);
6342 auth_key_length_from_auth_name (const gchar * auth)
6344 if (g_strcmp0 (auth, "hmac-sha1-32") == 0)
6345 return HMAC_32_KEY_LEN;
6346 else if (g_strcmp0 (auth, "hmac-sha1-80") == 0)
6347 return HMAC_80_KEY_LEN;
6349 GST_ERROR ("authentication algorithm '%s' not supported", auth);
6355 signal_get_srtcp_params (GstRTSPSrc * src, GstRTSPStream * stream)
6357 GstCaps *caps = NULL;
6359 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_REQUEST_RTCP_KEY], 0,
6363 GST_DEBUG_OBJECT (src, "SRTP parameters received");
6369 default_srtcp_params (void)
6376 guint data_size = GST_ROUND_UP_4 (KEY_SIZE);
6378 /* create a random key */
6379 key_data = g_malloc (data_size);
6380 for (i = 0; i < data_size; i += 4)
6381 GST_WRITE_UINT32_BE (key_data + i, g_random_int ());
6383 buf = gst_buffer_new_wrapped (key_data, KEY_SIZE);
6385 caps = gst_caps_new_simple ("application/x-srtp",
6386 "srtp-key", GST_TYPE_BUFFER, buf,
6387 "srtcp-cipher", G_TYPE_STRING, "aes-128-icm",
6388 "srtcp-auth", G_TYPE_STRING, "hmac-sha1-80", NULL);
6390 gst_buffer_unref (buf);
6396 gst_rtspsrc_stream_make_keymgmt (GstRTSPSrc * src, GstRTSPStream * stream)
6399 gchar *result, *base64;
6402 GstMIKEYMessage *msg;
6403 GstMIKEYPayload *payload, *pkd;
6409 const gchar *srtcpcipher, *srtcpauth;
6411 stream->srtcpparams = signal_get_srtcp_params (src, stream);
6412 if (stream->srtcpparams == NULL)
6413 stream->srtcpparams = default_srtcp_params ();
6415 s = gst_caps_get_structure (stream->srtcpparams, 0);
6417 srtcpcipher = gst_structure_get_string (s, "srtcp-cipher");
6418 srtcpauth = gst_structure_get_string (s, "srtcp-auth");
6419 val = gst_structure_get_value (s, "srtp-key");
6421 if (srtcpcipher == NULL || srtcpauth == NULL || val == NULL) {
6422 GST_ERROR_OBJECT (src, "could not find the right SRTP parameters in caps");
6426 srtpkey = gst_value_get_buffer (val);
6428 msg = gst_mikey_message_new ();
6429 /* unencrypted MIKEY message, we send this over TLS so this is allowed */
6430 gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT,
6431 FALSE, GST_MIKEY_PRF_MIKEY_1, g_random_int (), GST_MIKEY_MAP_TYPE_SRTP);
6432 /* add policy '0' for our SSRC */
6433 gst_mikey_message_add_cs_srtp (msg, 0, stream->send_ssrc, 0);
6434 /* timestamp is now */
6435 gst_mikey_message_add_t_now_ntp_utc (msg);
6436 /* add some random data */
6437 gst_mikey_message_add_rand_len (msg, 16);
6439 /* the policy '0' is SRTP */
6440 payload = gst_mikey_payload_new (GST_MIKEY_PT_SP);
6441 gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP);
6443 /* only AES-CM is supported */
6445 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1, &byte);
6446 /* encryption key length */
6447 byte = enc_key_length_from_cipher_name (srtcpcipher);
6448 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_KEY_LEN, 1,
6450 /* only HMAC-SHA1 */
6451 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1,
6453 /* authentication key length */
6454 byte = auth_key_length_from_auth_name (srtcpauth);
6455 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_KEY_LEN, 1,
6457 /* we enable encryption on RTP and RTCP */
6458 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1,
6460 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1,
6462 /* we enable authentication on RTP and RTCP */
6463 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1,
6465 gst_mikey_message_add_payload (msg, payload);
6467 /* make unencrypted KEMAC */
6468 payload = gst_mikey_payload_new (GST_MIKEY_PT_KEMAC);
6469 gst_mikey_payload_kemac_set (payload, GST_MIKEY_ENC_NULL, GST_MIKEY_MAC_NULL);
6470 /* add the key in KEMAC */
6471 pkd = gst_mikey_payload_new (GST_MIKEY_PT_KEY_DATA);
6472 gst_buffer_map (srtpkey, &info, GST_MAP_READ);
6473 gst_mikey_payload_key_data_set_key (pkd, GST_MIKEY_KD_TEK, info.size,
6475 gst_buffer_unmap (srtpkey, &info);
6476 gst_mikey_payload_kemac_add_sub (payload, pkd);
6477 gst_mikey_message_add_payload (msg, payload);
6479 /* now serialize this to bytes */
6480 bytes = gst_mikey_message_to_bytes (msg, NULL, NULL);
6481 gst_mikey_message_unref (msg);
6482 /* and make it into base64 */
6483 data = g_bytes_get_data (bytes, &size);
6484 base64 = g_base64_encode (data, size);
6485 g_bytes_unref (bytes);
6487 result = g_strdup_printf ("prot=mikey;uri=\"%s\";data=\"%s\"",
6488 stream->conninfo.location, base64);
6495 /* Perform the SETUP request for all the streams.
6497 * We ask the server for a specific transport, which initially includes all the
6498 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
6499 * two local UDP ports that we send to the server.
6501 * Once the server replied with a transport, we configure the other streams
6502 * with the same transport.
6504 * This function will also configure the stream for the selected transport,
6505 * which basically means creating the pipeline.
6507 static GstRTSPResult
6508 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
6511 GstRTSPResult res = GST_RTSP_ERROR;
6512 GstRTSPMessage request = { 0 };
6513 GstRTSPMessage response = { 0 };
6514 GstRTSPStream *stream = NULL;
6515 GstRTSPLowerTrans protocols;
6516 GstRTSPStatusCode code;
6517 gboolean unsupported_real = FALSE;
6518 gint rtpport, rtcpport;
6522 if (src->conninfo.connection) {
6523 url = gst_rtsp_connection_get_url (src->conninfo.connection);
6524 /* we initially allow all configured lower transports. based on the URL
6525 * transports and the replies from the server we narrow them down. */
6526 protocols = url->transports & src->cur_protocols;
6529 protocols = src->cur_protocols;
6535 /* reset some state */
6536 src->free_channel = 0;
6537 src->interleaved = FALSE;
6538 src->need_activate = FALSE;
6539 /* keep track of next port number, 0 is random */
6540 src->next_port_num = src->client_port_range.min;
6541 rtpport = rtcpport = 0;
6543 if (G_UNLIKELY (src->streams == NULL))
6546 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6547 GstRTSPConnection *conn;
6554 stream = (GstRTSPStream *) walk->data;
6556 caps = stream_get_caps_for_pt (stream, stream->default_pt);
6558 GST_DEBUG_OBJECT (src, "skipping stream %p, no caps", stream);
6562 if (stream->skipped) {
6563 GST_DEBUG_OBJECT (src, "skipping stream %p", stream);
6567 /* see if we need to configure this stream */
6568 if (!gst_rtsp_ext_list_configure_stream (src->extensions, caps)) {
6569 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
6574 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_SELECT_STREAM], 0,
6575 stream->id, caps, &selected);
6577 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by signal", stream);
6581 /* merge/overwrite global caps */
6586 s = gst_caps_get_structure (caps, 0);
6588 num = gst_structure_n_fields (src->props);
6589 for (j = 0; j < num; j++) {
6593 name = gst_structure_nth_field_name (src->props, j);
6594 val = gst_structure_get_value (src->props, name);
6595 gst_structure_set_value (s, name, val);
6597 GST_DEBUG_OBJECT (src, "copied %s", name);
6601 /* skip setup if we have no URL for it */
6602 if (stream->conninfo.location == NULL) {
6603 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
6607 if (src->conninfo.connection == NULL) {
6608 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
6609 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
6612 conn = stream->conninfo.connection;
6614 conn = src->conninfo.connection;
6616 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
6617 stream->conninfo.location);
6619 /* if we have a multicast connection, only suggest multicast from now on */
6620 if (stream->is_multicast)
6621 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
6624 /* first selectable protocol */
6625 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
6627 if (!protocol_masks[mask])
6631 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
6632 protocol_masks[mask]);
6633 /* create a string with first transport in line */
6635 res = gst_rtspsrc_create_transports_string (src,
6636 protocols & protocol_masks[mask], stream->profile, &transports);
6637 if (res < 0 || transports == NULL)
6638 goto setup_transport_failed;
6640 if (strlen (transports) == 0) {
6641 g_free (transports);
6642 GST_DEBUG_OBJECT (src, "no transports found");
6647 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
6649 /* replace placeholders with real values, this function will optionally
6650 * allocate UDP ports and other info needed to execute the setup request */
6651 res = gst_rtspsrc_prepare_transports (stream, &transports,
6652 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
6654 g_free (transports);
6655 goto setup_transport_failed;
6658 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
6660 /* create SETUP request */
6662 gst_rtspsrc_init_request (src, &request, GST_RTSP_SETUP,
6663 stream->conninfo.location);
6665 g_free (transports);
6666 goto create_request_failed;
6669 /* select transport */
6670 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
6673 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
6674 stream->profile == GST_RTSP_PROFILE_SAVPF) {
6675 hval = gst_rtspsrc_stream_make_keymgmt (src, stream);
6676 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_KEYMGMT, hval);
6679 /* if the user wants a non default RTP packet size we add the blocksize
6681 if (src->rtp_blocksize > 0) {
6682 hval = g_strdup_printf ("%d", src->rtp_blocksize);
6683 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
6687 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
6690 /* handle the code ourselves */
6691 res = gst_rtspsrc_send (src, conn, &request, &response, &code);
6696 case GST_RTSP_STS_OK:
6698 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
6699 gst_rtsp_message_unset (&request);
6700 gst_rtsp_message_unset (&response);
6701 /* cleanup of leftover transport */
6702 gst_rtspsrc_stream_free_udp (stream);
6703 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
6704 * we might be in this case */
6705 if (stream->container && rtpport && rtcpport && !retry) {
6706 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
6711 /* this transport did not go down well, but we may have others to try
6712 * that we did not send yet, try those and only give up then
6713 * but not without checking for lost cause/extension so we can
6714 * post a nicer/more useful error message later */
6715 if (!unsupported_real)
6716 unsupported_real = stream->is_real;
6717 /* select next available protocol, give up on this stream if none */
6719 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
6721 if (!protocol_masks[mask] || unsupported_real)
6726 /* cleanup of leftover transport and move to the next stream */
6727 gst_rtspsrc_stream_free_udp (stream);
6728 goto response_error;
6731 /* parse response transport */
6733 gchar *resptrans = NULL;
6734 GstRTSPTransport transport = { 0 };
6736 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
6739 gst_rtspsrc_stream_free_udp (stream);
6743 /* parse transport, go to next stream on parse error */
6744 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
6745 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
6749 /* update allowed transports for other streams. once the transport of
6750 * one stream has been determined, we make sure that all other streams
6751 * are configured in the same way */
6752 switch (transport.lower_transport) {
6753 case GST_RTSP_LOWER_TRANS_TCP:
6754 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
6755 protocols = GST_RTSP_LOWER_TRANS_TCP;
6756 src->interleaved = TRUE;
6757 /* update free channels */
6759 MAX (transport.interleaved.min, src->free_channel);
6761 MAX (transport.interleaved.max, src->free_channel);
6762 src->free_channel++;
6764 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
6765 /* only allow multicast for other streams */
6766 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
6767 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
6768 /* if the server selected our ports, increment our counters so that
6769 * we select a new port later */
6770 if (src->next_port_num == transport.port.min &&
6771 src->next_port_num + 1 == transport.port.max) {
6772 src->next_port_num += 2;
6775 case GST_RTSP_LOWER_TRANS_UDP:
6776 /* only allow unicast for other streams */
6777 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
6778 protocols = GST_RTSP_LOWER_TRANS_UDP;
6781 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
6782 transport.lower_transport);
6786 if (!src->interleaved || !retry) {
6787 /* now configure the stream with the selected transport */
6788 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
6789 GST_DEBUG_OBJECT (src,
6790 "could not configure stream %p transport, skipping stream",
6793 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
6794 /* retain the first allocated UDP port pair */
6795 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
6796 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
6799 /* we need to activate at least one streams when we detect activity */
6800 src->need_activate = TRUE;
6802 /* stream is setup now */
6803 stream->setup = TRUE;
6808 GstRTSPStream *sskip;
6810 skip = g_list_next (skip);
6814 sskip = (GstRTSPStream *) skip->data;
6816 /* skip all streams with the same control url */
6817 if (g_str_equal (stream->conninfo.location, sskip->conninfo.location)) {
6818 GST_DEBUG_OBJECT (src, "found stream %p with same control %s",
6819 sskip, sskip->conninfo.location);
6820 sskip->skipped = TRUE;
6825 /* clean up our transport struct */
6826 gst_rtsp_transport_init (&transport);
6827 /* clean up used RTSP messages */
6828 gst_rtsp_message_unset (&request);
6829 gst_rtsp_message_unset (&response);
6833 /* store the transport protocol that was configured */
6834 src->cur_protocols = protocols;
6836 gst_rtsp_ext_list_stream_select (src->extensions, url);
6838 /* if there is nothing to activate, error out */
6839 if (!src->need_activate)
6840 goto nothing_to_activate;
6847 /* no transport possible, post an error and stop */
6848 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
6849 ("Could not connect to server, no protocols left"));
6850 return GST_RTSP_ERROR;
6854 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6855 ("SDP contains no streams"));
6856 return GST_RTSP_ERROR;
6858 create_request_failed:
6860 gchar *str = gst_rtsp_strresult (res);
6862 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6863 ("Could not create request. (%s)", str));
6867 setup_transport_failed:
6869 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6870 ("Could not setup transport."));
6871 res = GST_RTSP_ERROR;
6876 const gchar *str = gst_rtsp_status_as_text (code);
6878 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6879 ("Error (%d): %s", code, GST_STR_NULL (str)));
6880 res = GST_RTSP_ERROR;
6885 gchar *str = gst_rtsp_strresult (res);
6887 if (res != GST_RTSP_EINTR) {
6888 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6889 ("Could not send message. (%s)", str));
6891 GST_WARNING_OBJECT (src, "send interrupted");
6898 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6899 ("Server did not select transport."));
6900 res = GST_RTSP_ERROR;
6903 nothing_to_activate:
6905 /* none of the available error codes is really right .. */
6906 if (unsupported_real) {
6907 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6908 (_("No supported stream was found. You might need to install a "
6909 "GStreamer RTSP extension plugin for Real media streams.")),
6912 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6913 (_("No supported stream was found. You might need to allow "
6914 "more transport protocols or may otherwise be missing "
6915 "the right GStreamer RTSP extension plugin.")), (NULL));
6917 return GST_RTSP_ERROR;
6921 gst_rtsp_message_unset (&request);
6922 gst_rtsp_message_unset (&response);
6928 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
6929 GstSegment * segment)
6932 GstRTSPTimeRange *therange;
6935 gst_rtsp_range_free (src->range);
6937 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
6938 GST_DEBUG_OBJECT (src, "parsed range %s", range);
6939 src->range = therange;
6941 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
6943 gst_segment_init (segment, GST_FORMAT_TIME);
6947 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
6948 therange->min.type, therange->min.seconds, therange->max.type,
6949 therange->max.seconds);
6951 if (therange->min.type == GST_RTSP_TIME_NOW)
6953 else if (therange->min.type == GST_RTSP_TIME_END)
6956 seconds = therange->min.seconds * GST_SECOND;
6958 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
6959 GST_TIME_ARGS (seconds));
6961 /* we need to start playback without clipping from the position reported by
6963 segment->start = seconds;
6964 segment->position = seconds;
6966 if (therange->max.type == GST_RTSP_TIME_NOW)
6968 else if (therange->max.type == GST_RTSP_TIME_END)
6971 seconds = therange->max.seconds * GST_SECOND;
6973 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
6974 GST_TIME_ARGS (seconds));
6976 /* live (WMS) server might send overflowed large max as its idea of infinity,
6977 * compensate to prevent problems later on */
6978 if (seconds != -1 && seconds < 0) {
6980 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
6983 /* live (WMS) might send min == max, which is not worth recording */
6984 if (segment->duration == -1 && seconds == segment->start)
6987 /* don't change duration with unknown value, we might have a valid value
6988 * there that we want to keep. */
6990 segment->duration = seconds;
6995 /* Parse clock profived by the server with following syntax:
6997 * "GstNetTimeProvider <wrapped-clock> <server-IP:port> <clock-time>"
7000 gst_rtspsrc_parse_gst_clock (GstRTSPSrc * src, const gchar * gstclock)
7002 gboolean res = FALSE;
7004 if (g_str_has_prefix (gstclock, "GstNetTimeProvider ")) {
7005 gchar **fields = NULL, **parts = NULL;
7006 gchar *remote_ip, *str;
7008 GstClockTime base_time;
7011 fields = g_strsplit (gstclock, " ", 0);
7013 /* wrapped clock, not very interesting for now */
7014 if (fields[1] == NULL)
7017 /* remote IP address and port */
7018 if ((str = fields[2]) == NULL)
7021 parts = g_strsplit (str, ":", 0);
7023 if ((remote_ip = parts[0]) == NULL)
7026 if ((str = parts[1]) == NULL)
7034 if ((str = fields[3]) == NULL)
7037 base_time = g_ascii_strtoull (str, NULL, 10);
7040 gst_net_client_clock_new ((gchar *) "GstRTSPClock", remote_ip, port,
7043 if (src->provided_clock)
7044 gst_object_unref (src->provided_clock);
7045 src->provided_clock = netclock;
7047 gst_element_post_message (GST_ELEMENT_CAST (src),
7048 gst_message_new_clock_provide (GST_OBJECT_CAST (src),
7049 src->provided_clock, TRUE));
7053 g_strfreev (fields);
7059 /* must be called with the RTSP state lock */
7060 static GstRTSPResult
7061 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
7067 /* prepare global stream caps properties */
7069 gst_structure_remove_all_fields (src->props);
7071 src->props = gst_structure_new_empty ("RTSPProperties");
7074 gst_sdp_message_dump (sdp);
7076 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
7078 /* let the app inspect and change the SDP */
7079 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_ON_SDP], 0, sdp);
7081 gst_segment_init (&src->segment, GST_FORMAT_TIME);
7083 /* parse range for duration reporting. */
7088 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
7092 /* keep track of the range and configure it in the segment */
7093 if (gst_rtspsrc_parse_range (src, range, &src->segment))
7097 /* parse clock information. This is GStreamer specific, a server can tell the
7098 * client what clock it is using and wrap that in a network clock. The
7099 * advantage of that is that we can slave to it. */
7101 const gchar *gstclock;
7104 gstclock = gst_sdp_message_get_attribute_val_n (sdp, "x-gst-clock", i);
7105 if (gstclock == NULL)
7108 /* parse the clock and expose it in the provide_clock method */
7109 if (gst_rtspsrc_parse_gst_clock (src, gstclock))
7113 /* try to find a global control attribute. Note that a '*' means that we should
7114 * do aggregate control with the current url (so we don't do anything and
7115 * leave the current connection as is) */
7117 const gchar *control;
7120 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
7121 if (control == NULL)
7124 /* only take fully qualified urls */
7125 if (g_str_has_prefix (control, "rtsp://"))
7129 g_free (src->conninfo.location);
7130 src->conninfo.location = g_strdup (control);
7131 /* make a connection for this, if there was a connection already, nothing
7133 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
7134 GST_ERROR_OBJECT (src, "could not connect");
7137 /* we need to keep the control url separate from the connection url because
7138 * the rules for constructing the media control url need it */
7139 g_free (src->control);
7140 src->control = g_strdup (control);
7143 /* create streams */
7144 n_streams = gst_sdp_message_medias_len (sdp);
7145 for (i = 0; i < n_streams; i++) {
7146 gst_rtspsrc_create_stream (src, sdp, i);
7149 src->state = GST_RTSP_STATE_INIT;
7152 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
7155 /* reset our state */
7156 src->need_range = TRUE;
7159 src->state = GST_RTSP_STATE_READY;
7166 GST_ERROR_OBJECT (src, "setup failed");
7167 gst_rtspsrc_cleanup (src);
7172 static GstRTSPResult
7173 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
7177 GstRTSPMessage request = { 0 };
7178 GstRTSPMessage response = { 0 };
7181 gchar *respcont = NULL;
7184 src->need_redirect = FALSE;
7186 /* can't continue without a valid url */
7187 if (G_UNLIKELY (src->conninfo.url == NULL)) {
7188 res = GST_RTSP_EINVAL;
7191 src->tried_url_auth = FALSE;
7193 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
7194 goto connect_failed;
7196 /* create OPTIONS */
7197 GST_DEBUG_OBJECT (src, "create options...");
7199 gst_rtspsrc_init_request (src, &request, GST_RTSP_OPTIONS,
7200 src->conninfo.url_str);
7202 goto create_request_failed;
7205 GST_DEBUG_OBJECT (src, "send options...");
7208 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
7211 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
7216 if (!gst_rtspsrc_parse_methods (src, &response))
7219 /* create DESCRIBE */
7220 GST_DEBUG_OBJECT (src, "create describe...");
7222 gst_rtspsrc_init_request (src, &request, GST_RTSP_DESCRIBE,
7223 src->conninfo.url_str);
7225 goto create_request_failed;
7227 /* we only accept SDP for now */
7228 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
7232 GST_DEBUG_OBJECT (src, "send describe...");
7235 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
7238 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
7242 /* we only perform redirect for the describe, currently */
7243 if (src->need_redirect) {
7244 /* close connection, we don't have to send a TEARDOWN yet, ignore the
7246 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
7248 gst_rtsp_message_unset (&request);
7249 gst_rtsp_message_unset (&response);
7255 /* it could be that the DESCRIBE method was not implemented */
7256 if (!(src->methods & GST_RTSP_DESCRIBE))
7259 /* check if reply is SDP */
7260 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
7262 /* could not be set but since the request returned OK, we assume it
7263 * was SDP, else check it. */
7265 if (g_ascii_strcasecmp (respcont, "application/sdp") != 0)
7266 goto wrong_content_type;
7269 /* get message body and parse as SDP */
7270 gst_rtsp_message_get_body (&response, &data, &size);
7271 if (data == NULL || size == 0)
7274 GST_DEBUG_OBJECT (src, "parse SDP...");
7275 gst_sdp_message_new (sdp);
7276 gst_sdp_message_parse_buffer (data, size, *sdp);
7278 /* clean up any messages */
7279 gst_rtsp_message_unset (&request);
7280 gst_rtsp_message_unset (&response);
7287 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
7288 ("No valid RTSP URL was provided"));
7293 gchar *str = gst_rtsp_strresult (res);
7295 if (res != GST_RTSP_EINTR) {
7296 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
7297 ("Failed to connect. (%s)", str));
7299 GST_WARNING_OBJECT (src, "connect interrupted");
7304 create_request_failed:
7306 gchar *str = gst_rtsp_strresult (res);
7308 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7309 ("Could not create request. (%s)", str));
7315 /* Don't post a message - the rtsp_send method will have
7316 * taken care of it because we passed NULL for the response code */
7321 /* error was posted */
7322 res = GST_RTSP_ERROR;
7327 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
7328 ("Server does not support SDP, got %s.", respcont));
7329 res = GST_RTSP_ERROR;
7334 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
7335 ("Server can not provide an SDP."));
7336 res = GST_RTSP_ERROR;
7341 if (src->conninfo.connection) {
7342 GST_DEBUG_OBJECT (src, "free connection");
7343 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
7345 gst_rtsp_message_unset (&request);
7346 gst_rtsp_message_unset (&response);
7351 static GstRTSPResult
7352 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
7357 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
7359 if (src->sdp == NULL) {
7360 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
7364 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
7369 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
7376 GST_WARNING_OBJECT (src, "can't get sdp");
7377 src->open_error = TRUE;
7382 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
7383 src->open_error = TRUE;
7388 static GstRTSPResult
7389 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
7391 GstRTSPMessage request = { 0 };
7392 GstRTSPMessage response = { 0 };
7393 GstRTSPResult res = GST_RTSP_OK;
7395 const gchar *control;
7397 GST_DEBUG_OBJECT (src, "TEARDOWN...");
7399 gst_rtspsrc_set_state (src, GST_STATE_READY);
7401 if (src->state < GST_RTSP_STATE_READY) {
7402 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
7409 /* construct a control url */
7410 control = get_aggregate_control (src);
7412 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
7415 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7416 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7417 const gchar *setup_url;
7418 GstRTSPConnInfo *info;
7420 /* try aggregate control first but do non-aggregate control otherwise */
7422 setup_url = control;
7423 else if ((setup_url = stream->conninfo.location) == NULL)
7426 if (src->conninfo.connection) {
7427 info = &src->conninfo;
7428 } else if (stream->conninfo.connection) {
7429 info = &stream->conninfo;
7433 if (!info->connected)
7438 gst_rtspsrc_init_request (src, &request, GST_RTSP_TEARDOWN, setup_url);
7440 goto create_request_failed;
7443 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
7446 gst_rtspsrc_send (src, info->connection, &request, &response,
7450 /* FIXME, parse result? */
7451 gst_rtsp_message_unset (&request);
7452 gst_rtsp_message_unset (&response);
7455 /* early exit when we did aggregate control */
7461 /* close connections */
7462 GST_DEBUG_OBJECT (src, "closing connection...");
7463 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
7464 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7465 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7466 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
7470 gst_rtspsrc_cleanup (src);
7472 src->state = GST_RTSP_STATE_INVALID;
7475 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
7480 create_request_failed:
7482 gchar *str = gst_rtsp_strresult (res);
7484 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7485 ("Could not create request. (%s)", str));
7491 gchar *str = gst_rtsp_strresult (res);
7493 gst_rtsp_message_unset (&request);
7494 if (res != GST_RTSP_EINTR) {
7495 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7496 ("Could not send message. (%s)", str));
7498 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
7505 GST_DEBUG_OBJECT (src,
7506 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
7511 /* RTP-Info is of the format:
7513 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
7515 * rtptime corresponds to the timestamp for the NPT time given in the header
7516 * seqbase corresponds to the next sequence number we received. This number
7517 * indicates the first seqnum after the seek and should be used to discard
7518 * packets that are from before the seek.
7521 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
7526 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
7528 infos = g_strsplit (rtpinfo, ",", 0);
7529 for (i = 0; infos[i]; i++) {
7531 GstRTSPStream *stream;
7535 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
7537 /* init values, types of seqbase and timebase are bigger than needed so we
7538 * can store -1 as uninitialized values */
7543 /* parse url, find stream for url.
7544 * parse seq and rtptime. The seq number should be configured in the rtp
7545 * depayloader or session manager to detect gaps. Same for the rtptime, it
7546 * should be used to create an initial time newsegment. */
7547 fields = g_strsplit (infos[i], ";", 0);
7548 for (j = 0; fields[j]; j++) {
7549 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
7550 /* remove leading whitespace */
7551 fields[j] = g_strchug (fields[j]);
7552 if (g_str_has_prefix (fields[j], "url=")) {
7553 /* get the url and the stream */
7555 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
7556 } else if (g_str_has_prefix (fields[j], "seq=")) {
7557 seqbase = atoi (fields[j] + 4);
7558 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
7559 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
7562 g_strfreev (fields);
7563 /* now we need to store the values for the caps of the stream */
7564 if (stream != NULL) {
7565 GST_DEBUG_OBJECT (src,
7566 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
7567 stream, seqbase, timebase);
7569 /* we have a stream, configure detected params */
7570 stream->seqbase = seqbase;
7571 stream->timebase = timebase;
7580 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
7585 interval = strtoul (rtcp, NULL, 10);
7586 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
7591 interval *= GST_MSECOND;
7593 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7594 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7596 /* already (optionally) retrieved this when configuring manager */
7597 if (stream->session) {
7598 GObject *rtpsession = stream->session;
7600 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
7602 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
7606 /* now it happens that (Xenon) server sending this may also provide bogus
7607 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
7608 * and just use RTP-Info to sync */
7610 GObjectClass *klass;
7612 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
7613 if (g_object_class_find_property (klass, "rtcp-sync")) {
7614 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
7615 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
7621 gst_rtspsrc_get_float (const gchar * dstr)
7623 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
7625 /* canonicalise floating point string so we can handle float strings
7626 * in the form "24.930" or "24,930" irrespective of the current locale */
7627 g_strlcpy (s, dstr, sizeof (s));
7628 g_strdelimit (s, ",", '.');
7629 return g_ascii_strtod (s, NULL);
7633 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
7635 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
7637 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
7638 g_strlcpy (val_str, "now", sizeof (val_str));
7640 if (segment->position == 0) {
7641 g_strlcpy (val_str, "0", sizeof (val_str));
7643 g_ascii_dtostr (val_str, sizeof (val_str),
7644 ((gdouble) segment->position) / GST_SECOND);
7647 return g_strdup_printf ("npt=%s-", val_str);
7651 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
7655 stream->timebase = -1;
7656 stream->seqbase = -1;
7658 len = stream->ptmap->len;
7659 for (i = 0; i < len; i++) {
7660 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
7663 if (item->caps == NULL)
7666 item->caps = gst_caps_make_writable (item->caps);
7667 s = gst_caps_get_structure (item->caps, 0);
7668 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
7672 static GstRTSPResult
7673 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
7675 GstRTSPResult res = GST_RTSP_OK;
7677 if (src->state < GST_RTSP_STATE_READY) {
7678 res = GST_RTSP_ERROR;
7679 if (src->open_error) {
7680 GST_DEBUG_OBJECT (src, "the stream was in error");
7684 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
7686 if ((res = gst_rtspsrc_open (src, async)) < 0) {
7687 GST_DEBUG_OBJECT (src, "failed to open stream");
7696 static GstRTSPResult
7697 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
7699 GstRTSPMessage request = { 0 };
7700 GstRTSPMessage response = { 0 };
7701 GstRTSPResult res = GST_RTSP_OK;
7705 const gchar *control;
7707 GST_DEBUG_OBJECT (src, "PLAY...");
7709 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
7712 if (!(src->methods & GST_RTSP_PLAY))
7715 if (src->state == GST_RTSP_STATE_PLAYING)
7718 if (!src->conninfo.connection || !src->conninfo.connected)
7721 /* send some dummy packets before we activate the receive in the
7723 gst_rtspsrc_send_dummy_packets (src);
7725 /* require new SR packets */
7727 g_signal_emit_by_name (src->manager, "reset-sync", NULL);
7729 /* construct a control url */
7730 control = get_aggregate_control (src);
7732 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7733 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7734 const gchar *setup_url;
7735 GstRTSPConnection *conn;
7737 /* try aggregate control first but do non-aggregate control otherwise */
7739 setup_url = control;
7740 else if ((setup_url = stream->conninfo.location) == NULL)
7743 if (src->conninfo.connection) {
7744 conn = src->conninfo.connection;
7745 } else if (stream->conninfo.connection) {
7746 conn = stream->conninfo.connection;
7752 res = gst_rtspsrc_init_request (src, &request, GST_RTSP_PLAY, setup_url);
7754 goto create_request_failed;
7756 if (src->need_range) {
7757 hval = gen_range_header (src, segment);
7759 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
7761 /* store the newsegment event so it can be sent from the streaming thread. */
7762 src->need_segment = TRUE;
7765 if (segment->rate != 1.0) {
7766 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
7768 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
7770 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
7772 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
7776 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
7778 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7781 /* seek may have silently failed as it is not supported */
7782 if (!(src->methods & GST_RTSP_PLAY)) {
7783 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
7784 /* obviously it is supported as we made it here */
7785 src->methods |= GST_RTSP_PLAY;
7786 src->seekable = FALSE;
7787 /* but there is nothing to parse in the response,
7788 * so convey we have no idea and not to expect anything particular */
7789 clear_rtp_base (src, stream);
7793 /* need to do for all streams */
7794 for (run = src->streams; run; run = g_list_next (run))
7795 clear_rtp_base (src, (GstRTSPStream *) run->data);
7797 /* NOTE the above also disables npt based eos detection */
7798 /* and below forces position to 0,
7799 * which is visible feedback we lost the plot */
7800 segment->start = segment->position = src->last_pos;
7803 gst_rtsp_message_unset (&request);
7805 /* parse RTP npt field. This is the current position in the stream (Normal
7806 * Play Time) and should be put in the NEWSEGMENT position field. */
7807 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
7809 gst_rtspsrc_parse_range (src, hval, segment);
7811 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
7812 segment->rate = 1.0;
7814 /* parse Speed header. This is the intended playback rate of the stream
7815 * and should be put in the NEWSEGMENT rate field. */
7816 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
7817 0) == GST_RTSP_OK) {
7818 segment->rate = gst_rtspsrc_get_float (hval);
7819 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
7820 &hval, 0) == GST_RTSP_OK) {
7821 segment->rate = gst_rtspsrc_get_float (hval);
7824 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
7825 * for the RTP packets. If this is not present, we assume all starts from 0...
7826 * This is info for the RTP session manager that we pass to it in caps. */
7828 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
7829 &hval, hval_idx++) == GST_RTSP_OK)
7830 gst_rtspsrc_parse_rtpinfo (src, hval);
7832 /* some servers indicate RTCP parameters in PLAY response,
7833 * rather than properly in SDP */
7834 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
7835 &hval, 0) == GST_RTSP_OK)
7836 gst_rtspsrc_handle_rtcp_interval (src, hval);
7838 gst_rtsp_message_unset (&response);
7840 /* early exit when we did aggregate control */
7844 /* configure the caps of the streams after we parsed all headers. Only reset
7845 * the manager object when we set a new Range header (we did a seek) */
7846 gst_rtspsrc_configure_caps (src, segment, src->need_range);
7848 /* set to PLAYING after we have configured the caps, otherwise we
7849 * might end up calling request_key (with SRTP) while caps are still
7850 * being configured. */
7851 gst_rtspsrc_set_state (src, GST_STATE_PLAYING);
7853 /* set again when needed */
7854 src->need_range = FALSE;
7856 src->running = TRUE;
7857 src->base_time = -1;
7858 src->state = GST_RTSP_STATE_PLAYING;
7861 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
7862 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7863 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7864 stream->discont = TRUE;
7869 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
7876 GST_DEBUG_OBJECT (src, "failed to open stream");
7881 GST_DEBUG_OBJECT (src, "PLAY is not supported");
7886 GST_DEBUG_OBJECT (src, "we were already PLAYING");
7889 create_request_failed:
7891 gchar *str = gst_rtsp_strresult (res);
7893 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7894 ("Could not create request. (%s)", str));
7900 gchar *str = gst_rtsp_strresult (res);
7902 gst_rtsp_message_unset (&request);
7903 if (res != GST_RTSP_EINTR) {
7904 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7905 ("Could not send message. (%s)", str));
7907 GST_WARNING_OBJECT (src, "PLAY interrupted");
7914 static GstRTSPResult
7915 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
7917 GstRTSPResult res = GST_RTSP_OK;
7918 GstRTSPMessage request = { 0 };
7919 GstRTSPMessage response = { 0 };
7921 const gchar *control;
7923 GST_DEBUG_OBJECT (src, "PAUSE...");
7925 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
7928 if (!(src->methods & GST_RTSP_PAUSE))
7931 if (src->state == GST_RTSP_STATE_READY)
7934 if (!src->conninfo.connection || !src->conninfo.connected)
7937 /* construct a control url */
7938 control = get_aggregate_control (src);
7940 /* loop over the streams. We might exit the loop early when we could do an
7941 * aggregate control */
7942 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7943 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7944 GstRTSPConnection *conn;
7945 const gchar *setup_url;
7947 /* try aggregate control first but do non-aggregate control otherwise */
7949 setup_url = control;
7950 else if ((setup_url = stream->conninfo.location) == NULL)
7953 if (src->conninfo.connection) {
7954 conn = src->conninfo.connection;
7955 } else if (stream->conninfo.connection) {
7956 conn = stream->conninfo.connection;
7962 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
7963 ("Sending PAUSE request"));
7966 gst_rtspsrc_init_request (src, &request, GST_RTSP_PAUSE,
7968 goto create_request_failed;
7970 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7973 gst_rtsp_message_unset (&request);
7974 gst_rtsp_message_unset (&response);
7976 /* exit early when we did agregate control */
7981 /* change element states now */
7982 gst_rtspsrc_set_state (src, GST_STATE_PAUSED);
7985 src->state = GST_RTSP_STATE_READY;
7989 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
7996 GST_DEBUG_OBJECT (src, "failed to open stream");
8001 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
8006 GST_DEBUG_OBJECT (src, "we were already PAUSED");
8009 create_request_failed:
8011 gchar *str = gst_rtsp_strresult (res);
8013 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
8014 ("Could not create request. (%s)", str));
8020 gchar *str = gst_rtsp_strresult (res);
8022 gst_rtsp_message_unset (&request);
8023 if (res != GST_RTSP_EINTR) {
8024 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
8025 ("Could not send message. (%s)", str));
8027 GST_WARNING_OBJECT (src, "PAUSE interrupted");
8035 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
8037 GstRTSPSrc *rtspsrc;
8039 rtspsrc = GST_RTSPSRC (bin);
8041 switch (GST_MESSAGE_TYPE (message)) {
8042 case GST_MESSAGE_EOS:
8043 gst_message_unref (message);
8045 case GST_MESSAGE_ELEMENT:
8047 const GstStructure *s = gst_message_get_structure (message);
8049 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
8050 gboolean ignore_timeout;
8052 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
8054 GST_OBJECT_LOCK (rtspsrc);
8055 ignore_timeout = rtspsrc->ignore_timeout;
8056 rtspsrc->ignore_timeout = TRUE;
8057 GST_OBJECT_UNLOCK (rtspsrc);
8059 /* we only act on the first udp timeout message, others are irrelevant
8060 * and can be ignored. */
8061 if (!ignore_timeout)
8062 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
8064 gst_message_unref (message);
8067 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
8070 case GST_MESSAGE_ERROR:
8073 GstRTSPStream *stream;
8076 udpsrc = GST_MESSAGE_SRC (message);
8078 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
8079 GST_ELEMENT_NAME (udpsrc));
8081 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
8085 /* we ignore the RTCP udpsrc */
8086 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
8089 /* if we get error messages from the udp sources, that's not a problem as
8090 * long as not all of them error out. We also don't really know what the
8091 * problem is, the message does not give enough detail... */
8092 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
8093 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
8094 if (ret != GST_FLOW_OK)
8098 gst_message_unref (message);
8102 /* fatal but not our message, forward */
8103 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
8108 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
8114 /* the thread where everything happens */
8116 gst_rtspsrc_thread (GstRTSPSrc * src)
8120 GST_OBJECT_LOCK (src);
8121 cmd = src->pending_cmd;
8122 if (cmd == CMD_RECONNECT || cmd == CMD_PLAY || cmd == CMD_PAUSE
8123 || cmd == CMD_LOOP || cmd == CMD_OPEN)
8124 src->pending_cmd = CMD_LOOP;
8126 src->pending_cmd = CMD_WAIT;
8127 GST_DEBUG_OBJECT (src, "got command %s", cmd_to_string (cmd));
8129 /* we got the message command, so ensure communication is possible again */
8130 gst_rtspsrc_connection_flush (src, FALSE);
8132 src->busy_cmd = cmd;
8133 GST_OBJECT_UNLOCK (src);
8137 gst_rtspsrc_open (src, TRUE);
8140 gst_rtspsrc_play (src, &src->segment, TRUE);
8143 gst_rtspsrc_pause (src, TRUE);
8146 gst_rtspsrc_close (src, TRUE, FALSE);
8149 gst_rtspsrc_loop (src);
8152 gst_rtspsrc_reconnect (src, FALSE);
8158 GST_OBJECT_LOCK (src);
8159 /* and go back to sleep */
8160 if (src->pending_cmd == CMD_WAIT) {
8162 gst_task_pause (src->task);
8165 src->busy_cmd = CMD_WAIT;
8166 GST_OBJECT_UNLOCK (src);
8170 gst_rtspsrc_start (GstRTSPSrc * src)
8172 GST_DEBUG_OBJECT (src, "starting");
8174 GST_OBJECT_LOCK (src);
8176 src->pending_cmd = CMD_WAIT;
8178 if (src->task == NULL) {
8179 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src, NULL);
8180 if (src->task == NULL)
8183 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
8185 GST_OBJECT_UNLOCK (src);
8192 GST_OBJECT_UNLOCK (src);
8193 GST_ERROR_OBJECT (src, "failed to create task");
8199 gst_rtspsrc_stop (GstRTSPSrc * src)
8203 GST_DEBUG_OBJECT (src, "stopping");
8205 /* also cancels pending task */
8206 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_ALL);
8208 GST_OBJECT_LOCK (src);
8209 if ((task = src->task)) {
8211 GST_OBJECT_UNLOCK (src);
8213 gst_task_stop (task);
8215 /* make sure it is not running */
8216 GST_RTSP_STREAM_LOCK (src);
8217 GST_RTSP_STREAM_UNLOCK (src);
8219 /* now wait for the task to finish */
8220 gst_task_join (task);
8222 /* and free the task */
8223 gst_object_unref (GST_OBJECT (task));
8225 GST_OBJECT_LOCK (src);
8227 GST_OBJECT_UNLOCK (src);
8229 /* ensure synchronously all is closed and clean */
8230 gst_rtspsrc_close (src, FALSE, TRUE);
8235 static GstStateChangeReturn
8236 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
8238 GstRTSPSrc *rtspsrc;
8239 GstStateChangeReturn ret;
8241 rtspsrc = GST_RTSPSRC (element);
8243 switch (transition) {
8244 case GST_STATE_CHANGE_NULL_TO_READY:
8245 if (!gst_rtspsrc_start (rtspsrc))
8248 case GST_STATE_CHANGE_READY_TO_PAUSED:
8249 /* init some state */
8250 rtspsrc->cur_protocols = rtspsrc->protocols;
8251 /* first attempt, don't ignore timeouts */
8252 rtspsrc->ignore_timeout = FALSE;
8253 rtspsrc->open_error = FALSE;
8254 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
8256 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
8257 set_manager_buffer_mode (rtspsrc);
8259 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
8260 /* unblock the tcp tasks and make the loop waiting */
8261 if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
8262 /* make sure it is waiting before we send PAUSE or PLAY below */
8263 GST_RTSP_STREAM_LOCK (rtspsrc);
8264 GST_RTSP_STREAM_UNLOCK (rtspsrc);
8267 case GST_STATE_CHANGE_PAUSED_TO_READY:
8273 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
8274 if (ret == GST_STATE_CHANGE_FAILURE)
8277 switch (transition) {
8278 case GST_STATE_CHANGE_NULL_TO_READY:
8279 ret = GST_STATE_CHANGE_SUCCESS;
8281 case GST_STATE_CHANGE_READY_TO_PAUSED:
8282 ret = GST_STATE_CHANGE_NO_PREROLL;
8284 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
8285 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
8286 ret = GST_STATE_CHANGE_SUCCESS;
8288 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
8289 /* send pause request and keep the idle task around */
8290 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
8291 ret = GST_STATE_CHANGE_NO_PREROLL;
8293 case GST_STATE_CHANGE_PAUSED_TO_READY:
8294 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
8295 ret = GST_STATE_CHANGE_SUCCESS;
8297 case GST_STATE_CHANGE_READY_TO_NULL:
8298 gst_rtspsrc_stop (rtspsrc);
8299 ret = GST_STATE_CHANGE_SUCCESS;
8310 GST_DEBUG_OBJECT (rtspsrc, "start failed");
8311 return GST_STATE_CHANGE_FAILURE;
8316 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
8319 GstRTSPSrc *rtspsrc;
8321 rtspsrc = GST_RTSPSRC (element);
8323 if (GST_EVENT_IS_DOWNSTREAM (event)) {
8324 res = gst_rtspsrc_push_event (rtspsrc, event);
8326 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
8333 /*** GSTURIHANDLER INTERFACE *************************************************/
8336 gst_rtspsrc_uri_get_type (GType type)
8341 static const gchar *const *
8342 gst_rtspsrc_uri_get_protocols (GType type)
8344 static const gchar *protocols[] =
8345 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
8346 "rtsps", "rtspsu", "rtspst", "rtspsh", NULL
8353 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
8355 GstRTSPSrc *src = GST_RTSPSRC (handler);
8357 /* FIXME: make thread-safe */
8358 return g_strdup (src->conninfo.location);
8362 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
8368 GstRTSPUrl *newurl = NULL;
8369 GstSDPMessage *sdp = NULL;
8371 src = GST_RTSPSRC (handler);
8373 /* same URI, we're fine */
8374 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
8377 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
8378 sres = gst_sdp_message_new (&sdp);
8382 GST_DEBUG_OBJECT (src, "parsing SDP message");
8383 sres = gst_sdp_message_parse_uri (uri, sdp);
8388 GST_DEBUG_OBJECT (src, "parsing URI");
8389 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
8393 /* if worked, free previous and store new url object along with the original
8395 GST_DEBUG_OBJECT (src, "configuring URI");
8396 g_free (src->conninfo.location);
8397 src->conninfo.location = g_strdup (uri);
8398 gst_rtsp_url_free (src->conninfo.url);
8399 src->conninfo.url = newurl;
8400 g_free (src->conninfo.url_str);
8402 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
8404 src->conninfo.url_str = NULL;
8407 gst_sdp_message_free (src->sdp);
8409 src->from_sdp = sdp != NULL;
8411 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
8412 GST_DEBUG_OBJECT (src, "request uri is: %s",
8413 GST_STR_NULL (src->conninfo.url_str));
8420 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
8425 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", sres);
8426 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
8427 "Could not create SDP");
8432 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", sres,
8433 GST_STR_NULL (uri));
8434 gst_sdp_message_free (sdp);
8435 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
8441 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
8442 GST_STR_NULL (uri), res);
8443 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
8444 "Invalid RTSP URI");
8450 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
8452 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
8454 iface->get_type = gst_rtspsrc_uri_get_type;
8455 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
8456 iface->get_uri = gst_rtspsrc_uri_get_uri;
8457 iface->set_uri = gst_rtspsrc_uri_set_uri;