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/gstrtppayloads.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 #define AES_128_KEY_LEN 16
168 #define AES_256_KEY_LEN 32
170 #define HMAC_32_KEY_LEN 4
171 #define HMAC_80_KEY_LEN 10
173 #define DEFAULT_LOCATION NULL
174 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
175 #define DEFAULT_DEBUG FALSE
176 #define DEFAULT_RETRY 20
177 #define DEFAULT_TIMEOUT 5000000
178 #define DEFAULT_UDP_BUFFER_SIZE 0x80000
179 #define DEFAULT_TCP_TIMEOUT 20000000
180 #define DEFAULT_LATENCY_MS 2000
181 #define DEFAULT_DROP_ON_LATENCY FALSE
182 #define DEFAULT_CONNECTION_SPEED 0
183 #define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
184 #define DEFAULT_DO_RTCP TRUE
185 #define DEFAULT_DO_RTSP_KEEP_ALIVE TRUE
186 #define DEFAULT_PROXY NULL
187 #define DEFAULT_RTP_BLOCKSIZE 0
188 #define DEFAULT_USER_ID NULL
189 #define DEFAULT_USER_PW NULL
190 #define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO
191 #define DEFAULT_PORT_RANGE NULL
192 #define DEFAULT_SHORT_HEADER FALSE
193 #define DEFAULT_PROBATION 2
194 #define DEFAULT_UDP_RECONNECT TRUE
195 #define DEFAULT_MULTICAST_IFACE NULL
196 #define DEFAULT_NTP_SYNC FALSE
197 #define DEFAULT_USE_PIPELINE_CLOCK FALSE
198 #define DEFAULT_TLS_VALIDATION_FLAGS G_TLS_CERTIFICATE_VALIDATE_ALL
199 #define DEFAULT_TLS_DATABASE NULL
200 #define DEFAULT_DO_RETRANSMISSION TRUE
212 PROP_DROP_ON_LATENCY,
213 PROP_CONNECTION_SPEED,
216 PROP_DO_RTSP_KEEP_ALIVE,
225 PROP_UDP_BUFFER_SIZE,
229 PROP_MULTICAST_IFACE,
231 PROP_USE_PIPELINE_CLOCK,
233 PROP_TLS_VALIDATION_FLAGS,
235 PROP_DO_RETRANSMISSION,
239 #define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
241 gst_rtsp_nat_method_get_type (void)
243 static GType rtsp_nat_method_type = 0;
244 static const GEnumValue rtsp_nat_method[] = {
245 {GST_RTSP_NAT_NONE, "None", "none"},
246 {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
250 if (!rtsp_nat_method_type) {
251 rtsp_nat_method_type =
252 g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
254 return rtsp_nat_method_type;
257 static void gst_rtspsrc_finalize (GObject * object);
259 static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
260 const GValue * value, GParamSpec * pspec);
261 static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
262 GValue * value, GParamSpec * pspec);
264 static GstClock *gst_rtspsrc_provide_clock (GstElement * element);
266 static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
267 gpointer iface_data);
269 static void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
272 static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
273 static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
275 static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
277 static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
278 GstStateChange transition);
279 static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
280 static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
282 static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
283 GstRTSPMessage * response);
285 static gboolean gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd,
287 static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
288 GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
290 static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
291 static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
293 static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
294 static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
295 gboolean only_close);
297 static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
298 const gchar * uri, GError ** error);
299 static gchar *gst_rtspsrc_uri_get_uri (GstURIHandler * handler);
301 static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
302 static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
303 static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
304 GstRTSPStream * stream, GstEvent * event);
305 static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
306 static void gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush);
314 /* commands we send to out loop to notify it of events */
315 #define CMD_OPEN (1 << 0)
316 #define CMD_PLAY (1 << 1)
317 #define CMD_PAUSE (1 << 2)
318 #define CMD_CLOSE (1 << 3)
319 #define CMD_WAIT (1 << 4)
320 #define CMD_RECONNECT (1 << 5)
321 #define CMD_LOOP (1 << 6)
323 /* mask for all commands */
324 #define CMD_ALL ((CMD_LOOP << 1) - 1)
326 #define GST_ELEMENT_PROGRESS(el, type, code, text) \
328 gchar *__txt = _gst_element_error_printf text; \
329 gst_element_post_message (GST_ELEMENT_CAST (el), \
330 gst_message_new_progress (GST_OBJECT_CAST (el), \
331 GST_PROGRESS_TYPE_ ##type, code, __txt)); \
335 static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 };
337 #define gst_rtspsrc_parent_class parent_class
338 G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
339 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
342 default_select_stream (GstRTSPSrc * src, guint id, GstCaps * caps)
344 GST_DEBUG_OBJECT (src, "default handler");
349 select_stream_accum (GSignalInvocationHint * ihint,
350 GValue * return_accu, const GValue * handler_return, gpointer data)
354 myboolean = g_value_get_boolean (handler_return);
355 GST_DEBUG ("accum %d", myboolean);
356 g_value_set_boolean (return_accu, myboolean);
358 /* stop emission if FALSE */
363 gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
365 GObjectClass *gobject_class;
366 GstElementClass *gstelement_class;
367 GstBinClass *gstbin_class;
369 gobject_class = (GObjectClass *) klass;
370 gstelement_class = (GstElementClass *) klass;
371 gstbin_class = (GstBinClass *) klass;
373 GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
375 gobject_class->set_property = gst_rtspsrc_set_property;
376 gobject_class->get_property = gst_rtspsrc_get_property;
378 gobject_class->finalize = gst_rtspsrc_finalize;
380 g_object_class_install_property (gobject_class, PROP_LOCATION,
381 g_param_spec_string ("location", "RTSP Location",
382 "Location of the RTSP url to read",
383 DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
385 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
386 g_param_spec_flags ("protocols", "Protocols",
387 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
388 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
390 g_object_class_install_property (gobject_class, PROP_DEBUG,
391 g_param_spec_boolean ("debug", "Debug",
392 "Dump request and response messages to stdout",
393 DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
395 g_object_class_install_property (gobject_class, PROP_RETRY,
396 g_param_spec_uint ("retry", "Retry",
397 "Max number of retries when allocating RTP ports.",
398 0, G_MAXUINT16, DEFAULT_RETRY,
399 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
401 g_object_class_install_property (gobject_class, PROP_TIMEOUT,
402 g_param_spec_uint64 ("timeout", "Timeout",
403 "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
404 0, G_MAXUINT64, DEFAULT_TIMEOUT,
405 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
407 g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
408 g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
409 "Fail after timeout microseconds on TCP connections (0 = disabled)",
410 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
411 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
413 g_object_class_install_property (gobject_class, PROP_LATENCY,
414 g_param_spec_uint ("latency", "Buffer latency in ms",
415 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
416 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
418 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
419 g_param_spec_boolean ("drop-on-latency",
420 "Drop buffers when maximum latency is reached",
421 "Tells the jitterbuffer to never exceed the given latency in size",
422 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
424 g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
425 g_param_spec_uint64 ("connection-speed", "Connection Speed",
426 "Network connection speed in kbps (0 = unknown)",
427 0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
428 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
430 g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
431 g_param_spec_enum ("nat-method", "NAT Method",
432 "Method to use for traversing firewalls and NAT",
433 GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
434 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
437 * GstRTSPSrc:do-rtcp:
439 * Enable RTCP support. Some old server don't like RTCP and then this property
440 * needs to be set to FALSE.
442 g_object_class_install_property (gobject_class, PROP_DO_RTCP,
443 g_param_spec_boolean ("do-rtcp", "Do RTCP",
444 "Send RTCP packets, disable for old incompatible server.",
445 DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
448 * GstRTSPSrc:do-rtsp-keep-alive:
450 * Enable RTSP keep alive support. Some old server don't like RTSP
451 * keep alive and then this property needs to be set to FALSE.
453 g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
454 g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
455 "Send RTSP keep alive packets, disable for old incompatible server.",
456 DEFAULT_DO_RTSP_KEEP_ALIVE,
457 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
462 * Set the proxy parameters. This has to be a string of the format
463 * [http://][user:passwd@]host[:port].
465 g_object_class_install_property (gobject_class, PROP_PROXY,
466 g_param_spec_string ("proxy", "Proxy",
467 "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
468 DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
470 * GstRTSPSrc:proxy-id:
472 * Sets the proxy URI user id for authentication. If the URI set via the
473 * "proxy" property contains a user-id already, that will take precedence.
477 g_object_class_install_property (gobject_class, PROP_PROXY_ID,
478 g_param_spec_string ("proxy-id", "proxy-id",
479 "HTTP proxy URI user id for authentication", "",
480 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
482 * GstRTSPSrc:proxy-pw:
484 * Sets the proxy URI password for authentication. If the URI set via the
485 * "proxy" property contains a password already, that will take precedence.
489 g_object_class_install_property (gobject_class, PROP_PROXY_PW,
490 g_param_spec_string ("proxy-pw", "proxy-pw",
491 "HTTP proxy URI user password for authentication", "",
492 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
495 * GstRTSPSrc:rtp-blocksize:
497 * RTP package size to suggest to server.
499 g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
500 g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
501 "RTP package size to suggest to server (0 = disabled)",
502 0, 65536, DEFAULT_RTP_BLOCKSIZE,
503 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
505 g_object_class_install_property (gobject_class,
507 g_param_spec_string ("user-id", "user-id",
508 "RTSP location URI user id for authentication", DEFAULT_USER_ID,
509 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
510 g_object_class_install_property (gobject_class, PROP_USER_PW,
511 g_param_spec_string ("user-pw", "user-pw",
512 "RTSP location URI user password for authentication", DEFAULT_USER_PW,
513 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
516 * GstRTSPSrc:buffer-mode:
518 * Control the buffering and timestamping mode used by the jitterbuffer.
520 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
521 g_param_spec_enum ("buffer-mode", "Buffer Mode",
522 "Control the buffering algorithm in use",
523 GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
524 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
527 * GstRTSPSrc:port-range:
529 * Configure the client port numbers that can be used to recieve RTP and
532 g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
533 g_param_spec_string ("port-range", "Port range",
534 "Client port range that can be used to receive RTP and RTCP data, "
535 "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
536 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
539 * GstRTSPSrc:udp-buffer-size:
541 * Size of the kernel UDP receive buffer in bytes.
543 g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
544 g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
545 "Size of the kernel UDP receive buffer in bytes, 0=default",
546 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
547 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
550 * GstRTSPSrc:short-header:
552 * Only send the basic RTSP headers for broken encoders.
554 g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
555 g_param_spec_boolean ("short-header", "Short Header",
556 "Only send the basic RTSP headers for broken encoders",
557 DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
559 g_object_class_install_property (gobject_class, PROP_PROBATION,
560 g_param_spec_uint ("probation", "Number of probations",
561 "Consecutive packet sequence numbers to accept the source",
562 0, G_MAXUINT, DEFAULT_PROBATION,
563 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
565 g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
566 g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
567 "Reconnect to the server if RTSP connection is closed when doing UDP",
568 DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
570 g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
571 g_param_spec_string ("multicast-iface", "Multicast Interface",
572 "The network interface on which to join the multicast group",
573 DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
575 g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
576 g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
577 "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
578 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
580 g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
581 g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
582 "Use the pipeline running-time to set the NTP time in the RTCP SR messages",
583 DEFAULT_USE_PIPELINE_CLOCK,
584 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
586 g_object_class_install_property (gobject_class, PROP_SDES,
587 g_param_spec_boxed ("sdes", "SDES",
588 "The SDES items of this session",
589 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
592 * GstRTSPSrc::tls-validation-flags:
594 * TLS certificate validation flags used to validate server
599 g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
600 g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
601 "TLS certificate validation flags used to validate the server certificate",
602 G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS,
603 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
606 * GstRTSPSrc::tls-database:
608 * TLS database with anchor certificate authorities used to validate
609 * the server certificate.
613 g_object_class_install_property (gobject_class, PROP_TLS_DATABASE,
614 g_param_spec_object ("tls-database", "TLS database",
615 "TLS database with anchor certificate authorities used to validate the server certificate",
616 G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
619 * GstRTSPSrc::do-retransmission:
621 * Attempt to ask the server to retransmit lost packets according to RFC4588.
623 * Note: currently only works with SSRC-multiplexed retransmission streams
627 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
628 g_param_spec_boolean ("do-retransmission", "Retransmission",
629 "Ask the server to retransmit lost packets",
630 DEFAULT_DO_RETRANSMISSION,
631 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
634 * GstRTSPSrc::handle-request:
635 * @rtspsrc: a #GstRTSPSrc
636 * @request: a #GstRTSPMessage
637 * @response: a #GstRTSPMessage
639 * Handle a server request in @request and prepare @response.
641 * This signal is called from the streaming thread, you should therefore not
642 * do any state changes on @rtspsrc because this might deadlock. If you want
643 * to modify the state as a result of this signal, post a
644 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
649 gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST] =
650 g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
651 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2,
652 G_TYPE_POINTER, G_TYPE_POINTER);
655 * GstRTSPSrc::on-sdp:
656 * @rtspsrc: a #GstRTSPSrc
657 * @sdp: a #GstSDPMessage
659 * Emited when the client has retrieved the SDP and before it configures the
660 * streams in the SDP. @sdp can be inspected and modified.
662 * This signal is called from the streaming thread, you should therefore not
663 * do any state changes on @rtspsrc because this might deadlock. If you want
664 * to modify the state as a result of this signal, post a
665 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
670 gst_rtspsrc_signals[SIGNAL_ON_SDP] =
671 g_signal_new ("on-sdp", G_TYPE_FROM_CLASS (klass), 0,
672 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
673 GST_TYPE_SDP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
676 * GstRTSPSrc::select-stream:
677 * @rtspsrc: a #GstRTSPSrc
678 * @num: the stream number
679 * @caps: the stream caps
681 * Emited before the client decides to configure the stream @num with
684 * Returns: %TRUE when the stream should be selected, %FALSE when the stream
689 gst_rtspsrc_signals[SIGNAL_SELECT_STREAM] =
690 g_signal_new_class_handler ("select-stream", G_TYPE_FROM_CLASS (klass),
691 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP,
692 (GCallback) default_select_stream, select_stream_accum, NULL,
693 g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2, G_TYPE_UINT,
696 * GstRTSPSrc::new-manager:
697 * @rtspsrc: a #GstRTSPSrc
698 * @manager: a #GstElement
700 * Emited after a new manager (like rtpbin) was created and the default
701 * properties were configured.
705 gst_rtspsrc_signals[SIGNAL_NEW_MANAGER] =
706 g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass),
707 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
708 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
711 * GstRTSPSrc::request-rtcp-key:
712 * @rtspsrc: a #GstRTSPSrc
713 * @num: the stream number
715 * Signal emited to get the crypto parameters relevant to the RTCP
716 * stream. User should provide the key and the RTCP encryption ciphers
717 * and authentication, and return them wrapped in a GstCaps.
721 gst_rtspsrc_signals[SIGNAL_REQUEST_RTCP_KEY] =
722 g_signal_new ("request-rtcp-key", G_TYPE_FROM_CLASS (klass),
723 G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
725 gstelement_class->send_event = gst_rtspsrc_send_event;
726 gstelement_class->provide_clock = gst_rtspsrc_provide_clock;
727 gstelement_class->change_state = gst_rtspsrc_change_state;
729 gst_element_class_add_pad_template (gstelement_class,
730 gst_static_pad_template_get (&rtptemplate));
732 gst_element_class_set_static_metadata (gstelement_class,
733 "RTSP packet receiver", "Source/Network",
734 "Receive data over the network via RTSP (RFC 2326)",
735 "Wim Taymans <wim@fluendo.com>, "
736 "Thijs Vermeir <thijs.vermeir@barco.com>, "
737 "Lutz Mueller <lutz@topfrose.de>");
739 gstbin_class->handle_message = gst_rtspsrc_handle_message;
741 gst_rtsp_ext_list_init ();
745 gst_rtspsrc_init (GstRTSPSrc * src)
747 src->conninfo.location = g_strdup (DEFAULT_LOCATION);
748 src->protocols = DEFAULT_PROTOCOLS;
749 src->debug = DEFAULT_DEBUG;
750 src->retry = DEFAULT_RETRY;
751 src->udp_timeout = DEFAULT_TIMEOUT;
752 gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
753 src->latency = DEFAULT_LATENCY_MS;
754 src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
755 src->connection_speed = DEFAULT_CONNECTION_SPEED;
756 src->nat_method = DEFAULT_NAT_METHOD;
757 src->do_rtcp = DEFAULT_DO_RTCP;
758 src->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
759 gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
760 src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
761 src->user_id = g_strdup (DEFAULT_USER_ID);
762 src->user_pw = g_strdup (DEFAULT_USER_PW);
763 src->buffer_mode = DEFAULT_BUFFER_MODE;
764 src->client_port_range.min = 0;
765 src->client_port_range.max = 0;
766 src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
767 src->short_header = DEFAULT_SHORT_HEADER;
768 src->probation = DEFAULT_PROBATION;
769 src->udp_reconnect = DEFAULT_UDP_RECONNECT;
770 src->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
771 src->ntp_sync = DEFAULT_NTP_SYNC;
772 src->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
774 src->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
775 src->tls_database = DEFAULT_TLS_DATABASE;
776 src->do_retransmission = DEFAULT_DO_RETRANSMISSION;
778 /* get a list of all extensions */
779 src->extensions = gst_rtsp_ext_list_get ();
781 /* connect to send signal */
782 gst_rtsp_ext_list_connect (src->extensions, "send",
783 (GCallback) gst_rtspsrc_send_cb, src);
785 /* protects the streaming thread in interleaved mode or the polling
786 * thread in UDP mode. */
787 g_rec_mutex_init (&src->stream_rec_lock);
789 /* protects our state changes from multiple invocations */
790 g_rec_mutex_init (&src->state_rec_lock);
792 src->state = GST_RTSP_STATE_INVALID;
794 GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
798 gst_rtspsrc_finalize (GObject * object)
802 rtspsrc = GST_RTSPSRC (object);
804 gst_rtsp_ext_list_free (rtspsrc->extensions);
805 g_free (rtspsrc->conninfo.location);
806 gst_rtsp_url_free (rtspsrc->conninfo.url);
807 g_free (rtspsrc->conninfo.url_str);
808 g_free (rtspsrc->user_id);
809 g_free (rtspsrc->user_pw);
810 g_free (rtspsrc->multi_iface);
813 gst_sdp_message_free (rtspsrc->sdp);
816 if (rtspsrc->provided_clock)
817 gst_object_unref (rtspsrc->provided_clock);
820 gst_structure_free (rtspsrc->sdes);
822 if (rtspsrc->tls_database)
823 g_object_unref (rtspsrc->tls_database);
826 g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
827 g_rec_mutex_clear (&rtspsrc->state_rec_lock);
829 G_OBJECT_CLASS (parent_class)->finalize (object);
833 gst_rtspsrc_provide_clock (GstElement * element)
835 GstRTSPSrc *src = GST_RTSPSRC (element);
838 if ((clock = src->provided_clock) != NULL)
839 gst_object_ref (clock);
844 /* a proxy string of the format [user:passwd@]host[:port] */
846 gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
850 g_free (rtsp->proxy_user);
851 rtsp->proxy_user = NULL;
852 g_free (rtsp->proxy_passwd);
853 rtsp->proxy_passwd = NULL;
854 g_free (rtsp->proxy_host);
855 rtsp->proxy_host = NULL;
856 rtsp->proxy_port = 0;
863 /* we allow http:// in front but ignore it */
864 if (g_str_has_prefix (p, "http://"))
867 at = strchr (p, '@');
869 /* look for user:passwd */
870 col = strchr (proxy, ':');
871 if (col == NULL || col > at)
874 rtsp->proxy_user = g_strndup (p, col - p);
876 rtsp->proxy_passwd = g_strndup (col, at - col);
881 if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
882 rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
883 if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
884 rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
885 if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
886 GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
887 GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
890 col = strchr (p, ':');
893 /* everything before the colon is the hostname */
894 rtsp->proxy_host = g_strndup (p, col - p);
896 rtsp->proxy_port = strtoul (p, (char **) &p, 10);
898 rtsp->proxy_host = g_strdup (p);
899 rtsp->proxy_port = 8080;
905 gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
907 rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
908 rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
911 rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
913 rtspsrc->ptcp_timeout = NULL;
917 gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
922 rtspsrc = GST_RTSPSRC (object);
926 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
927 g_value_get_string (value), NULL);
930 rtspsrc->protocols = g_value_get_flags (value);
933 rtspsrc->debug = g_value_get_boolean (value);
936 rtspsrc->retry = g_value_get_uint (value);
939 rtspsrc->udp_timeout = g_value_get_uint64 (value);
941 case PROP_TCP_TIMEOUT:
942 gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
945 rtspsrc->latency = g_value_get_uint (value);
947 case PROP_DROP_ON_LATENCY:
948 rtspsrc->drop_on_latency = g_value_get_boolean (value);
950 case PROP_CONNECTION_SPEED:
951 rtspsrc->connection_speed = g_value_get_uint64 (value);
953 case PROP_NAT_METHOD:
954 rtspsrc->nat_method = g_value_get_enum (value);
957 rtspsrc->do_rtcp = g_value_get_boolean (value);
959 case PROP_DO_RTSP_KEEP_ALIVE:
960 rtspsrc->do_rtsp_keep_alive = g_value_get_boolean (value);
963 gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
966 if (rtspsrc->prop_proxy_id)
967 g_free (rtspsrc->prop_proxy_id);
968 rtspsrc->prop_proxy_id = g_value_dup_string (value);
971 if (rtspsrc->prop_proxy_pw)
972 g_free (rtspsrc->prop_proxy_pw);
973 rtspsrc->prop_proxy_pw = g_value_dup_string (value);
975 case PROP_RTP_BLOCKSIZE:
976 rtspsrc->rtp_blocksize = g_value_get_uint (value);
979 if (rtspsrc->user_id)
980 g_free (rtspsrc->user_id);
981 rtspsrc->user_id = g_value_dup_string (value);
984 if (rtspsrc->user_pw)
985 g_free (rtspsrc->user_pw);
986 rtspsrc->user_pw = g_value_dup_string (value);
988 case PROP_BUFFER_MODE:
989 rtspsrc->buffer_mode = g_value_get_enum (value);
991 case PROP_PORT_RANGE:
995 str = g_value_get_string (value);
997 sscanf (str, "%u-%u",
998 &rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
1000 rtspsrc->client_port_range.min = 0;
1001 rtspsrc->client_port_range.max = 0;
1005 case PROP_UDP_BUFFER_SIZE:
1006 rtspsrc->udp_buffer_size = g_value_get_int (value);
1008 case PROP_SHORT_HEADER:
1009 rtspsrc->short_header = g_value_get_boolean (value);
1011 case PROP_PROBATION:
1012 rtspsrc->probation = g_value_get_uint (value);
1014 case PROP_UDP_RECONNECT:
1015 rtspsrc->udp_reconnect = g_value_get_boolean (value);
1017 case PROP_MULTICAST_IFACE:
1018 g_free (rtspsrc->multi_iface);
1020 if (g_value_get_string (value) == NULL)
1021 rtspsrc->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
1023 rtspsrc->multi_iface = g_value_dup_string (value);
1026 rtspsrc->ntp_sync = g_value_get_boolean (value);
1028 case PROP_USE_PIPELINE_CLOCK:
1029 rtspsrc->use_pipeline_clock = g_value_get_boolean (value);
1032 rtspsrc->sdes = g_value_dup_boxed (value);
1034 case PROP_TLS_VALIDATION_FLAGS:
1035 rtspsrc->tls_validation_flags = g_value_get_flags (value);
1037 case PROP_TLS_DATABASE:
1038 g_clear_object (&rtspsrc->tls_database);
1039 rtspsrc->tls_database = g_value_dup_object (value);
1041 case PROP_DO_RETRANSMISSION:
1042 rtspsrc->do_retransmission = g_value_get_boolean (value);
1045 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1051 gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
1054 GstRTSPSrc *rtspsrc;
1056 rtspsrc = GST_RTSPSRC (object);
1060 g_value_set_string (value, rtspsrc->conninfo.location);
1062 case PROP_PROTOCOLS:
1063 g_value_set_flags (value, rtspsrc->protocols);
1066 g_value_set_boolean (value, rtspsrc->debug);
1069 g_value_set_uint (value, rtspsrc->retry);
1072 g_value_set_uint64 (value, rtspsrc->udp_timeout);
1074 case PROP_TCP_TIMEOUT:
1078 timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
1079 rtspsrc->tcp_timeout.tv_usec;
1080 g_value_set_uint64 (value, timeout);
1084 g_value_set_uint (value, rtspsrc->latency);
1086 case PROP_DROP_ON_LATENCY:
1087 g_value_set_boolean (value, rtspsrc->drop_on_latency);
1089 case PROP_CONNECTION_SPEED:
1090 g_value_set_uint64 (value, rtspsrc->connection_speed);
1092 case PROP_NAT_METHOD:
1093 g_value_set_enum (value, rtspsrc->nat_method);
1096 g_value_set_boolean (value, rtspsrc->do_rtcp);
1098 case PROP_DO_RTSP_KEEP_ALIVE:
1099 g_value_set_boolean (value, rtspsrc->do_rtsp_keep_alive);
1105 if (rtspsrc->proxy_host) {
1107 g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
1111 g_value_take_string (value, str);
1115 g_value_set_string (value, rtspsrc->prop_proxy_id);
1118 g_value_set_string (value, rtspsrc->prop_proxy_pw);
1120 case PROP_RTP_BLOCKSIZE:
1121 g_value_set_uint (value, rtspsrc->rtp_blocksize);
1124 g_value_set_string (value, rtspsrc->user_id);
1127 g_value_set_string (value, rtspsrc->user_pw);
1129 case PROP_BUFFER_MODE:
1130 g_value_set_enum (value, rtspsrc->buffer_mode);
1132 case PROP_PORT_RANGE:
1136 if (rtspsrc->client_port_range.min != 0) {
1137 str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
1138 rtspsrc->client_port_range.max);
1142 g_value_take_string (value, str);
1145 case PROP_UDP_BUFFER_SIZE:
1146 g_value_set_int (value, rtspsrc->udp_buffer_size);
1148 case PROP_SHORT_HEADER:
1149 g_value_set_boolean (value, rtspsrc->short_header);
1151 case PROP_PROBATION:
1152 g_value_set_uint (value, rtspsrc->probation);
1154 case PROP_UDP_RECONNECT:
1155 g_value_set_boolean (value, rtspsrc->udp_reconnect);
1157 case PROP_MULTICAST_IFACE:
1158 g_value_set_string (value, rtspsrc->multi_iface);
1161 g_value_set_boolean (value, rtspsrc->ntp_sync);
1163 case PROP_USE_PIPELINE_CLOCK:
1164 g_value_set_boolean (value, rtspsrc->use_pipeline_clock);
1167 g_value_set_boxed (value, rtspsrc->sdes);
1169 case PROP_TLS_VALIDATION_FLAGS:
1170 g_value_set_flags (value, rtspsrc->tls_validation_flags);
1172 case PROP_TLS_DATABASE:
1173 g_value_set_object (value, rtspsrc->tls_database);
1175 case PROP_DO_RETRANSMISSION:
1176 g_value_set_boolean (value, rtspsrc->do_retransmission);
1179 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1185 find_stream_by_id (GstRTSPStream * stream, gint * id)
1187 if (stream->id == *id)
1194 find_stream_by_channel (GstRTSPStream * stream, gint * channel)
1196 if (stream->channel[0] == *channel || stream->channel[1] == *channel)
1203 find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
1205 GstElement *src = (GstElement *) a;
1207 if (stream->udpsrc[0] == src)
1209 if (stream->udpsrc[1] == src)
1216 find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
1218 if (stream->conninfo.location) {
1219 /* check qualified setup_url */
1220 if (!strcmp (stream->conninfo.location, (gchar *) a))
1223 if (stream->control_url) {
1224 /* check original control_url */
1225 if (!strcmp (stream->control_url, (gchar *) a))
1228 /* check if qualified setup_url ends with string */
1229 if (g_str_has_suffix (stream->control_url, (gchar *) a))
1236 static GstRTSPStream *
1237 find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
1241 /* find and get stream */
1242 if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
1243 return (GstRTSPStream *) lstream->data;
1248 static const GstSDPBandwidth *
1249 gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1250 const GstSDPMedia * media, const gchar * type)
1254 /* first look in the media specific section */
1255 len = gst_sdp_media_bandwidths_len (media);
1256 for (i = 0; i < len; i++) {
1257 const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
1259 if (strcmp (bw->bwtype, type) == 0)
1262 /* then look in the message specific section */
1263 len = gst_sdp_message_bandwidths_len (sdp);
1264 for (i = 0; i < len; i++) {
1265 const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
1267 if (strcmp (bw->bwtype, type) == 0)
1274 gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1275 const GstSDPMedia * media, GstRTSPStream * stream)
1277 const GstSDPBandwidth *bw;
1279 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
1280 stream->as_bandwidth = bw->bandwidth;
1282 stream->as_bandwidth = -1;
1284 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
1285 stream->rr_bandwidth = bw->bandwidth;
1287 stream->rr_bandwidth = -1;
1289 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
1290 stream->rs_bandwidth = bw->bandwidth;
1292 stream->rs_bandwidth = -1;
1296 gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
1297 const GstSDPConnection * conn)
1299 if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
1302 if (conn->addrtype == NULL)
1305 /* check for IPV6 */
1306 if (strcmp (conn->addrtype, "IP4") == 0)
1307 stream->is_ipv6 = FALSE;
1308 else if (strcmp (conn->addrtype, "IP6") == 0)
1309 stream->is_ipv6 = TRUE;
1314 g_free (stream->destination);
1315 stream->destination = g_strdup (conn->address);
1317 /* check for multicast */
1318 stream->is_multicast =
1319 gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
1321 stream->ttl = conn->ttl;
1324 /* Go over the connections for a stream.
1325 * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
1327 * - If we are dealing with a localhost address, we disable multicast
1330 gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
1331 const GstSDPMedia * media, GstRTSPStream * stream)
1333 const GstSDPConnection *conn;
1336 /* first look in the media specific section */
1337 len = gst_sdp_media_connections_len (media);
1338 for (i = 0; i < len; i++) {
1339 conn = gst_sdp_media_get_connection (media, i);
1341 gst_rtspsrc_do_stream_connection (src, stream, conn);
1343 /* then look in the message specific section */
1344 if ((conn = gst_sdp_message_get_connection (sdp))) {
1345 gst_rtspsrc_do_stream_connection (src, stream, conn);
1349 /* m=<media> <UDP port> RTP/AVP <payload>
1352 gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
1353 const GstSDPMedia * media, GstRTSPStream * stream)
1359 proto = gst_sdp_media_get_proto (media);
1363 if (g_str_equal (proto, "RTP/AVP"))
1364 stream->profile = GST_RTSP_PROFILE_AVP;
1365 else if (g_str_equal (proto, "RTP/SAVP"))
1366 stream->profile = GST_RTSP_PROFILE_SAVP;
1367 else if (g_str_equal (proto, "RTP/AVPF"))
1368 stream->profile = GST_RTSP_PROFILE_AVPF;
1369 else if (g_str_equal (proto, "RTP/SAVPF"))
1370 stream->profile = GST_RTSP_PROFILE_SAVPF;
1374 len = gst_sdp_media_formats_len (media);
1375 for (i = 0; i < len; i++) {
1382 pt = atoi (gst_sdp_media_get_format (media, i));
1384 GST_DEBUG_OBJECT (src, " looking at %d pt: %d", i, pt);
1387 caps = gst_rtspsrc_media_to_caps (pt, media);
1389 GST_WARNING_OBJECT (src, " skipping pt %d without caps", pt);
1393 /* do some tweaks */
1394 s = gst_caps_get_structure (caps, 0);
1395 if ((enc = gst_structure_get_string (s, "encoding-name"))) {
1396 stream->is_real = (strstr (enc, "-REAL") != NULL);
1397 if (strcmp (enc, "X-ASF-PF") == 0)
1398 stream->container = TRUE;
1400 GST_DEBUG ("mapping sdp session level attributes to caps");
1401 gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, caps);
1402 GST_DEBUG ("mapping sdp media level attributes to caps");
1403 gst_rtspsrc_sdp_attributes_to_caps (media->attributes, caps);
1405 /* the first pt will be the default */
1406 if (stream->ptmap->len == 0)
1407 stream->default_pt = pt;
1411 g_array_append_val (stream->ptmap, item);
1417 GST_ERROR_OBJECT (src, "can't find proto in media");
1422 GST_ERROR_OBJECT (src, "unknown proto in media %s", proto);
1427 static const gchar *
1428 get_aggregate_control (GstRTSPSrc * src)
1433 base = src->control;
1434 else if (src->content_base)
1435 base = src->content_base;
1436 else if (src->conninfo.url_str)
1437 base = src->conninfo.url_str;
1445 clear_ptmap_item (PtMapItem * item)
1448 gst_caps_unref (item->caps);
1451 static GstRTSPStream *
1452 gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
1454 GstRTSPStream *stream;
1455 const gchar *control_url;
1456 const GstSDPMedia *media;
1458 /* get media, should not return NULL */
1459 media = gst_sdp_message_get_media (sdp, idx);
1463 stream = g_new0 (GstRTSPStream, 1);
1464 stream->parent = src;
1465 /* we mark the pad as not linked, we will mark it as OK when we add the pad to
1467 stream->last_ret = GST_FLOW_NOT_LINKED;
1468 stream->added = FALSE;
1469 stream->setup = FALSE;
1470 stream->skipped = FALSE;
1472 stream->eos = FALSE;
1473 stream->discont = TRUE;
1474 stream->seqbase = -1;
1475 stream->timebase = -1;
1476 stream->send_ssrc = g_random_int ();
1477 stream->profile = GST_RTSP_PROFILE_AVP;
1478 stream->ptmap = g_array_new (FALSE, FALSE, sizeof (PtMapItem));
1479 g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);
1481 /* collect bandwidth information for this steam. FIXME, configure in the RTP
1482 * session manager to scale RTCP. */
1483 gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
1485 /* collect connection info */
1486 gst_rtspsrc_collect_connections (src, sdp, media, stream);
1488 /* make the payload type map */
1489 gst_rtspsrc_collect_payloads (src, sdp, media, stream);
1491 /* collect port number */
1492 stream->port = gst_sdp_media_get_port (media);
1494 /* get control url to construct the setup url. The setup url is used to
1495 * configure the transport of the stream and is used to identity the stream in
1496 * the RTP-Info header field returned from PLAY. */
1497 control_url = gst_sdp_media_get_attribute_val (media, "control");
1498 if (control_url == NULL)
1499 control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
1501 GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
1502 GST_DEBUG_OBJECT (src, " port: %d", stream->port);
1503 GST_DEBUG_OBJECT (src, " container: %d", stream->container);
1504 GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
1506 if (control_url != NULL) {
1507 stream->control_url = g_strdup (control_url);
1508 /* Build a fully qualified url using the content_base if any or by prefixing
1509 * the original request.
1510 * If the control_url starts with a '/' or a non rtsp: protocol we will most
1511 * likely build a URL that the server will fail to understand, this is ok,
1512 * we will fail then. */
1513 if (g_str_has_prefix (control_url, "rtsp://"))
1514 stream->conninfo.location = g_strdup (control_url);
1519 if (g_strcmp0 (control_url, "*") == 0)
1522 base = get_aggregate_control (src);
1524 /* check if the base ends or control starts with / */
1525 has_slash = g_str_has_prefix (control_url, "/");
1526 has_slash = has_slash || g_str_has_suffix (base, "/");
1528 /* concatenate the two strings, insert / when not present */
1529 stream->conninfo.location =
1530 g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
1533 GST_DEBUG_OBJECT (src, " setup: %s",
1534 GST_STR_NULL (stream->conninfo.location));
1536 /* we keep track of all streams */
1537 src->streams = g_list_append (src->streams, stream);
1545 gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
1549 GST_DEBUG_OBJECT (src, "free stream %p", stream);
1551 g_array_free (stream->ptmap, TRUE);
1553 g_free (stream->destination);
1554 g_free (stream->control_url);
1555 g_free (stream->conninfo.location);
1557 for (i = 0; i < 2; i++) {
1558 if (stream->udpsrc[i]) {
1559 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
1560 gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
1561 gst_object_unref (stream->udpsrc[i]);
1563 if (stream->channelpad[i])
1564 gst_object_unref (stream->channelpad[i]);
1566 if (stream->udpsink[i]) {
1567 gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
1568 gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
1569 gst_object_unref (stream->udpsink[i]);
1572 if (stream->fakesrc) {
1573 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
1574 gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
1575 gst_object_unref (stream->fakesrc);
1577 if (stream->srcpad) {
1578 gst_pad_set_active (stream->srcpad, FALSE);
1580 gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
1582 if (stream->srtpenc)
1583 gst_object_unref (stream->srtpenc);
1584 if (stream->srtpdec)
1585 gst_object_unref (stream->srtpdec);
1586 if (stream->srtcpparams)
1587 gst_caps_unref (stream->srtcpparams);
1588 if (stream->rtcppad)
1589 gst_object_unref (stream->rtcppad);
1590 if (stream->session)
1591 g_object_unref (stream->session);
1592 if (stream->rtx_pt_map)
1593 gst_structure_free (stream->rtx_pt_map);
1598 gst_rtspsrc_cleanup (GstRTSPSrc * src)
1602 GST_DEBUG_OBJECT (src, "cleanup");
1604 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1605 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1607 gst_rtspsrc_stream_free (src, stream);
1609 g_list_free (src->streams);
1610 src->streams = NULL;
1612 if (src->manager_sig_id) {
1613 g_signal_handler_disconnect (src->manager, src->manager_sig_id);
1614 src->manager_sig_id = 0;
1616 gst_element_set_state (src->manager, GST_STATE_NULL);
1617 gst_bin_remove (GST_BIN_CAST (src), src->manager);
1618 src->manager = NULL;
1621 gst_structure_free (src->props);
1624 g_free (src->content_base);
1625 src->content_base = NULL;
1627 g_free (src->control);
1628 src->control = NULL;
1631 gst_rtsp_range_free (src->range);
1634 /* don't clear the SDP when it was used in the url */
1635 if (src->sdp && !src->from_sdp) {
1636 gst_sdp_message_free (src->sdp);
1639 if (src->start_segment) {
1640 gst_event_unref (src->start_segment);
1641 src->start_segment = NULL;
1643 if (src->provided_clock) {
1644 gst_object_unref (src->provided_clock);
1645 src->provided_clock = NULL;
1649 #define PARSE_INT(p, del, res) \
1652 p = strstr (p, del); \
1662 #define PARSE_STRING(p, del, res) \
1665 p = strstr (p, del); \
1677 #define SKIP_SPACES(p) \
1678 while (*p && g_ascii_isspace (*p)) \
1683 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1686 gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
1687 gint * rate, gchar ** params)
1691 p = (gchar *) rtpmap;
1693 PARSE_INT (p, " ", *payload);
1701 PARSE_STRING (p, "/", *name);
1702 if (*name == NULL) {
1703 GST_DEBUG ("no rate, name %s", p);
1704 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
1705 * streams seem to omit the rate. */
1712 p = strstr (p, "/");
1730 parse_keymgmt (const gchar * keymgmt, GstCaps * caps)
1732 gboolean res = FALSE;
1736 GstMIKEYMessage *msg;
1737 const GstMIKEYPayload *payload;
1738 const gchar *srtp_cipher;
1739 const gchar *srtp_auth;
1741 p = (gchar *) keymgmt;
1747 PARSE_STRING (p, " ", kmpid);
1748 if (!g_str_equal (kmpid, "mikey"))
1751 data = g_base64_decode (p, &size);
1755 msg = gst_mikey_message_new_from_data (data, size, NULL, NULL);
1760 srtp_cipher = "aes-128-icm";
1761 srtp_auth = "hmac-sha1-80";
1763 /* check the Security policy if any */
1764 if ((payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, 0))) {
1765 GstMIKEYPayloadSP *p = (GstMIKEYPayloadSP *) payload;
1768 if (p->proto != GST_MIKEY_SEC_PROTO_SRTP)
1771 len = gst_mikey_payload_sp_get_n_params (payload);
1772 for (i = 0; i < len; i++) {
1773 const GstMIKEYPayloadSPParam *param =
1774 gst_mikey_payload_sp_get_param (payload, i);
1776 switch (param->type) {
1777 case GST_MIKEY_SP_SRTP_ENC_ALG:
1778 switch (param->val[0]) {
1780 srtp_cipher = "null";
1784 srtp_cipher = "aes-128-icm";
1790 case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
1791 switch (param->val[0]) {
1792 case AES_128_KEY_LEN:
1793 srtp_cipher = "aes-128-icm";
1795 case AES_256_KEY_LEN:
1796 srtp_cipher = "aes-256-icm";
1802 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1803 switch (param->val[0]) {
1809 srtp_auth = "hmac-sha1-80";
1815 case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
1816 switch (param->val[0]) {
1817 case HMAC_32_KEY_LEN:
1818 srtp_auth = "hmac-sha1-32";
1820 case HMAC_80_KEY_LEN:
1821 srtp_auth = "hmac-sha1-80";
1827 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1829 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1837 if (!(payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_KEMAC, 0)))
1840 GstMIKEYPayloadKEMAC *p = (GstMIKEYPayloadKEMAC *) payload;
1841 const GstMIKEYPayload *sub;
1842 GstMIKEYPayloadKeyData *pkd;
1845 if (p->enc_alg != GST_MIKEY_ENC_NULL || p->mac_alg != GST_MIKEY_MAC_NULL)
1848 if (!(sub = gst_mikey_payload_kemac_get_sub (payload, 0)))
1851 if (sub->type != GST_MIKEY_PT_KEY_DATA)
1854 pkd = (GstMIKEYPayloadKeyData *) sub;
1856 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
1858 gst_caps_set_simple (caps, "srtp-key", GST_TYPE_BUFFER, buf, NULL);
1861 gst_caps_set_simple (caps,
1862 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
1863 "srtp-auth", G_TYPE_STRING, srtp_auth,
1864 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
1865 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
1869 gst_mikey_message_unref (msg);
1875 * Mapping SDP attributes to caps
1877 * prepend 'a-' to IANA registered sdp attributes names
1878 * (ie: not prefixed with 'x-') in order to avoid
1879 * collision with gstreamer standard caps properties names
1882 gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
1884 if (attributes->len > 0) {
1888 s = gst_caps_get_structure (caps, 0);
1890 for (i = 0; i < attributes->len; i++) {
1891 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
1892 gchar *tofree, *key;
1896 /* skip some of the attribute we already handle */
1897 if (!strcmp (key, "fmtp"))
1899 if (!strcmp (key, "rtpmap"))
1901 if (!strcmp (key, "control"))
1903 if (!strcmp (key, "range"))
1905 if (!strcmp (key, "framesize"))
1907 if (g_str_equal (key, "key-mgmt")) {
1908 parse_keymgmt (attr->value, caps);
1912 /* string must be valid UTF8 */
1913 if (!g_utf8_validate (attr->value, -1, NULL))
1916 if (!g_str_has_prefix (key, "x-"))
1917 tofree = key = g_strdup_printf ("a-%s", key);
1921 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
1922 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
1928 static const gchar *
1929 rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
1938 if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
1941 if (sscanf (attr, "%d ", &val) != 1)
1951 * Mapping of caps to and from SDP fields:
1953 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1954 * a=framesize:<payload> <width>-<height>
1955 * a=fmtp:<payload> <param>[=<value>];...
1958 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
1961 const gchar *rtpmap;
1963 const gchar *framesize;
1966 gchar *params = NULL;
1972 /* get and parse rtpmap */
1973 rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
1976 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
1978 g_warning ("error parsing rtpmap, ignoring");
1982 /* dynamic payloads need rtpmap or we fail */
1983 if (rtpmap == NULL && pt >= 96)
1986 /* check if we have a rate, if not, we need to look up the rate from the
1987 * default rates based on the payload types. */
1989 const GstRTPPayloadInfo *info;
1991 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
1992 /* dynamic types, use media and encoding_name */
1993 tmp = g_ascii_strdown (media->media, -1);
1994 info = gst_rtp_payload_info_for_name (tmp, name);
1997 /* static types, use payload type */
1998 info = gst_rtp_payload_info_for_pt (pt);
2002 if ((rate = info->clock_rate) == 0)
2005 /* we fail if we cannot find one */
2010 tmp = g_ascii_strdown (media->media, -1);
2011 caps = gst_caps_new_simple ("application/x-unknown",
2012 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
2014 s = gst_caps_get_structure (caps, 0);
2016 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
2018 /* encoding name must be upper case */
2020 tmp = g_ascii_strup (name, -1);
2021 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
2025 /* params must be lower case */
2026 if (params != NULL) {
2027 tmp = g_ascii_strdown (params, -1);
2028 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
2032 /* parse optional fmtp: field */
2033 if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
2039 /* p is now of the format <payload> <param>[=<value>];... */
2040 PARSE_INT (p, " ", payload);
2041 if (payload != -1 && payload == pt) {
2045 /* <param>[=<value>] are separated with ';' */
2046 pairs = g_strsplit (p, ";", 0);
2047 for (i = 0; pairs[i]; i++) {
2049 const gchar *val, *key;
2051 /* the key may not have a '=', the value can have other '='s */
2052 valpos = strstr (pairs[i], "=");
2054 /* we have a '=' and thus a value, remove the '=' with \0 */
2056 /* value is everything between '=' and ';'. We split the pairs at ;
2057 * boundaries so we can take the remainder of the value. Some servers
2058 * put spaces around the value which we strip off here. Alternatively
2059 * we could strip those spaces in the depayloaders should these spaces
2060 * actually carry any meaning in the future. */
2061 val = g_strstrip (valpos + 1);
2063 /* simple <param>;.. is translated into <param>=1;... */
2066 /* strip the key of spaces, convert key to lowercase but not the value. */
2067 key = g_strstrip (pairs[i]);
2068 if (strlen (key) > 1) {
2069 tmp = g_ascii_strdown (key, -1);
2070 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
2078 /* parse framesize: field */
2079 if ((framesize = gst_sdp_media_get_attribute_val (media, "framesize"))) {
2082 /* p is now of the format <payload> <width>-<height> */
2083 p = (gchar *) framesize;
2085 PARSE_INT (p, " ", payload);
2086 if (payload != -1 && payload == pt) {
2087 gst_structure_set (s, "a-framesize", G_TYPE_STRING, p, NULL);
2095 g_warning ("rtpmap type not given for dynamic payload %d", pt);
2100 g_warning ("rate unknown for payload type %d", pt);
2106 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
2107 gint * rtpport, gint * rtcpport)
2110 GstStateChangeReturn ret;
2111 GstElement *udpsrc0, *udpsrc1;
2112 gint tmp_rtp, tmp_rtcp;
2116 src = stream->parent;
2122 /* Start at next port */
2123 tmp_rtp = src->next_port_num;
2125 if (stream->is_ipv6)
2126 host = "udp://[::0]";
2128 host = "udp://0.0.0.0";
2130 /* try to allocate 2 UDP ports, the RTP port should be an even
2131 * number and the RTCP port should be the next (uneven) port */
2134 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
2135 tmp_rtp >= src->client_port_range.max)
2138 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2139 if (udpsrc0 == NULL)
2140 goto no_udp_protocol;
2141 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
2143 if (src->udp_buffer_size != 0)
2144 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
2147 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
2148 if (ret == GST_STATE_CHANGE_FAILURE) {
2150 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
2153 if (++count > src->retry)
2156 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2157 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2158 gst_object_unref (udpsrc0);
2161 GST_DEBUG_OBJECT (src, "retry %d", count);
2164 goto no_udp_protocol;
2167 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
2168 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
2170 /* check if port is even */
2171 if ((tmp_rtp & 0x01) != 0) {
2172 /* port not even, close and allocate another */
2173 if (++count > src->retry)
2176 GST_DEBUG_OBJECT (src, "RTP port not even");
2178 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2179 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2180 gst_object_unref (udpsrc0);
2183 GST_DEBUG_OBJECT (src, "retry %d", count);
2188 /* allocate port+1 for RTCP now */
2189 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2190 if (udpsrc1 == NULL)
2191 goto no_udp_rtcp_protocol;
2194 tmp_rtcp = tmp_rtp + 1;
2195 if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
2198 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
2200 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
2201 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
2202 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
2203 if (ret == GST_STATE_CHANGE_FAILURE) {
2204 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
2206 if (++count > src->retry)
2209 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2210 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2211 gst_object_unref (udpsrc0);
2214 GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
2215 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2216 gst_object_unref (udpsrc1);
2220 GST_DEBUG_OBJECT (src, "retry %d", count);
2224 /* all fine, do port check */
2225 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
2226 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
2228 /* this should not happen... */
2229 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
2232 /* we keep these elements, we configure all in configure_transport when the
2233 * server told us to really use the UDP ports. */
2234 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
2235 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
2236 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
2237 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
2239 /* keep track of next available port number when we have a range
2241 if (src->next_port_num != 0)
2242 src->next_port_num = tmp_rtcp + 1;
2249 GST_DEBUG_OBJECT (src, "could not get UDP source");
2254 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
2258 no_udp_rtcp_protocol:
2260 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
2265 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
2266 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
2272 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2273 gst_object_unref (udpsrc0);
2276 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2277 gst_object_unref (udpsrc1);
2284 gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
2289 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
2291 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2292 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2295 for (i = 0; i < 2; i++) {
2296 if (stream->udpsrc[i])
2297 gst_element_set_state (stream->udpsrc[i], state);
2303 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
2310 event = gst_event_new_flush_start ();
2311 GST_DEBUG_OBJECT (src, "start flush");
2313 state = GST_STATE_PAUSED;
2315 event = gst_event_new_flush_stop (FALSE);
2316 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
2319 state = GST_STATE_PLAYING;
2321 state = GST_STATE_PAUSED;
2323 gst_rtspsrc_push_event (src, event);
2324 gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
2325 gst_rtspsrc_set_state (src, state);
2328 static GstRTSPResult
2329 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
2330 GstRTSPMessage * message, GTimeVal * timeout)
2335 ret = gst_rtsp_connection_send (conn, message, timeout);
2337 ret = GST_RTSP_ERROR;
2342 static GstRTSPResult
2343 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
2344 GstRTSPMessage * message, GTimeVal * timeout)
2349 ret = gst_rtsp_connection_receive (conn, message, timeout);
2351 ret = GST_RTSP_ERROR;
2357 gst_rtspsrc_get_position (GstRTSPSrc * src)
2362 query = gst_query_new_position (GST_FORMAT_TIME);
2363 /* should be known somewhere down the stream (e.g. jitterbuffer) */
2364 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2365 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2369 if (stream->srcpad) {
2370 if (gst_pad_query (stream->srcpad, query)) {
2371 gst_query_parse_position (query, &fmt, &pos);
2372 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
2373 GST_TIME_ARGS (pos));
2374 src->last_pos = pos;
2384 gst_query_unref (query);
2388 gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
2390 src->state = GST_RTSP_STATE_SEEKING;
2391 /* PLAY will add the range header now. */
2392 src->need_range = TRUE;
2398 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
2403 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
2405 gboolean flush, skip;
2408 GstSegment seeksegment = { 0, };
2412 GST_DEBUG_OBJECT (src, "doing seek with event");
2414 gst_event_parse_seek (event, &rate, &format, &flags,
2415 &cur_type, &cur, &stop_type, &stop);
2417 /* no negative rates yet */
2421 /* we need TIME format */
2422 if (format != src->segment.format)
2425 GST_DEBUG_OBJECT (src, "doing seek without event");
2427 cur_type = GST_SEEK_TYPE_SET;
2428 stop_type = GST_SEEK_TYPE_SET;
2431 /* get flush flag */
2432 flush = flags & GST_SEEK_FLAG_FLUSH;
2433 skip = flags & GST_SEEK_FLAG_SKIP;
2435 /* now we need to make sure the streaming thread is stopped. We do this by
2436 * either sending a FLUSH_START event downstream which will cause the
2437 * streaming thread to stop with a WRONG_STATE.
2438 * For a non-flushing seek we simply pause the task, which will happen as soon
2439 * as it completes one iteration (and thus might block when the sink is
2440 * blocking in preroll). */
2442 GST_DEBUG_OBJECT (src, "starting flush");
2443 gst_rtspsrc_flush (src, TRUE, FALSE);
2446 gst_task_pause (src->task);
2450 /* we should now be able to grab the streaming thread because we stopped it
2451 * with the above flush/pause code */
2452 GST_RTSP_STREAM_LOCK (src);
2454 GST_DEBUG_OBJECT (src, "stopped streaming");
2456 /* stop flushing the rtsp connection so we can send PAUSE/PLAY below */
2457 gst_rtspsrc_connection_flush (src, FALSE);
2459 /* copy segment, we need this because we still need the old
2460 * segment when we close the current segment. */
2461 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
2463 /* configure the seek parameters in the seeksegment. We will then have the
2464 * right values in the segment to perform the seek */
2466 GST_DEBUG_OBJECT (src, "configuring seek");
2467 gst_segment_do_seek (&seeksegment, rate, format, flags,
2468 cur_type, cur, stop_type, stop, &update);
2471 /* figure out the last position we need to play. If it's configured (stop !=
2472 * -1), use that, else we play until the total duration of the file */
2473 if ((stop = seeksegment.stop) == -1)
2474 stop = seeksegment.duration;
2476 playing = (src->state == GST_RTSP_STATE_PLAYING);
2478 /* if we were playing, pause first */
2480 /* obtain current position in case seek fails */
2481 gst_rtspsrc_get_position (src);
2482 gst_rtspsrc_pause (src, FALSE);
2486 gst_rtspsrc_do_seek (src, &seeksegment);
2488 /* and continue playing */
2490 gst_rtspsrc_play (src, &seeksegment, FALSE);
2492 /* prepare for streaming again */
2494 /* if we started flush, we stop now */
2495 GST_DEBUG_OBJECT (src, "stopping flush");
2496 gst_rtspsrc_flush (src, FALSE, playing);
2499 /* now we did the seek and can activate the new segment values */
2500 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
2502 /* if we're doing a segment seek, post a SEGMENT_START message */
2503 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2504 gst_element_post_message (GST_ELEMENT_CAST (src),
2505 gst_message_new_segment_start (GST_OBJECT_CAST (src),
2506 src->segment.format, src->segment.position));
2509 /* now create the newsegment */
2510 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
2511 " to %" G_GINT64_FORMAT, src->segment.position, stop);
2514 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
2515 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2516 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2517 stream->discont = TRUE;
2520 GST_RTSP_STREAM_UNLOCK (src);
2527 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
2532 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
2538 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
2542 gboolean res = TRUE;
2545 src = GST_RTSPSRC_CAST (parent);
2547 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
2548 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
2550 switch (GST_EVENT_TYPE (event)) {
2551 case GST_EVENT_SEEK:
2552 res = gst_rtspsrc_perform_seek (src, event);
2556 case GST_EVENT_NAVIGATION:
2557 case GST_EVENT_LATENCY:
2565 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
2566 res = gst_pad_send_event (target, event);
2567 gst_object_unref (target);
2569 gst_event_unref (event);
2572 gst_event_unref (event);
2578 /* this is the final event function we receive on the internal source pad when
2579 * we deal with TCP connections */
2581 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
2586 GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
2588 switch (GST_EVENT_TYPE (event)) {
2589 case GST_EVENT_SEEK:
2591 case GST_EVENT_NAVIGATION:
2592 case GST_EVENT_LATENCY:
2594 gst_event_unref (event);
2601 /* this is the final query function we receive on the internal source pad when
2602 * we deal with TCP connections */
2604 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
2608 gboolean res = TRUE;
2610 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
2612 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2613 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2615 switch (GST_QUERY_TYPE (query)) {
2616 case GST_QUERY_POSITION:
2621 case GST_QUERY_DURATION:
2625 gst_query_parse_duration (query, &format, NULL);
2628 case GST_FORMAT_TIME:
2629 gst_query_set_duration (query, format, src->segment.duration);
2637 case GST_QUERY_LATENCY:
2639 /* we are live with a min latency of 0 and unlimited max latency, this
2640 * result will be updated by the session manager if there is any. */
2641 gst_query_set_latency (query, TRUE, 0, -1);
2651 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2653 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2657 gboolean res = FALSE;
2659 src = GST_RTSPSRC_CAST (parent);
2661 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2662 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2664 switch (GST_QUERY_TYPE (query)) {
2665 case GST_QUERY_DURATION:
2669 gst_query_parse_duration (query, &format, NULL);
2672 case GST_FORMAT_TIME:
2673 gst_query_set_duration (query, format, src->segment.duration);
2681 case GST_QUERY_SEEKING:
2685 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2686 if (format == GST_FORMAT_TIME) {
2688 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2690 /* seeking without duration is unlikely */
2691 seekable = seekable && src->seekable && src->segment.duration &&
2692 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2694 /* FIXME ?? should we have 0 and segment.duration here; see demuxers */
2695 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
2696 src->segment.start, src->segment.stop);
2705 uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
2707 gst_query_set_uri (query, uri);
2715 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2717 /* forward the query to the proxy target pad */
2719 res = gst_pad_query (target, query);
2720 gst_object_unref (target);
2729 /* callback for RTCP messages to be sent to the server when operating in TCP
2731 static GstFlowReturn
2732 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2735 GstRTSPStream *stream;
2736 GstFlowReturn res = GST_FLOW_OK;
2741 GstRTSPMessage message = { 0 };
2742 GstRTSPConnection *conn;
2744 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2745 src = stream->parent;
2747 gst_buffer_map (buffer, &map, GST_MAP_READ);
2751 gst_rtsp_message_init_data (&message, stream->channel[1]);
2753 /* lend the body data to the message */
2754 gst_rtsp_message_take_body (&message, data, size);
2756 if (stream->conninfo.connection)
2757 conn = stream->conninfo.connection;
2759 conn = src->conninfo.connection;
2761 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2762 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2763 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2765 /* and steal it away again because we will free it when unreffing the
2767 gst_rtsp_message_steal_body (&message, &data, &size);
2768 gst_rtsp_message_unset (&message);
2770 gst_buffer_unmap (buffer, &map);
2771 gst_buffer_unref (buffer);
2776 static GstPadProbeReturn
2777 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2779 GstRTSPSrc *src = user_data;
2781 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2782 GST_DEBUG_PAD_NAME (pad));
2784 /* activate the streams */
2785 GST_OBJECT_LOCK (src);
2786 if (!src->need_activate)
2789 src->need_activate = FALSE;
2790 GST_OBJECT_UNLOCK (src);
2792 gst_rtspsrc_activate_streams (src);
2794 return GST_PAD_PROBE_OK;
2798 GST_OBJECT_UNLOCK (src);
2799 return GST_PAD_PROBE_OK;
2804 copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
2806 GstPad *gpad = GST_PAD_CAST (user_data);
2808 GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
2809 gst_pad_store_sticky_event (gpad, *event);
2814 /* this callback is called when the session manager generated a new src pad with
2815 * payloaded RTP packets. We simply ghost the pad here. */
2817 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2820 GstPadTemplate *template;
2823 GstRTSPStream *stream;
2826 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2828 GST_RTSP_STATE_LOCK (src);
2830 name = gst_object_get_name (GST_OBJECT_CAST (pad));
2831 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
2832 goto unknown_stream;
2834 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);
2836 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
2838 goto unknown_stream;
2841 stream->ssrc = ssrc;
2843 /* we'll add it later see below */
2844 stream->added = TRUE;
2846 /* check if we added all streams */
2848 for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
2849 GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;
2851 GST_DEBUG_OBJECT (src, "stream %p, container %d, added %d, setup %d",
2852 ostream, ostream->container, ostream->added, ostream->setup);
2854 /* if we find a stream for which we did a setup that is not added, we
2855 * need to wait some more */
2856 if (ostream->setup && !ostream->added) {
2861 GST_RTSP_STATE_UNLOCK (src);
2863 /* create a new pad we will use to stream to */
2864 template = gst_static_pad_template_get (&rtptemplate);
2865 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
2866 gst_object_unref (template);
2869 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
2870 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
2871 gst_pad_set_active (stream->srcpad, TRUE);
2872 gst_pad_sticky_events_foreach (pad, copy_sticky_events, stream->srcpad);
2873 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
2876 GST_DEBUG_OBJECT (src, "We added all streams");
2877 /* when we get here, all stream are added and we can fire the no-more-pads
2879 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
2887 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
2888 GST_RTSP_STATE_UNLOCK (src);
2895 stream_get_caps_for_pt (GstRTSPStream * stream, guint pt)
2899 len = stream->ptmap->len;
2900 for (i = 0; i < len; i++) {
2901 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
2909 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
2911 GstRTSPStream *stream;
2914 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
2916 GST_RTSP_STATE_LOCK (src);
2917 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2919 goto unknown_stream;
2921 if ((caps = stream_get_caps_for_pt (stream, pt)))
2922 gst_caps_ref (caps);
2923 GST_RTSP_STATE_UNLOCK (src);
2929 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
2930 GST_RTSP_STATE_UNLOCK (src);
2936 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
2938 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
2944 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
2950 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
2956 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2958 GstRTSPSrc *src = stream->parent;
2961 g_object_get (source, "ssrc", &ssrc, NULL);
2963 GST_DEBUG_OBJECT (src, "source %08x, stream %08x, session %u received BYE",
2964 ssrc, stream->ssrc, stream->id);
2966 if (ssrc == stream->ssrc)
2967 gst_rtspsrc_do_stream_eos (src, stream);
2971 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2973 GstRTSPSrc *src = stream->parent;
2976 g_object_get (source, "ssrc", &ssrc, NULL);
2978 GST_WARNING_OBJECT (src, "source %08x, stream %08x in session %u timed out",
2979 ssrc, stream->ssrc, stream->id);
2981 if (ssrc == stream->ssrc)
2982 gst_rtspsrc_do_stream_eos (src, stream);
2986 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
2988 GstRTSPStream *stream;
2990 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
2992 /* get stream for session */
2993 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2995 gst_rtspsrc_do_stream_eos (src, stream);
3000 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
3002 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
3007 set_manager_buffer_mode (GstRTSPSrc * src)
3009 GObjectClass *klass;
3011 if (src->manager == NULL)
3014 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
3016 if (!g_object_class_find_property (klass, "buffer-mode"))
3019 if (src->buffer_mode != BUFFER_MODE_AUTO) {
3020 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
3025 GST_DEBUG_OBJECT (src,
3026 "auto buffering mode, have clock %" GST_PTR_FORMAT, src->provided_clock);
3028 if (src->provided_clock) {
3029 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
3031 if (clock == src->provided_clock) {
3032 GST_DEBUG_OBJECT (src, "selected synced");
3033 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SYNCED, NULL);
3036 gst_object_unref (clock);
3041 /* Otherwise fall-through and use another buffer mode */
3043 gst_object_unref (clock);
3046 GST_DEBUG_OBJECT (src, "auto buffering mode");
3047 if (src->use_buffering) {
3048 GST_DEBUG_OBJECT (src, "selected buffer");
3049 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER, NULL);
3051 GST_DEBUG_OBJECT (src, "selected slave");
3052 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
3057 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
3059 GST_DEBUG ("request key %u", ssrc);
3060 return gst_caps_ref (stream_get_caps_for_pt (stream, stream->default_pt));
3064 request_rtp_decoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
3066 GST_DEBUG ("decoder session %u, stream %p, %d", session, stream, stream->id);
3067 if (stream->id != session)
3070 if (stream->profile != GST_RTSP_PROFILE_SAVP &&
3071 stream->profile != GST_RTSP_PROFILE_SAVPF)
3074 if (stream->srtpdec == NULL) {
3077 name = g_strdup_printf ("srtpdec_%u", session);
3078 stream->srtpdec = gst_element_factory_make ("srtpdec", name);
3081 g_signal_connect (stream->srtpdec, "request-key",
3082 (GCallback) request_key, stream);
3084 return gst_object_ref (stream->srtpdec);
3088 request_rtcp_encoder (GstElement * rtpbin, guint session,
3089 GstRTSPStream * stream)
3094 GST_DEBUG ("decoder session %u, stream %p, %d", session, stream, stream->id);
3095 if (stream->id != session)
3098 if (stream->profile != GST_RTSP_PROFILE_SAVP &&
3099 stream->profile != GST_RTSP_PROFILE_SAVPF)
3102 if (stream->srtpenc == NULL) {
3105 name = g_strdup_printf ("srtpenc_%u", session);
3106 stream->srtpenc = gst_element_factory_make ("srtpenc", name);
3109 /* get RTCP crypto parameters from caps */
3110 s = gst_caps_get_structure (stream->srtcpparams, 0);
3114 GType ciphertype, authtype;
3115 GValue rtcp_cipher = G_VALUE_INIT, rtcp_auth = G_VALUE_INIT;
3117 ciphertype = g_type_from_name ("GstSrtpCipherType");
3118 authtype = g_type_from_name ("GstSrtpAuthType");
3119 g_value_init (&rtcp_cipher, ciphertype);
3120 g_value_init (&rtcp_auth, authtype);
3122 str = gst_structure_get_string (s, "srtcp-cipher");
3123 gst_value_deserialize (&rtcp_cipher, str);
3124 str = gst_structure_get_string (s, "srtcp-auth");
3125 gst_value_deserialize (&rtcp_auth, str);
3126 gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL);
3128 g_object_set_property (G_OBJECT (stream->srtpenc), "rtcp-cipher",
3130 g_object_set_property (G_OBJECT (stream->srtpenc), "rtcp-auth",
3132 g_object_set (stream->srtpenc, "key", buf, NULL);
3134 g_value_unset (&rtcp_cipher);
3135 g_value_unset (&rtcp_auth);
3136 gst_buffer_unref (buf);
3139 name = g_strdup_printf ("rtcp_sink_%d", session);
3140 pad = gst_element_get_request_pad (stream->srtpenc, name);
3142 gst_object_unref (pad);
3144 return gst_object_ref (stream->srtpenc);
3148 request_aux_receiver (GstElement * rtpbin, guint sessid, GstRTSPSrc * src)
3150 GstElement *rtx, *bin;
3153 GstRTSPStream *stream;
3155 stream = find_stream (src, &sessid, (gpointer) find_stream_by_id);
3157 GST_WARNING_OBJECT (src, "Stream %u not found", sessid);
3161 GST_INFO_OBJECT (src, "creating retransmision receiver for session %u "
3162 "with map %" GST_PTR_FORMAT, sessid, stream->rtx_pt_map);
3163 bin = gst_bin_new (NULL);
3164 rtx = gst_element_factory_make ("rtprtxreceive", NULL);
3165 g_object_set (rtx, "payload-type-map", stream->rtx_pt_map, NULL);
3166 gst_bin_add (GST_BIN (bin), rtx);
3168 pad = gst_element_get_static_pad (rtx, "src");
3169 name = g_strdup_printf ("src_%u", sessid);
3170 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
3172 gst_object_unref (pad);
3174 pad = gst_element_get_static_pad (rtx, "sink");
3175 name = g_strdup_printf ("sink_%u", sessid);
3176 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
3178 gst_object_unref (pad);
3184 add_retransmission (GstRTSPSrc * src, GstRTSPTransport * transport)
3189 if (transport->trans != GST_RTSP_TRANS_RTP)
3192 signal_id = g_signal_lookup ("request-aux-receiver",
3193 G_OBJECT_TYPE (src->manager));
3194 /* there's already something connected */
3195 if (g_signal_handler_find (src->manager, G_SIGNAL_MATCH_ID, signal_id, 0,
3196 NULL, NULL, NULL) != 0) {
3197 GST_DEBUG_OBJECT (src, "Not adding RTX AUX element as "
3198 "\"request-aux-receiver\" signal is "
3199 "already used by the application");
3203 /* build the retransmission payload type map */
3204 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3205 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3208 if (stream->rtx_pt_map)
3209 gst_structure_free (stream->rtx_pt_map);
3210 stream->rtx_pt_map = gst_structure_new_empty ("application/x-rtp-pt-map");
3212 for (i = 0; i < stream->ptmap->len; i++) {
3213 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
3214 GstStructure *s = gst_caps_get_structure (item->caps, 0);
3215 const gchar *encoding;
3217 /* we only care about RTX streams */
3218 if ((encoding = gst_structure_get_string (s, "encoding-name"))
3219 && g_strcmp0 (encoding, "RTX") == 0) {
3220 const gchar *stream_pt_s;
3223 if (gst_structure_get_int (s, "payload", &rtx_pt)
3224 && (stream_pt_s = gst_structure_get_string (s, "apt"))) {
3227 gst_structure_set (stream->rtx_pt_map, stream_pt_s, G_TYPE_UINT,
3234 GST_DEBUG_OBJECT (src, "built retransmission payload map for stream "
3235 "id %i: %" GST_PTR_FORMAT, stream->id, stream->rtx_pt_map);
3238 g_object_set (src->manager, "do-retransmission", TRUE, NULL);
3240 /* enable RFC4588 retransmission handling by setting rtprtxreceive
3241 * as the "aux" element of rtpbin */
3242 g_signal_connect (src->manager, "request-aux-receiver",
3243 (GCallback) request_aux_receiver, src);
3246 /* try to get and configure a manager */
3248 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
3249 GstRTSPTransport * transport)
3251 const gchar *manager;
3253 GstStateChangeReturn ret;
3255 /* find a manager */
3256 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
3260 GST_DEBUG_OBJECT (src, "using manager %s", manager);
3262 /* configure the manager */
3263 if (src->manager == NULL) {
3264 GObjectClass *klass;
3266 if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
3268 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
3272 goto use_no_manager;
3274 if (!(src->manager = gst_element_factory_make (manager, "manager")))
3275 goto manager_failed;
3278 /* we manage this element */
3279 gst_element_set_locked_state (src->manager, TRUE);
3280 gst_bin_add (GST_BIN_CAST (src), src->manager);
3282 ret = gst_element_set_state (src->manager, GST_STATE_PAUSED);
3283 if (ret == GST_STATE_CHANGE_FAILURE)
3284 goto start_manager_failure;
3286 g_object_set (src->manager, "latency", src->latency, NULL);
3288 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
3290 if (g_object_class_find_property (klass, "ntp-sync")) {
3291 g_object_set (src->manager, "ntp-sync", src->ntp_sync, NULL);
3294 if (g_object_class_find_property (klass, "use-pipeline-clock")) {
3295 g_object_set (src->manager, "use-pipeline-clock",
3296 src->use_pipeline_clock, NULL);
3299 if (src->sdes && g_object_class_find_property (klass, "sdes")) {
3300 g_object_set (src->manager, "sdes", src->sdes, NULL);
3303 if (g_object_class_find_property (klass, "drop-on-latency")) {
3304 g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
3308 /* buffer mode pauses are handled by adding offsets to buffer times,
3309 * but some depayloaders may have a hard time syncing output times
3310 * with such input times, e.g. container ones, most notably ASF */
3311 /* TODO alternatives are having an event that indicates these shifts,
3312 * or having rtsp extensions provide suggestion on buffer mode */
3313 /* valid duration implies not likely live pipeline,
3314 * so slaving in jitterbuffer does not make much sense
3315 * (and might mess things up due to bursts) */
3316 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
3317 src->segment.duration && stream->container) {
3318 src->use_buffering = TRUE;
3320 src->use_buffering = FALSE;
3323 set_manager_buffer_mode (src);
3325 /* connect to signals */
3326 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
3328 src->manager_sig_id =
3329 g_signal_connect (src->manager, "pad-added",
3330 (GCallback) new_manager_pad, src);
3331 src->manager_ptmap_id =
3332 g_signal_connect (src->manager, "request-pt-map",
3333 (GCallback) request_pt_map, src);
3335 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
3338 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_NEW_MANAGER], 0,
3341 if (src->do_retransmission)
3342 add_retransmission (src, transport);
3344 g_signal_connect (src->manager, "request-rtp-decoder",
3345 (GCallback) request_rtp_decoder, stream);
3346 g_signal_connect (src->manager, "request-rtcp-decoder",
3347 (GCallback) request_rtp_decoder, stream);
3348 g_signal_connect (src->manager, "request-rtcp-encoder",
3349 (GCallback) request_rtcp_encoder, stream);
3351 /* we stream directly to the manager, get some pads. Each RTSP stream goes
3352 * into a separate RTP session. */
3353 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
3354 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
3356 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
3357 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
3360 /* now configure the bandwidth in the manager */
3361 if (g_signal_lookup ("get-internal-session",
3362 G_OBJECT_TYPE (src->manager)) != 0) {
3363 GObject *rtpsession;
3365 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
3368 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
3370 stream->session = rtpsession;
3372 if (stream->as_bandwidth != -1) {
3373 GST_INFO_OBJECT (src, "setting AS: %f",
3374 (gdouble) (stream->as_bandwidth * 1000));
3375 g_object_set (rtpsession, "bandwidth",
3376 (gdouble) (stream->as_bandwidth * 1000), NULL);
3378 if (stream->rr_bandwidth != -1) {
3379 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
3380 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
3383 if (stream->rs_bandwidth != -1) {
3384 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
3385 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
3389 g_object_set (rtpsession, "probation", src->probation, NULL);
3391 g_object_set (rtpsession, "internal-ssrc", stream->send_ssrc, NULL);
3393 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
3395 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
3397 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
3399 g_signal_connect (rtpsession, "on-ssrc-active",
3400 (GCallback) on_ssrc_active, stream);
3411 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3416 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
3419 start_manager_failure:
3421 GST_DEBUG_OBJECT (src, "could not start session manager");
3426 /* free the UDP sources allocated when negotiating a transport.
3427 * This function is called when the server negotiated to a transport where the
3428 * UDP sources are not needed anymore, such as TCP or multicast. */
3430 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
3434 for (i = 0; i < 2; i++) {
3435 if (stream->udpsrc[i]) {
3436 GST_DEBUG ("free UDP source %d for stream %p", i, stream);
3437 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
3438 gst_object_unref (stream->udpsrc[i]);
3439 stream->udpsrc[i] = NULL;
3444 /* for TCP, create pads to send and receive data to and from the manager and to
3445 * intercept various events and queries
3448 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
3449 GstRTSPTransport * transport, GstPad ** outpad)
3452 GstPadTemplate *template;
3453 GstPad *pad0, *pad1;
3455 /* configure for interleaved delivery, nothing needs to be done
3456 * here, the loop function will call the chain functions of the
3457 * session manager. */
3458 stream->channel[0] = transport->interleaved.min;
3459 stream->channel[1] = transport->interleaved.max;
3460 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
3461 stream->channel[0], stream->channel[1]);
3463 /* we can remove the allocated UDP ports now */
3464 gst_rtspsrc_stream_free_udp (stream);
3466 /* no session manager, send data to srcpad directly */
3467 if (!stream->channelpad[0]) {
3468 GST_DEBUG_OBJECT (src, "no manager, creating pad");
3470 /* create a new pad we will use to stream to */
3471 name = g_strdup_printf ("stream_%u", stream->id);
3472 template = gst_static_pad_template_get (&rtptemplate);
3473 stream->channelpad[0] = gst_pad_new_from_template (template, name);
3474 gst_object_unref (template);
3477 /* set caps and activate */
3478 gst_pad_use_fixed_caps (stream->channelpad[0]);
3479 gst_pad_set_active (stream->channelpad[0], TRUE);
3481 *outpad = gst_object_ref (stream->channelpad[0]);
3483 GST_DEBUG_OBJECT (src, "using manager source pad");
3485 template = gst_static_pad_template_get (&anysrctemplate);
3487 /* allocate pads for sending the channel data into the manager */
3488 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
3489 gst_pad_link_full (pad0, stream->channelpad[0], GST_PAD_LINK_CHECK_NOTHING);
3490 gst_object_unref (stream->channelpad[0]);
3491 stream->channelpad[0] = pad0;
3492 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
3493 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
3494 gst_pad_set_element_private (pad0, src);
3495 gst_pad_set_active (pad0, TRUE);
3497 if (stream->channelpad[1]) {
3498 /* if we have a sinkpad for the other channel, create a pad and link to the
3500 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
3501 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
3502 gst_pad_link_full (pad1, stream->channelpad[1],
3503 GST_PAD_LINK_CHECK_NOTHING);
3504 gst_object_unref (stream->channelpad[1]);
3505 stream->channelpad[1] = pad1;
3506 gst_pad_set_active (pad1, TRUE);
3508 gst_object_unref (template);
3510 /* setup RTCP transport back to the server if we have to. */
3511 if (src->manager && src->do_rtcp) {
3514 template = gst_static_pad_template_get (&anysinktemplate);
3516 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
3517 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
3518 gst_pad_set_element_private (stream->rtcppad, stream);
3519 gst_pad_set_active (stream->rtcppad, TRUE);
3521 /* get session RTCP pad */
3522 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3523 pad = gst_element_get_request_pad (src->manager, name);
3528 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3529 gst_object_unref (pad);
3532 gst_object_unref (template);
3538 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
3539 GstRTSPTransport * transport, const gchar ** destination, gint * min,
3540 gint * max, guint * ttl)
3542 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3544 if (!(*destination = transport->destination))
3545 *destination = stream->destination;
3548 /* transport first */
3549 *min = transport->port.min;
3550 *max = transport->port.max;
3551 if (*min == -1 && *max == -1) {
3552 /* then try from SDP */
3553 if (stream->port != 0) {
3554 *min = stream->port;
3555 *max = stream->port + 1;
3561 if (!(*ttl = transport->ttl))
3566 /* first take the source, then the endpoint to figure out where to send
3568 if (!(*destination = transport->source)) {
3569 if (src->conninfo.connection)
3570 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
3571 else if (stream->conninfo.connection)
3573 gst_rtsp_connection_get_ip (stream->conninfo.connection);
3577 /* for unicast we only expect the ports here */
3578 *min = transport->server_port.min;
3579 *max = transport->server_port.max;
3584 /* For multicast create UDP sources and join the multicast group. */
3586 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
3587 GstRTSPTransport * transport, GstPad ** outpad)
3590 const gchar *destination;
3593 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
3595 /* we can remove the allocated UDP ports now */
3596 gst_rtspsrc_stream_free_udp (stream);
3598 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
3601 /* we need a destination now */
3602 if (destination == NULL)
3603 goto no_destination;
3605 /* we really need ports now or we won't be able to receive anything at all */
3606 if (min == -1 && max == -1)
3609 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
3610 destination, min, max);
3612 /* creating UDP source for RTP */
3614 uri = g_strdup_printf ("udp://%s:%d", destination, min);
3616 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3618 if (stream->udpsrc[0] == NULL)
3621 /* take ownership */
3622 gst_object_ref_sink (stream->udpsrc[0]);
3624 if (src->udp_buffer_size != 0)
3625 g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
3626 src->udp_buffer_size, NULL);
3628 if (src->multi_iface != NULL)
3629 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3630 src->multi_iface, NULL);
3633 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3634 gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
3637 /* creating another UDP source for RTCP */
3641 uri = g_strdup_printf ("udp://%s:%d", destination, max);
3643 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3645 if (stream->udpsrc[1] == NULL)
3648 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3649 stream->profile == GST_RTSP_PROFILE_SAVPF)
3650 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3652 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3653 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3654 gst_caps_unref (caps);
3656 /* take ownership */
3657 gst_object_ref_sink (stream->udpsrc[1]);
3659 if (src->multi_iface != NULL)
3660 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3661 src->multi_iface, NULL);
3663 gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
3670 GST_DEBUG_OBJECT (src, "no UDP source element found");
3675 GST_DEBUG_OBJECT (src, "no destination found");
3680 GST_DEBUG_OBJECT (src, "no ports found");
3685 /* configure the remainder of the UDP ports */
3687 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
3688 GstRTSPTransport * transport, GstPad ** outpad)
3690 /* we manage the UDP elements now. For unicast, the UDP sources where
3691 * allocated in the stream when we suggested a transport. */
3692 if (stream->udpsrc[0]) {
3695 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3696 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
3698 GST_DEBUG_OBJECT (src, "setting up UDP source");
3700 /* configure a timeout on the UDP port. When the timeout message is
3701 * posted, we assume UDP transport is not possible. We reconnect using TCP
3703 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
3704 src->udp_timeout * 1000, NULL);
3706 if ((caps = stream_get_caps_for_pt (stream, stream->default_pt)))
3707 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
3709 /* get output pad of the UDP source. */
3710 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
3712 /* save it so we can unblock */
3713 stream->blockedpad = *outpad;
3715 /* configure pad block on the pad. As soon as there is dataflow on the
3716 * UDP source, we know that UDP is not blocked by a firewall and we can
3717 * configure all the streams to let the application autoplug decoders. */
3719 gst_pad_add_probe (stream->blockedpad,
3720 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3721 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocked, src, NULL);
3723 if (stream->channelpad[0]) {
3724 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
3725 /* configure for UDP delivery, we need to connect the UDP pads to
3726 * the session plugin. */
3727 gst_pad_link_full (*outpad, stream->channelpad[0],
3728 GST_PAD_LINK_CHECK_NOTHING);
3729 gst_object_unref (*outpad);
3731 /* we connected to pad-added signal to get pads from the manager */
3733 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
3738 if (stream->udpsrc[1]) {
3741 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
3742 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
3744 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3745 stream->profile == GST_RTSP_PROFILE_SAVPF)
3746 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3748 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3749 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3750 gst_caps_unref (caps);
3752 if (stream->channelpad[1]) {
3755 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
3757 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
3758 gst_pad_link_full (pad, stream->channelpad[1],
3759 GST_PAD_LINK_CHECK_NOTHING);
3760 gst_object_unref (pad);
3762 /* leave unlinked */
3768 /* configure the UDP sink back to the server for status reports */
3770 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
3771 GstRTSPStream * stream, GstRTSPTransport * transport)
3774 gint rtp_port, rtcp_port;
3775 gboolean do_rtp, do_rtcp;
3776 const gchar *destination;
3781 /* get transport info */
3782 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
3783 &rtp_port, &rtcp_port, &ttl);
3785 /* see what we need to do */
3786 do_rtp = (rtp_port != -1);
3787 /* it's possible that the server does not want us to send RTCP in which case
3789 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
3791 /* we need a destination when we have RTP or RTCP ports */
3792 if (destination == NULL && (do_rtp || do_rtcp))
3793 goto no_destination;
3795 /* try to construct the fakesrc to the RTP port of the server to open up any
3798 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
3801 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
3802 stream->udpsink[0] =
3803 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3805 if (stream->udpsink[0] == NULL)
3806 goto no_sink_element;
3808 /* don't join multicast group, we will have the source socket do that */
3809 /* no sync or async state changes needed */
3810 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
3811 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3813 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3815 if (stream->udpsrc[0]) {
3816 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
3817 * so that NAT firewalls will open a hole for us */
3818 g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
3819 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
3820 /* configure socket and make sure udpsink does not close it when shutting
3821 * down, it belongs to udpsrc after all. */
3822 g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
3823 "close-socket", FALSE, NULL);
3824 g_object_unref (socket);
3827 /* the source for the dummy packets to open up NAT */
3828 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
3829 if (stream->fakesrc == NULL)
3830 goto no_fakesrc_element;
3832 /* random data in 5 buffers, a size of 200 bytes should be fine */
3833 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
3834 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
3836 /* we don't want to consider this a sink */
3837 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
3839 /* keep everything locked */
3840 gst_element_set_locked_state (stream->udpsink[0], TRUE);
3841 gst_element_set_locked_state (stream->fakesrc, TRUE);
3843 gst_object_ref (stream->udpsink[0]);
3844 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
3845 gst_object_ref (stream->fakesrc);
3846 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
3848 gst_element_link_pads_full (stream->fakesrc, "src", stream->udpsink[0],
3849 "sink", GST_PAD_LINK_CHECK_NOTHING);
3852 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
3855 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
3856 stream->udpsink[1] =
3857 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3859 if (stream->udpsink[1] == NULL)
3860 goto no_sink_element;
3862 /* don't join multicast group, we will have the source socket do that */
3863 /* no sync or async state changes needed */
3864 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
3865 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3867 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3869 if (stream->udpsrc[1]) {
3870 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
3871 * because some servers check the port number of where it sends RTCP to identify
3872 * the RTCP packets it receives */
3873 g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
3874 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
3875 /* configure socket and make sure udpsink does not close it when shutting
3876 * down, it belongs to udpsrc after all. */
3877 g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
3878 "close-socket", FALSE, NULL);
3879 g_object_unref (socket);
3882 /* we don't want to consider this a sink */
3883 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
3885 /* we keep this playing always */
3886 gst_element_set_locked_state (stream->udpsink[1], TRUE);
3887 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
3889 gst_object_ref (stream->udpsink[1]);
3890 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
3892 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
3894 /* get session RTCP pad */
3895 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3896 pad = gst_element_get_request_pad (src->manager, name);
3901 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3902 gst_object_unref (pad);
3911 GST_DEBUG_OBJECT (src, "no destination address specified");
3916 GST_DEBUG_OBJECT (src, "no UDP sink element found");
3921 GST_DEBUG_OBJECT (src, "no fakesrc element found");
3926 /* sets up all elements needed for streaming over the specified transport.
3927 * Does not yet expose the element pads, this will be done when there is actuall
3928 * dataflow detected, which might never happen when UDP is blocked in a
3929 * firewall, for example.
3932 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
3933 GstRTSPTransport * transport)
3936 GstPad *outpad = NULL;
3937 GstPadTemplate *template;
3939 const gchar *media_type;
3942 src = stream->parent;
3944 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
3946 /* get the proper media type for this stream now */
3947 if (gst_rtsp_transport_get_media_type (transport, &media_type) < 0)
3948 goto unknown_transport;
3950 goto unknown_transport;
3952 /* configure the final media type */
3953 GST_DEBUG_OBJECT (src, "setting media type to %s", media_type);
3955 len = stream->ptmap->len;
3956 for (i = 0; i < len; i++) {
3958 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
3960 if (item->caps == NULL)
3963 s = gst_caps_get_structure (item->caps, 0);
3964 gst_structure_set_name (s, media_type);
3965 /* set ssrc if known */
3966 if (transport->ssrc)
3967 gst_structure_set (s, "ssrc", G_TYPE_UINT, transport->ssrc, NULL);
3970 /* try to get and configure a manager, channelpad[0-1] will be configured with
3971 * the pads for the manager, or NULL when no manager is needed. */
3972 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
3975 switch (transport->lower_transport) {
3976 case GST_RTSP_LOWER_TRANS_TCP:
3977 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
3978 goto transport_failed;
3980 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3981 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
3982 goto transport_failed;
3983 /* fallthrough, the rest is the same for UDP and MCAST */
3984 case GST_RTSP_LOWER_TRANS_UDP:
3985 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
3986 goto transport_failed;
3987 /* configure udpsinks back to the server for RTCP messages and for the
3988 * dummy RTP messages to open NAT. */
3989 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
3990 goto transport_failed;
3993 goto unknown_transport;
3997 GST_DEBUG_OBJECT (src, "creating ghostpad");
3999 gst_pad_use_fixed_caps (outpad);
4001 /* create ghostpad, don't add just yet, this will be done when we activate
4003 name = g_strdup_printf ("stream_%u", stream->id);
4004 template = gst_static_pad_template_get (&rtptemplate);
4005 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
4006 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
4007 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
4008 gst_object_unref (template);
4011 gst_object_unref (outpad);
4013 /* mark pad as ok */
4014 stream->last_ret = GST_FLOW_OK;
4021 GST_DEBUG_OBJECT (src, "failed to configure transport");
4026 GST_DEBUG_OBJECT (src, "unknown transport");
4031 GST_DEBUG_OBJECT (src, "cannot get a session manager");
4036 /* send a couple of dummy random packets on the receiver RTP port to the server,
4037 * this should make a firewall think we initiated the data transfer and
4038 * hopefully allow packets to go from the sender port to our RTP receiver port */
4040 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
4044 if (src->nat_method != GST_RTSP_NAT_DUMMY)
4047 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4048 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4050 if (stream->fakesrc && stream->udpsink[0]) {
4051 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
4052 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
4053 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
4054 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
4055 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
4061 /* Adds the source pads of all configured streams to the element.
4062 * This code is performed when we detected dataflow.
4064 * We detect dataflow from either the _loop function or with pad probes on the
4068 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
4072 GST_DEBUG_OBJECT (src, "activating streams");
4074 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4075 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4077 if (stream->udpsrc[0]) {
4078 /* remove timeout, we are streaming now and timeouts will be handled by
4079 * the session manager and jitter buffer */
4080 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
4082 if (stream->srcpad) {
4083 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
4084 gst_pad_set_active (stream->srcpad, TRUE);
4086 /* if we don't have a session manager, set the caps now. If we have a
4087 * session, we will get a notification of the pad and the caps. */
4088 if (!src->manager) {
4091 caps = stream_get_caps_for_pt (stream, stream->default_pt);
4092 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
4093 gst_pad_set_caps (stream->srcpad, caps);
4096 if (!stream->added) {
4097 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
4098 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
4099 stream->added = TRUE;
4104 /* unblock all pads */
4105 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4106 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4108 if (stream->blockid) {
4109 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
4110 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
4111 stream->blockid = 0;
4119 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
4120 gboolean reset_manager)
4123 guint64 start, stop;
4124 gdouble play_speed, play_scale;
4126 GST_DEBUG_OBJECT (src, "configuring stream caps");
4128 start = segment->position;
4129 stop = segment->duration;
4130 play_speed = segment->rate;
4131 play_scale = segment->applied_rate;
4133 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4134 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4140 len = stream->ptmap->len;
4141 for (j = 0; j < len; j++) {
4143 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, j);
4145 if (item->caps == NULL)
4148 caps = gst_caps_make_writable (item->caps);
4150 if (stream->timebase != -1)
4151 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
4152 (guint) stream->timebase, NULL);
4153 if (stream->seqbase != -1)
4154 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
4155 (guint) stream->seqbase, NULL);
4156 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
4158 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
4159 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
4160 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
4163 GST_DEBUG_OBJECT (src, "stream %p, pt %d, caps %" GST_PTR_FORMAT, stream,
4166 if (item->pt == stream->default_pt && stream->udpsrc[0]) {
4167 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
4171 if (reset_manager && src->manager) {
4172 GST_DEBUG_OBJECT (src, "clear session");
4173 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
4177 static GstFlowReturn
4178 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
4183 /* store the value */
4184 stream->last_ret = ret;
4186 /* if it's success we can return the value right away */
4187 if (ret == GST_FLOW_OK)
4190 /* any other error that is not-linked can be returned right
4192 if (ret != GST_FLOW_NOT_LINKED)
4195 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
4196 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4197 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4199 ret = ostream->last_ret;
4200 /* some other return value (must be SUCCESS but we can return
4201 * other values as well) */
4202 if (ret != GST_FLOW_NOT_LINKED)
4205 /* if we get here, all other pads were unlinked and we return
4206 * NOT_LINKED then */
4212 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
4215 gboolean res = TRUE;
4217 /* only streams that have a connection to the outside world */
4221 if (stream->udpsrc[0]) {
4222 gst_event_ref (event);
4223 res = gst_element_send_event (stream->udpsrc[0], event);
4224 } else if (stream->channelpad[0]) {
4225 gst_event_ref (event);
4226 if (GST_PAD_IS_SRC (stream->channelpad[0]))
4227 res = gst_pad_push_event (stream->channelpad[0], event);
4229 res = gst_pad_send_event (stream->channelpad[0], event);
4232 if (stream->udpsrc[1]) {
4233 gst_event_ref (event);
4234 res &= gst_element_send_event (stream->udpsrc[1], event);
4235 } else if (stream->channelpad[1]) {
4236 gst_event_ref (event);
4237 if (GST_PAD_IS_SRC (stream->channelpad[1]))
4238 res &= gst_pad_push_event (stream->channelpad[1], event);
4240 res &= gst_pad_send_event (stream->channelpad[1], event);
4244 gst_event_unref (event);
4250 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
4253 gboolean res = TRUE;
4255 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4256 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4258 gst_event_ref (event);
4259 res &= gst_rtspsrc_stream_push_event (src, ostream, event);
4261 gst_event_unref (event);
4266 static GstRTSPResult
4267 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
4272 if (info->connection == NULL) {
4273 if (info->url == NULL) {
4274 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
4275 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
4279 /* create connection */
4280 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
4281 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
4282 goto could_not_create;
4285 g_free (info->url_str);
4286 info->url_str = gst_rtsp_url_get_request_uri (info->url);
4288 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
4290 if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
4291 if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
4292 src->tls_validation_flags))
4293 GST_WARNING_OBJECT (src, "Unable to set TLS validation flags");
4295 if (src->tls_database)
4296 gst_rtsp_connection_set_tls_database (info->connection,
4300 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
4301 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
4303 if (src->proxy_host) {
4304 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
4306 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
4311 if (!info->connected) {
4314 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
4315 ("Connecting to %s", info->location));
4316 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
4318 gst_rtsp_connection_connect (info->connection,
4319 src->ptcp_timeout)) < 0)
4320 goto could_not_connect;
4322 info->connected = TRUE;
4329 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
4334 gchar *str = gst_rtsp_strresult (res);
4335 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
4341 gchar *str = gst_rtsp_strresult (res);
4342 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
4348 static GstRTSPResult
4349 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
4352 GST_RTSP_STATE_LOCK (src);
4353 if (info->connected) {
4354 GST_DEBUG_OBJECT (src, "closing connection...");
4355 gst_rtsp_connection_close (info->connection);
4356 info->connected = FALSE;
4358 if (free && info->connection) {
4359 /* free connection */
4360 GST_DEBUG_OBJECT (src, "freeing connection...");
4361 gst_rtsp_connection_free (info->connection);
4362 info->connection = NULL;
4364 GST_RTSP_STATE_UNLOCK (src);
4368 static GstRTSPResult
4369 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
4374 GST_DEBUG_OBJECT (src, "reconnecting connection...");
4375 gst_rtsp_conninfo_close (src, info, FALSE);
4376 res = gst_rtsp_conninfo_connect (src, info, async);
4382 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
4386 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
4387 GST_RTSP_STATE_LOCK (src);
4388 if (src->conninfo.connection && src->conninfo.flushing != flush) {
4389 GST_DEBUG_OBJECT (src, "connection flush");
4390 gst_rtsp_connection_flush (src->conninfo.connection, flush);
4391 src->conninfo.flushing = flush;
4393 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4394 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4395 if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
4396 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
4397 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
4398 stream->conninfo.flushing = flush;
4401 GST_RTSP_STATE_UNLOCK (src);
4404 /* FIXME, handle server request, reply with OK, for now */
4405 static GstRTSPResult
4406 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
4407 GstRTSPMessage * request)
4409 GstRTSPMessage response = { 0 };
4412 GST_DEBUG_OBJECT (src, "got server request message");
4415 gst_rtsp_message_dump (request);
4417 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
4419 if (res == GST_RTSP_ENOTIMPL) {
4420 /* default implementation, send OK */
4421 GST_DEBUG_OBJECT (src, "prepare OK reply");
4423 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
4428 /* let app parse and reply */
4429 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST],
4430 0, request, &response);
4433 gst_rtsp_message_dump (&response);
4435 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
4439 gst_rtsp_message_unset (&response);
4440 } else if (res == GST_RTSP_EEOF)
4448 gst_rtsp_message_unset (&response);
4453 /* send server keep-alive */
4454 static GstRTSPResult
4455 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
4457 GstRTSPMessage request = { 0 };
4459 GstRTSPMethod method;
4460 const gchar *control;
4462 if (src->do_rtsp_keep_alive == FALSE) {
4463 GST_DEBUG_OBJECT (src, "do-rtsp-keep-alive is FALSE, not sending.");
4464 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4468 GST_DEBUG_OBJECT (src, "creating server keep-alive");
4470 /* find a method to use for keep-alive */
4471 if (src->methods & GST_RTSP_GET_PARAMETER)
4472 method = GST_RTSP_GET_PARAMETER;
4474 method = GST_RTSP_OPTIONS;
4476 control = get_aggregate_control (src);
4477 if (control == NULL)
4480 res = gst_rtsp_message_init_request (&request, method, control);
4485 gst_rtsp_message_dump (&request);
4488 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
4493 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4494 gst_rtsp_message_unset (&request);
4501 GST_WARNING_OBJECT (src, "no control url to send keepalive");
4506 gchar *str = gst_rtsp_strresult (res);
4508 gst_rtsp_message_unset (&request);
4509 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
4510 ("Could not send keep-alive. (%s)", str));
4516 static GstFlowReturn
4517 gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
4519 GstFlowReturn ret = GST_FLOW_OK;
4521 GstRTSPStream *stream;
4522 GstPad *outpad = NULL;
4529 channel = message->type_data.data.channel;
4531 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
4533 goto unknown_stream;
4535 if (channel == stream->channel[0]) {
4536 outpad = stream->channelpad[0];
4538 } else if (channel == stream->channel[1]) {
4539 outpad = stream->channelpad[1];
4545 /* take a look at the body to figure out what we have */
4546 gst_rtsp_message_get_body (message, &data, &size);
4548 goto invalid_length;
4550 /* channels are not correct on some servers, do extra check */
4551 if (data[1] >= 200 && data[1] <= 204) {
4552 /* hmm RTCP message switch to the RTCP pad of the same stream. */
4553 outpad = stream->channelpad[1];
4557 /* we have no clue what this is, just ignore then. */
4559 goto unknown_stream;
4561 /* take the message body for further processing */
4562 gst_rtsp_message_steal_body (message, &data, &size);
4564 /* strip the trailing \0 */
4567 buf = gst_buffer_new ();
4568 gst_buffer_append_memory (buf,
4569 gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
4571 /* don't need message anymore */
4572 gst_rtsp_message_unset (message);
4574 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
4577 if (src->need_activate) {
4583 guint group_id = gst_util_group_id_next ();
4586 /* generate an SHA256 sum of the URI */
4587 cs = g_checksum_new (G_CHECKSUM_SHA256);
4588 uri = src->conninfo.location;
4589 g_checksum_update (cs, (const guchar *) uri, strlen (uri));
4591 gst_segment_init (&segment, GST_FORMAT_TIME);
4593 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4594 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4598 g_strdup_printf ("%s/%d", g_checksum_get_string (cs), ostream->id);
4599 event = gst_event_new_stream_start (stream_id);
4600 gst_event_set_group_id (event, group_id);
4603 gst_rtspsrc_stream_push_event (src, ostream, event);
4605 if ((caps = stream_get_caps_for_pt (ostream, ostream->default_pt))) {
4606 /* only streams that have a connection to the outside world */
4607 if (ostream->setup) {
4608 if (ostream->udpsrc[0]) {
4609 gst_element_send_event (ostream->udpsrc[0],
4610 gst_event_new_caps (caps));
4611 } else if (ostream->channelpad[0]) {
4612 if (GST_PAD_IS_SRC (ostream->channelpad[0]))
4613 gst_pad_push_event (ostream->channelpad[0],
4614 gst_event_new_caps (caps));
4616 gst_pad_send_event (ostream->channelpad[0],
4617 gst_event_new_caps (caps));
4620 caps = gst_caps_new_empty_simple ("application/x-rtcp");
4622 if (ostream->udpsrc[1]) {
4623 gst_element_send_event (ostream->udpsrc[1],
4624 gst_event_new_caps (caps));
4625 } else if (ostream->channelpad[1]) {
4626 if (GST_PAD_IS_SRC (ostream->channelpad[1]))
4627 gst_pad_push_event (ostream->channelpad[1],
4628 gst_event_new_caps (caps));
4630 gst_pad_send_event (ostream->channelpad[1],
4631 gst_event_new_caps (caps));
4636 /* Push a SEGMENT event if we don't have one pending, if we have one
4637 * pending we will just send that one a few lines below to all pads
4639 if (!src->start_segment)
4640 gst_rtspsrc_stream_push_event (src, ostream,
4641 gst_event_new_segment (&segment));
4643 g_checksum_free (cs);
4645 gst_rtspsrc_activate_streams (src);
4646 src->need_activate = FALSE;
4649 if ((event = src->start_segment) != NULL) {
4650 src->start_segment = NULL;
4651 gst_rtspsrc_push_event (src, event);
4654 if (src->base_time == -1) {
4655 /* Take current running_time. This timestamp will be put on
4656 * the first buffer of each stream because we are a live source and so we
4657 * timestamp with the running_time. When we are dealing with TCP, we also
4658 * only timestamp the first buffer (using the DISCONT flag) because a server
4659 * typically bursts data, for which we don't want to compensate by speeding
4660 * up the media. The other timestamps will be interpollated from this one
4661 * using the RTP timestamps. */
4662 GST_OBJECT_LOCK (src);
4663 if (GST_ELEMENT_CLOCK (src)) {
4665 GstClockTime base_time;
4667 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
4668 base_time = GST_ELEMENT_CAST (src)->base_time;
4670 src->base_time = now - base_time;
4672 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
4673 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
4675 GST_OBJECT_UNLOCK (src);
4678 if (stream->discont && !is_rtcp) {
4679 /* mark first RTP buffer as discont */
4680 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
4681 stream->discont = FALSE;
4682 /* first buffer gets the timestamp, other buffers are not timestamped and
4683 * their presentation time will be interpollated from the rtp timestamps. */
4684 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
4685 GST_TIME_ARGS (src->base_time));
4687 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
4690 /* chain to the peer pad */
4691 if (GST_PAD_IS_SINK (outpad))
4692 ret = gst_pad_chain (outpad, buf);
4694 ret = gst_pad_push (outpad, buf);
4697 /* combine all stream flows for the data transport */
4698 ret = gst_rtspsrc_combine_flows (src, stream, ret);
4705 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
4706 gst_rtsp_message_unset (message);
4711 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4712 ("Short message received, ignoring."));
4713 gst_rtsp_message_unset (message);
4718 static GstFlowReturn
4719 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
4721 GstRTSPMessage message = { 0 };
4723 GstFlowReturn ret = GST_FLOW_OK;
4724 GTimeVal tv_timeout;
4727 /* get the next timeout interval */
4728 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4730 /* see if the timeout period expired */
4731 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
4732 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
4733 /* send keep-alive, only act on interrupt, a warning will be posted for
4735 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4737 /* get new timeout */
4738 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4741 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
4742 tv_timeout.tv_sec, tv_timeout.tv_usec);
4744 /* protect the connection with the connection lock so that we can see when
4745 * we are finished doing server communication */
4747 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4748 &message, src->ptcp_timeout);
4752 GST_DEBUG_OBJECT (src, "we received a server message");
4754 case GST_RTSP_EINTR:
4755 /* we got interrupted this means we need to stop */
4757 case GST_RTSP_ETIMEOUT:
4758 /* no reply, send keep alive */
4759 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4760 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4764 /* go EOS when the server closed the connection */
4770 switch (message.type) {
4771 case GST_RTSP_MESSAGE_REQUEST:
4772 /* server sends us a request message, handle it */
4774 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4776 if (res == GST_RTSP_EEOF)
4779 goto handle_request_failed;
4781 case GST_RTSP_MESSAGE_RESPONSE:
4782 /* we ignore response messages */
4783 GST_DEBUG_OBJECT (src, "ignoring response message");
4785 gst_rtsp_message_dump (&message);
4787 case GST_RTSP_MESSAGE_DATA:
4788 GST_DEBUG_OBJECT (src, "got data message");
4789 ret = gst_rtspsrc_handle_data (src, &message);
4790 if (ret != GST_FLOW_OK)
4791 goto handle_data_failed;
4794 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4799 g_assert_not_reached ();
4804 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4805 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4806 ("The server closed the connection."));
4807 src->conninfo.connected = FALSE;
4808 gst_rtsp_message_unset (&message);
4809 return GST_FLOW_EOS;
4813 gst_rtsp_message_unset (&message);
4814 GST_DEBUG_OBJECT (src, "got interrupted");
4815 return GST_FLOW_FLUSHING;
4819 gchar *str = gst_rtsp_strresult (res);
4821 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4822 ("Could not receive message. (%s)", str));
4825 gst_rtsp_message_unset (&message);
4826 return GST_FLOW_ERROR;
4828 handle_request_failed:
4830 gchar *str = gst_rtsp_strresult (res);
4832 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4833 ("Could not handle server message. (%s)", str));
4835 gst_rtsp_message_unset (&message);
4836 return GST_FLOW_ERROR;
4840 GST_DEBUG_OBJECT (src, "could no handle data message");
4845 static GstFlowReturn
4846 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
4849 GstRTSPMessage message = { 0 };
4853 GTimeVal tv_timeout;
4855 /* get the next timeout interval */
4856 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4858 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
4859 (gint) tv_timeout.tv_sec);
4861 gst_rtsp_message_unset (&message);
4863 /* we should continue reading the TCP socket because the server might
4864 * send us requests. When the session timeout expires, we need to send a
4865 * keep-alive request to keep the session open. */
4866 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4867 &message, &tv_timeout);
4871 GST_DEBUG_OBJECT (src, "we received a server message");
4873 case GST_RTSP_EINTR:
4874 /* we got interrupted, see what we have to do */
4876 case GST_RTSP_ETIMEOUT:
4877 /* send keep-alive, ignore the result, a warning will be posted. */
4878 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4879 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4883 /* server closed the connection. not very fatal for UDP, reconnect and
4884 * see what happens. */
4885 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4886 ("The server closed the connection."));
4887 if (src->udp_reconnect) {
4889 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
4896 GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
4898 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4899 ("Unhandled return value %d.", res));
4903 switch (message.type) {
4904 case GST_RTSP_MESSAGE_REQUEST:
4905 /* server sends us a request message, handle it */
4907 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4909 if (res == GST_RTSP_EEOF)
4912 goto handle_request_failed;
4914 case GST_RTSP_MESSAGE_RESPONSE:
4915 /* we ignore response and data messages */
4916 GST_DEBUG_OBJECT (src, "ignoring response message");
4918 gst_rtsp_message_dump (&message);
4919 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
4920 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
4921 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
4922 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
4923 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4930 case GST_RTSP_MESSAGE_DATA:
4931 /* we ignore response and data messages */
4932 GST_DEBUG_OBJECT (src, "ignoring data message");
4935 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4940 g_assert_not_reached ();
4942 /* we get here when the connection got interrupted */
4945 gst_rtsp_message_unset (&message);
4946 GST_DEBUG_OBJECT (src, "got interrupted");
4947 return GST_FLOW_FLUSHING;
4951 gchar *str = gst_rtsp_strresult (res);
4954 src->conninfo.connected = FALSE;
4955 if (res != GST_RTSP_EINTR) {
4956 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
4957 ("Could not connect to server. (%s)", str));
4959 ret = GST_FLOW_ERROR;
4961 ret = GST_FLOW_FLUSHING;
4967 gchar *str = gst_rtsp_strresult (res);
4969 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4970 ("Could not receive message. (%s)", str));
4972 return GST_FLOW_ERROR;
4974 handle_request_failed:
4976 gchar *str = gst_rtsp_strresult (res);
4979 gst_rtsp_message_unset (&message);
4980 if (res != GST_RTSP_EINTR) {
4981 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4982 ("Could not handle server message. (%s)", str));
4984 ret = GST_FLOW_ERROR;
4986 ret = GST_FLOW_FLUSHING;
4992 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4993 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4994 ("The server closed the connection."));
4995 src->conninfo.connected = FALSE;
4996 gst_rtsp_message_unset (&message);
4997 return GST_FLOW_EOS;
5001 static GstRTSPResult
5002 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
5004 GstRTSPResult res = GST_RTSP_OK;
5007 GST_DEBUG_OBJECT (src, "doing reconnect");
5009 GST_OBJECT_LOCK (src);
5010 /* only restart when the pads were not yet activated, else we were
5011 * streaming over UDP */
5012 restart = src->need_activate;
5013 GST_OBJECT_UNLOCK (src);
5015 /* no need to restart, we're done */
5019 /* we can try only TCP now */
5020 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
5022 /* close and cleanup our state */
5023 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
5026 /* see if we have TCP left to try. Also don't try TCP when we were configured
5028 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
5031 /* We post a warning message now to inform the user
5032 * that nothing happened. It's most likely a firewall thing. */
5033 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5034 ("Could not receive any UDP packets for %.4f seconds, maybe your "
5035 "firewall is blocking it. Retrying using a TCP connection.",
5036 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
5038 /* open new connection using tcp */
5039 if (gst_rtspsrc_open (src, async) < 0)
5042 /* start playback */
5043 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
5052 src->cur_protocols = 0;
5053 /* no transport possible, post an error and stop */
5054 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5055 ("Could not receive any UDP packets for %.4f seconds, maybe your "
5056 "firewall is blocking it. No other protocols to try.",
5057 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
5058 return GST_RTSP_ERROR;
5062 GST_DEBUG_OBJECT (src, "open failed");
5067 GST_DEBUG_OBJECT (src, "play failed");
5073 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
5077 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
5080 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
5083 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
5086 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
5094 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
5098 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
5101 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
5104 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
5107 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
5115 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
5119 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
5122 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
5125 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
5128 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
5136 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
5140 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
5143 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
5146 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
5149 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
5157 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
5159 if (ret == GST_RTSP_OK)
5160 gst_rtspsrc_loop_complete_cmd (src, cmd);
5161 else if (ret == GST_RTSP_EINTR)
5162 gst_rtspsrc_loop_cancel_cmd (src, cmd);
5164 gst_rtspsrc_loop_error_cmd (src, cmd);
5168 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
5171 gboolean flushed = FALSE;
5173 /* start new request */
5174 gst_rtspsrc_loop_start_cmd (src, cmd);
5176 GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
5178 GST_OBJECT_LOCK (src);
5179 old = src->pending_cmd;
5180 if (old == CMD_RECONNECT) {
5181 GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
5182 cmd = CMD_RECONNECT;
5184 if (old != CMD_WAIT) {
5185 src->pending_cmd = CMD_WAIT;
5186 GST_OBJECT_UNLOCK (src);
5187 /* cancel previous request */
5188 GST_DEBUG_OBJECT (src, "cancel previous request %d", old);
5189 gst_rtspsrc_loop_cancel_cmd (src, old);
5190 GST_OBJECT_LOCK (src);
5192 src->pending_cmd = cmd;
5193 /* interrupt if allowed */
5194 if (src->busy_cmd & mask) {
5195 GST_DEBUG_OBJECT (src, "connection flush busy %d", src->busy_cmd);
5196 gst_rtspsrc_connection_flush (src, TRUE);
5199 GST_DEBUG_OBJECT (src, "not interrupting busy cmd %d", src->busy_cmd);
5202 gst_task_start (src->task);
5203 GST_OBJECT_UNLOCK (src);
5209 gst_rtspsrc_loop (GstRTSPSrc * src)
5213 if (!src->conninfo.connection || !src->conninfo.connected)
5216 if (src->interleaved)
5217 ret = gst_rtspsrc_loop_interleaved (src);
5219 ret = gst_rtspsrc_loop_udp (src);
5221 if (ret != GST_FLOW_OK)
5229 GST_WARNING_OBJECT (src, "we are not connected");
5230 ret = GST_FLOW_FLUSHING;
5235 const gchar *reason = gst_flow_get_name (ret);
5237 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
5238 src->running = FALSE;
5239 if (ret == GST_FLOW_EOS) {
5240 /* perform EOS logic */
5241 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
5242 gst_element_post_message (GST_ELEMENT_CAST (src),
5243 gst_message_new_segment_done (GST_OBJECT_CAST (src),
5244 src->segment.format, src->segment.position));
5245 gst_rtspsrc_push_event (src,
5246 gst_event_new_segment_done (src->segment.format,
5247 src->segment.position));
5249 gst_rtspsrc_push_event (src, gst_event_new_eos ());
5251 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
5252 /* for fatal errors we post an error message, post the error before the
5253 * EOS so the app knows about the error first. */
5254 GST_ELEMENT_ERROR (src, STREAM, FAILED,
5255 ("Internal data flow error."),
5256 ("streaming task paused, reason %s (%d)", reason, ret));
5257 gst_rtspsrc_push_event (src, gst_event_new_eos ());
5259 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_LOOP);
5264 #ifndef GST_DISABLE_GST_DEBUG
5265 static const gchar *
5266 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
5270 while (method != 0) {
5287 static const gchar *
5288 gst_rtspsrc_skip_lws (const gchar * s)
5290 while (g_ascii_isspace (*s))
5295 static const gchar *
5296 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
5298 while (s > start && g_ascii_isspace (*(s - 1)))
5303 static const gchar *
5304 gst_rtspsrc_skip_commas (const gchar * s)
5306 /* The grammar allows for multiple commas */
5307 while (g_ascii_isspace (*s) || *s == ',')
5312 static const gchar *
5313 gst_rtspsrc_skip_item (const gchar * s)
5315 gboolean quoted = FALSE;
5316 const gchar *start = s;
5318 /* A list item ends at the last non-whitespace character
5319 * before a comma which is not inside a quoted-string. Or at
5320 * the end of the string.
5326 if (*s == '\\' && *(s + 1))
5335 return gst_rtspsrc_unskip_lws (s, start);
5339 gst_rtsp_decode_quoted_string (gchar * quoted_string)
5343 src = quoted_string + 1;
5344 dst = quoted_string;
5345 while (*src && *src != '"') {
5346 if (*src == '\\' && *(src + 1))
5353 /* Extract the authentication tokens that the server provided for each method
5354 * into an array of structures and give those to the connection object.
5357 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
5358 const gchar * header, gboolean * stale)
5360 GSList *list = NULL, *iter;
5362 gchar *item, *eq, *name_end, *value;
5364 g_return_if_fail (stale != NULL);
5366 gst_rtsp_connection_clear_auth_params (conn);
5369 /* Parse a header whose content is described by RFC2616 as
5370 * "#something", where "something" does not itself contain commas,
5371 * except as part of quoted-strings, into a list of allocated strings.
5373 header = gst_rtspsrc_skip_commas (header);
5375 end = gst_rtspsrc_skip_item (header);
5376 list = g_slist_prepend (list, g_strndup (header, end - header));
5377 header = gst_rtspsrc_skip_commas (end);
5382 list = g_slist_reverse (list);
5383 for (iter = list; iter; iter = iter->next) {
5386 eq = strchr (item, '=');
5388 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
5389 if (name_end == item) {
5390 /* That's no good... */
5397 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
5399 gst_rtsp_decode_quoted_string (value);
5403 if (value && strcmp (item, "stale") == 0 && strcmp (value, "TRUE") == 0)
5405 gst_rtsp_connection_set_auth_param (conn, item, value);
5409 g_slist_free (list);
5412 /* Parse a WWW-Authenticate Response header and determine the
5413 * available authentication methods
5415 * This code should also cope with the fact that each WWW-Authenticate
5416 * header can contain multiple challenge methods + tokens
5418 * At the moment, for Basic auth, we just do a minimal check and don't
5419 * even parse out the realm */
5421 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
5422 GstRTSPConnection * conn, gboolean * stale)
5426 g_return_if_fail (hdr != NULL);
5427 g_return_if_fail (methods != NULL);
5428 g_return_if_fail (stale != NULL);
5430 /* Skip whitespace at the start of the string */
5431 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
5433 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
5434 *methods |= GST_RTSP_AUTH_BASIC;
5435 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
5436 *methods |= GST_RTSP_AUTH_DIGEST;
5437 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
5442 * gst_rtspsrc_setup_auth:
5443 * @src: the rtsp source
5445 * Configure a username and password and auth method on the
5446 * connection object based on a response we received from the
5449 * Currently, this requires that a username and password were supplied
5450 * in the uri. In the future, they may be requested on demand by sending
5451 * a message up the bus.
5453 * Returns: TRUE if authentication information could be set up correctly.
5456 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
5460 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
5461 GstRTSPAuthMethod method;
5462 GstRTSPResult auth_result;
5464 GstRTSPConnection *conn;
5466 gboolean stale = FALSE;
5468 conn = src->conninfo.connection;
5470 /* Identify the available auth methods and see if any are supported */
5471 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
5472 &hdr, 0) == GST_RTSP_OK) {
5473 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
5476 if (avail_methods == GST_RTSP_AUTH_NONE)
5477 goto no_auth_available;
5479 /* For digest auth, if the response indicates that the session
5480 * data are stale, we just update them in the connection object and
5481 * return TRUE to retry the request */
5483 src->tried_url_auth = FALSE;
5485 url = gst_rtsp_connection_get_url (conn);
5487 /* Do we have username and password available? */
5488 if (url != NULL && !src->tried_url_auth && url->user != NULL
5489 && url->passwd != NULL) {
5492 src->tried_url_auth = TRUE;
5493 GST_DEBUG_OBJECT (src,
5494 "Attempting authentication using credentials from the URL");
5496 user = src->user_id;
5497 pass = src->user_pw;
5498 GST_DEBUG_OBJECT (src,
5499 "Attempting authentication using credentials from the properties");
5502 /* FIXME: If the url didn't contain username and password or we tried them
5503 * already, request a username and passwd from the application via some kind
5504 * of credentials request message */
5506 /* If we don't have a username and passwd at this point, bail out. */
5507 if (user == NULL || pass == NULL)
5510 /* Try to configure for each available authentication method, strongest to
5512 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
5513 /* Check if this method is available on the server */
5514 if ((method & avail_methods) == 0)
5517 /* Pass the credentials to the connection to try on the next request */
5518 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
5519 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
5520 * ignore it and end up retrying later */
5521 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
5522 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
5523 gst_rtsp_auth_method_to_string (method));
5528 if (method == GST_RTSP_AUTH_NONE)
5529 goto no_auth_available;
5535 /* Output an error indicating that we couldn't connect because there were
5536 * no supported authentication protocols */
5537 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5538 ("No supported authentication protocol was found"));
5543 /* We don't fire an error message, we just return FALSE and let the
5544 * normal NOT_AUTHORIZED error be propagated */
5549 static GstRTSPResult
5550 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5551 GstRTSPMessage * request, GstRTSPMessage * response,
5552 GstRTSPStatusCode * code)
5555 GstRTSPStatusCode thecode;
5556 gchar *content_base = NULL;
5560 if (!src->short_header)
5561 gst_rtsp_ext_list_before_send (src->extensions, request);
5563 GST_DEBUG_OBJECT (src, "sending message");
5566 gst_rtsp_message_dump (request);
5568 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
5572 gst_rtsp_connection_reset_timeout (conn);
5575 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
5580 gst_rtsp_message_dump (response);
5582 switch (response->type) {
5583 case GST_RTSP_MESSAGE_REQUEST:
5584 res = gst_rtspsrc_handle_request (src, conn, response);
5585 if (res == GST_RTSP_EEOF)
5588 goto handle_request_failed;
5590 case GST_RTSP_MESSAGE_RESPONSE:
5591 /* ok, a response is good */
5592 GST_DEBUG_OBJECT (src, "received response message");
5594 case GST_RTSP_MESSAGE_DATA:
5595 /* get next response */
5596 GST_DEBUG_OBJECT (src, "handle data response message");
5597 gst_rtspsrc_handle_data (src, response);
5600 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
5605 thecode = response->type_data.response.code;
5607 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
5609 /* if the caller wanted the result code, we store it. */
5613 /* If the request didn't succeed, bail out before doing any more */
5614 if (thecode != GST_RTSP_STS_OK)
5617 /* store new content base if any */
5618 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
5621 g_free (src->content_base);
5622 src->content_base = g_strdup (content_base);
5624 gst_rtsp_ext_list_after_send (src->extensions, request, response);
5631 gchar *str = gst_rtsp_strresult (res);
5633 if (res != GST_RTSP_EINTR) {
5634 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5635 ("Could not send message. (%s)", str));
5637 GST_WARNING_OBJECT (src, "send interrupted");
5646 GST_WARNING_OBJECT (src, "server closed connection");
5647 if ((try == 0) && !src->interleaved && src->udp_reconnect) {
5649 /* if reconnect succeeds, try again */
5651 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
5655 /* only try once after reconnect, then fallthrough and error out */
5658 gchar *str = gst_rtsp_strresult (res);
5660 if (res != GST_RTSP_EINTR) {
5661 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5662 ("Could not receive message. (%s)", str));
5664 GST_WARNING_OBJECT (src, "receive interrupted");
5672 handle_request_failed:
5674 /* ERROR was posted */
5675 gst_rtsp_message_unset (response);
5680 GST_DEBUG_OBJECT (src, "we got an eof from the server");
5681 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5682 ("The server closed the connection."));
5683 gst_rtsp_message_unset (response);
5690 * @src: the rtsp source
5691 * @conn: the connection to send on
5692 * @request: must point to a valid request
5693 * @response: must point to an empty #GstRTSPMessage
5694 * @code: an optional code result
5696 * send @request and retrieve the response in @response. optionally @code can be
5697 * non-NULL in which case it will contain the status code of the response.
5699 * If This function returns #GST_RTSP_OK, @response will contain a valid response
5700 * message that should be cleaned with gst_rtsp_message_unset() after usage.
5702 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
5703 * @response message) if the response code was not 200 (OK).
5705 * If the attempt results in an authentication failure, then this will attempt
5706 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
5709 * Returns: #GST_RTSP_OK if the processing was successful.
5711 static GstRTSPResult
5712 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5713 GstRTSPMessage * request, GstRTSPMessage * response,
5714 GstRTSPStatusCode * code)
5716 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
5717 GstRTSPResult res = GST_RTSP_ERROR;
5720 GstRTSPMethod method = GST_RTSP_INVALID;
5726 /* make sure we don't loop forever */
5730 /* save method so we can disable it when the server complains */
5731 method = request->type_data.request.method;
5734 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
5738 case GST_RTSP_STS_UNAUTHORIZED:
5739 if (gst_rtspsrc_setup_auth (src, response)) {
5740 /* Try the request/response again after configuring the auth info
5748 } while (retry == TRUE);
5750 /* If the user requested the code, let them handle errors, otherwise
5751 * post an error below */
5754 else if (int_code != GST_RTSP_STS_OK)
5755 goto error_response;
5762 GST_DEBUG_OBJECT (src, "got error %d", res);
5767 res = GST_RTSP_ERROR;
5769 switch (response->type_data.response.code) {
5770 case GST_RTSP_STS_NOT_FOUND:
5771 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
5772 response->type_data.response.reason));
5774 case GST_RTSP_STS_UNAUTHORIZED:
5775 GST_ELEMENT_ERROR (src, RESOURCE, NOT_AUTHORIZED, (NULL), ("%s",
5776 response->type_data.response.reason));
5778 case GST_RTSP_STS_MOVED_PERMANENTLY:
5779 case GST_RTSP_STS_MOVE_TEMPORARILY:
5781 gchar *new_location;
5782 GstRTSPLowerTrans transports;
5784 GST_DEBUG_OBJECT (src, "got redirection");
5785 /* if we don't have a Location Header, we must error */
5786 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
5787 &new_location, 0) < 0)
5790 /* When we receive a redirect result, we go back to the INIT state after
5791 * parsing the new URI. The caller should do the needed steps to issue
5792 * a new setup when it detects this state change. */
5793 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
5795 /* save current transports */
5796 if (src->conninfo.url)
5797 transports = src->conninfo.url->transports;
5799 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
5801 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
5803 /* set old transports */
5804 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
5805 src->conninfo.url->transports = transports;
5807 src->need_redirect = TRUE;
5808 src->state = GST_RTSP_STATE_INIT;
5812 case GST_RTSP_STS_NOT_ACCEPTABLE:
5813 case GST_RTSP_STS_NOT_IMPLEMENTED:
5814 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
5815 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
5816 gst_rtsp_method_as_text (method));
5817 src->methods &= ~method;
5821 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5822 ("Got error response: %d (%s).", response->type_data.response.code,
5823 response->type_data.response.reason));
5826 /* if we return ERROR we should unset the response ourselves */
5827 if (res == GST_RTSP_ERROR)
5828 gst_rtsp_message_unset (response);
5834 static GstRTSPResult
5835 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
5836 GstRTSPMessage * response, GstRTSPSrc * src)
5838 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
5843 /* parse the response and collect all the supported methods. We need this
5844 * information so that we don't try to send an unsupported request to the
5848 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
5850 GstRTSPHeaderField field;
5854 /* reset supported methods */
5857 /* Try Allow Header first */
5858 field = GST_RTSP_HDR_ALLOW;
5861 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5862 if (indx == 0 && !respoptions) {
5863 /* if no Allow header was found then try the Public header... */
5864 field = GST_RTSP_HDR_PUBLIC;
5865 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5870 src->methods |= gst_rtsp_options_from_text (respoptions);
5875 if (src->methods == 0) {
5876 /* neither Allow nor Public are required, assume the server supports
5877 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
5879 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
5880 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
5882 /* always assume PLAY, FIXME, extensions should be able to override
5884 src->methods |= GST_RTSP_PLAY;
5885 /* also assume it will support Range */
5886 src->seekable = TRUE;
5888 /* we need describe and setup */
5889 if (!(src->methods & GST_RTSP_DESCRIBE))
5891 if (!(src->methods & GST_RTSP_SETUP))
5899 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5900 ("Server does not support DESCRIBE."));
5905 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5906 ("Server does not support SETUP."));
5911 /* masks to be kept in sync with the hardcoded protocol order of preference
5913 static const guint protocol_masks[] = {
5914 GST_RTSP_LOWER_TRANS_UDP,
5915 GST_RTSP_LOWER_TRANS_UDP_MCAST,
5916 GST_RTSP_LOWER_TRANS_TCP,
5920 static GstRTSPResult
5921 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
5922 GstRTSPLowerTrans protocols, GstRTSPProfile profile, gchar ** transports)
5926 gboolean add_udp_str;
5931 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
5936 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
5938 /* extension listed transports, use those */
5939 if (*transports != NULL)
5942 /* it's the default */
5943 add_udp_str = FALSE;
5945 /* the default RTSP transports */
5946 result = g_string_new ("RTP");
5949 case GST_RTSP_PROFILE_AVP:
5950 g_string_append (result, "/AVP");
5952 case GST_RTSP_PROFILE_SAVP:
5953 g_string_append (result, "/SAVP");
5955 case GST_RTSP_PROFILE_AVPF:
5956 g_string_append (result, "/AVPF");
5958 case GST_RTSP_PROFILE_SAVPF:
5959 g_string_append (result, "/SAVPF");
5965 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
5966 GST_DEBUG_OBJECT (src, "adding UDP unicast");
5968 g_string_append (result, "/UDP");
5969 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
5970 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
5971 GST_DEBUG_OBJECT (src, "adding UDP multicast");
5972 /* we don't have to allocate any UDP ports yet, if the selected transport
5973 * turns out to be multicast we can create them and join the multicast
5974 * group indicated in the transport reply */
5976 g_string_append (result, "/UDP");
5977 g_string_append (result, ";multicast");
5978 if (src->next_port_num != 0) {
5979 if (src->client_port_range.max > 0 &&
5980 src->next_port_num >= src->client_port_range.max)
5983 g_string_append_printf (result, ";client_port=%d-%d",
5984 src->next_port_num, src->next_port_num + 1);
5986 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
5987 GST_DEBUG_OBJECT (src, "adding TCP");
5989 g_string_append (result, "/TCP;unicast;interleaved=%%i1-%%i2");
5991 *transports = g_string_free (result, FALSE);
5993 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
6000 GST_ERROR ("extension gave error %d", res);
6005 GST_ERROR ("no more ports available");
6006 return GST_RTSP_ERROR;
6010 static GstRTSPResult
6011 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
6012 gint orig_rtpport, gint orig_rtcpport)
6015 gint nr_udp, nr_int;
6017 gint rtpport = 0, rtcpport = 0;
6020 src = stream->parent;
6022 /* find number of placeholders first */
6023 if (strstr (*transports, "%%i2"))
6025 else if (strstr (*transports, "%%i1"))
6030 if (strstr (*transports, "%%u2"))
6032 else if (strstr (*transports, "%%u1"))
6037 if (nr_udp == 0 && nr_int == 0)
6041 if (!orig_rtpport || !orig_rtcpport) {
6042 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
6045 rtpport = orig_rtpport;
6046 rtcpport = orig_rtcpport;
6050 str = g_string_new ("");
6052 while ((next = strstr (p, "%%"))) {
6053 g_string_append_len (str, p, next - p);
6054 if (next[2] == 'u') {
6056 g_string_append_printf (str, "%d", rtpport);
6057 else if (next[3] == '2')
6058 g_string_append_printf (str, "%d", rtcpport);
6060 if (next[2] == 'i') {
6062 g_string_append_printf (str, "%d", src->free_channel);
6063 else if (next[3] == '2')
6064 g_string_append_printf (str, "%d", src->free_channel + 1);
6069 /* append final part */
6070 g_string_append (str, p);
6072 g_free (*transports);
6073 *transports = g_string_free (str, FALSE);
6081 GST_ERROR ("failed to allocate udp ports");
6082 return GST_RTSP_ERROR;
6087 enc_key_length_from_cipher_name (const gchar * cipher)
6089 if (g_strcmp0 (cipher, "aes-128-icm") == 0)
6090 return AES_128_KEY_LEN;
6091 else if (g_strcmp0 (cipher, "aes-256-icm") == 0)
6092 return AES_256_KEY_LEN;
6094 GST_ERROR ("encryption algorithm '%s' not supported", cipher);
6100 auth_key_length_from_auth_name (const gchar * auth)
6102 if (g_strcmp0 (auth, "hmac-sha1-32") == 0)
6103 return HMAC_32_KEY_LEN;
6104 else if (g_strcmp0 (auth, "hmac-sha1-80") == 0)
6105 return HMAC_80_KEY_LEN;
6107 GST_ERROR ("authentication algorithm '%s' not supported", auth);
6113 signal_get_srtcp_params (GstRTSPSrc * src, GstRTSPStream * stream)
6115 GstCaps *caps = NULL;
6117 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_REQUEST_RTCP_KEY], 0,
6121 GST_DEBUG_OBJECT (src, "SRTP parameters received");
6127 default_srtcp_params (void)
6135 /* create a random key */
6136 key_data = g_malloc (KEY_SIZE);
6137 for (i = 0; i < KEY_SIZE; i += 4)
6138 GST_WRITE_UINT32_BE (key_data + i, g_random_int ());
6140 buf = gst_buffer_new_wrapped (key_data, KEY_SIZE);
6142 caps = gst_caps_new_simple ("application/x-srtp",
6143 "srtp-key", GST_TYPE_BUFFER, buf,
6144 "srtcp-cipher", G_TYPE_STRING, "aes-128-icm",
6145 "srtcp-auth", G_TYPE_STRING, "hmac-sha1-80", NULL);
6147 gst_buffer_unref (buf);
6153 gst_rtspsrc_stream_make_keymgmt (GstRTSPSrc * src, GstRTSPStream * stream)
6156 gchar *result, *base64;
6159 GstMIKEYMessage *msg;
6160 GstMIKEYPayload *payload, *pkd;
6166 const gchar *srtcpcipher, *srtcpauth;
6168 stream->srtcpparams = signal_get_srtcp_params (src, stream);
6169 if (stream->srtcpparams == NULL)
6170 stream->srtcpparams = default_srtcp_params ();
6172 s = gst_caps_get_structure (stream->srtcpparams, 0);
6174 srtcpcipher = gst_structure_get_string (s, "srtcp-cipher");
6175 srtcpauth = gst_structure_get_string (s, "srtcp-auth");
6176 val = gst_structure_get_value (s, "srtp-key");
6178 if (srtcpcipher == NULL || srtcpauth == NULL || val == NULL) {
6179 GST_ERROR_OBJECT (src, "could not find the right SRTP parameters in caps");
6183 srtpkey = gst_value_get_buffer (val);
6185 msg = gst_mikey_message_new ();
6186 /* unencrypted MIKEY message, we send this over TLS so this is allowed */
6187 gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT,
6188 FALSE, GST_MIKEY_PRF_MIKEY_1, g_random_int (), GST_MIKEY_MAP_TYPE_SRTP);
6189 /* add policy '0' for our SSRC */
6190 gst_mikey_message_add_cs_srtp (msg, 0, stream->send_ssrc, 0);
6191 /* timestamp is now */
6192 gst_mikey_message_add_t_now_ntp_utc (msg);
6193 /* add some random data */
6194 gst_mikey_message_add_rand_len (msg, 16);
6196 /* the policy '0' is SRTP */
6197 payload = gst_mikey_payload_new (GST_MIKEY_PT_SP);
6198 gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP);
6200 /* only AES-CM is supported */
6202 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1, &byte);
6203 /* encryption key length */
6204 byte = enc_key_length_from_cipher_name (srtcpcipher);
6205 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_KEY_LEN, 1,
6207 /* only HMAC-SHA1 */
6208 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1,
6210 /* authentication key length */
6211 byte = auth_key_length_from_auth_name (srtcpauth);
6212 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_KEY_LEN, 1,
6214 /* we enable encryption on RTP and RTCP */
6215 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1,
6217 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1,
6219 /* we enable authentication on RTP and RTCP */
6220 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1,
6222 gst_mikey_message_add_payload (msg, payload);
6224 /* make unencrypted KEMAC */
6225 payload = gst_mikey_payload_new (GST_MIKEY_PT_KEMAC);
6226 gst_mikey_payload_kemac_set (payload, GST_MIKEY_ENC_NULL, GST_MIKEY_MAC_NULL);
6227 /* add the key in KEMAC */
6228 pkd = gst_mikey_payload_new (GST_MIKEY_PT_KEY_DATA);
6229 gst_buffer_map (srtpkey, &info, GST_MAP_READ);
6230 gst_mikey_payload_key_data_set_key (pkd, GST_MIKEY_KD_TEK, info.size,
6232 gst_buffer_unmap (srtpkey, &info);
6233 gst_mikey_payload_kemac_add_sub (payload, pkd);
6234 gst_mikey_message_add_payload (msg, payload);
6236 /* now serialize this to bytes */
6237 bytes = gst_mikey_message_to_bytes (msg, NULL, NULL);
6238 gst_mikey_message_unref (msg);
6239 /* and make it into base64 */
6240 data = g_bytes_get_data (bytes, &size);
6241 base64 = g_base64_encode (data, size);
6242 g_bytes_unref (bytes);
6244 result = g_strdup_printf ("prot=mikey;uri=\"%s\";data=\"%s\"",
6245 stream->conninfo.location, base64);
6252 /* Perform the SETUP request for all the streams.
6254 * We ask the server for a specific transport, which initially includes all the
6255 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
6256 * two local UDP ports that we send to the server.
6258 * Once the server replied with a transport, we configure the other streams
6259 * with the same transport.
6261 * This function will also configure the stream for the selected transport,
6262 * which basically means creating the pipeline.
6264 static GstRTSPResult
6265 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
6268 GstRTSPResult res = GST_RTSP_ERROR;
6269 GstRTSPMessage request = { 0 };
6270 GstRTSPMessage response = { 0 };
6271 GstRTSPStream *stream = NULL;
6272 GstRTSPLowerTrans protocols;
6273 GstRTSPStatusCode code;
6274 gboolean unsupported_real = FALSE;
6275 gint rtpport, rtcpport;
6279 if (src->conninfo.connection) {
6280 url = gst_rtsp_connection_get_url (src->conninfo.connection);
6281 /* we initially allow all configured lower transports. based on the URL
6282 * transports and the replies from the server we narrow them down. */
6283 protocols = url->transports & src->cur_protocols;
6286 protocols = src->cur_protocols;
6292 /* reset some state */
6293 src->free_channel = 0;
6294 src->interleaved = FALSE;
6295 src->need_activate = FALSE;
6296 /* keep track of next port number, 0 is random */
6297 src->next_port_num = src->client_port_range.min;
6298 rtpport = rtcpport = 0;
6300 if (G_UNLIKELY (src->streams == NULL))
6303 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6304 GstRTSPConnection *conn;
6311 stream = (GstRTSPStream *) walk->data;
6313 caps = stream_get_caps_for_pt (stream, stream->default_pt);
6315 GST_DEBUG_OBJECT (src, "skipping stream %p, no caps", stream);
6319 if (stream->skipped) {
6320 GST_DEBUG_OBJECT (src, "skipping stream %p", stream);
6324 /* see if we need to configure this stream */
6325 if (!gst_rtsp_ext_list_configure_stream (src->extensions, caps)) {
6326 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
6331 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_SELECT_STREAM], 0,
6332 stream->id, caps, &selected);
6334 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by signal", stream);
6338 /* merge/overwrite global caps */
6343 s = gst_caps_get_structure (caps, 0);
6345 num = gst_structure_n_fields (src->props);
6346 for (j = 0; j < num; j++) {
6350 name = gst_structure_nth_field_name (src->props, j);
6351 val = gst_structure_get_value (src->props, name);
6352 gst_structure_set_value (s, name, val);
6354 GST_DEBUG_OBJECT (src, "copied %s", name);
6358 /* skip setup if we have no URL for it */
6359 if (stream->conninfo.location == NULL) {
6360 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
6364 if (src->conninfo.connection == NULL) {
6365 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
6366 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
6369 conn = stream->conninfo.connection;
6371 conn = src->conninfo.connection;
6373 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
6374 stream->conninfo.location);
6376 /* if we have a multicast connection, only suggest multicast from now on */
6377 if (stream->is_multicast)
6378 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
6381 /* first selectable protocol */
6382 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
6384 if (!protocol_masks[mask])
6388 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
6389 protocol_masks[mask]);
6390 /* create a string with first transport in line */
6392 res = gst_rtspsrc_create_transports_string (src,
6393 protocols & protocol_masks[mask], stream->profile, &transports);
6394 if (res < 0 || transports == NULL)
6395 goto setup_transport_failed;
6397 if (strlen (transports) == 0) {
6398 g_free (transports);
6399 GST_DEBUG_OBJECT (src, "no transports found");
6404 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
6406 /* replace placeholders with real values, this function will optionally
6407 * allocate UDP ports and other info needed to execute the setup request */
6408 res = gst_rtspsrc_prepare_transports (stream, &transports,
6409 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
6411 g_free (transports);
6412 goto setup_transport_failed;
6415 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
6417 /* create SETUP request */
6419 gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
6420 stream->conninfo.location);
6422 g_free (transports);
6423 goto create_request_failed;
6426 /* select transport */
6427 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
6430 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
6431 stream->profile == GST_RTSP_PROFILE_SAVPF) {
6432 hval = gst_rtspsrc_stream_make_keymgmt (src, stream);
6433 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_KEYMGMT, hval);
6436 /* if the user wants a non default RTP packet size we add the blocksize
6438 if (src->rtp_blocksize > 0) {
6439 hval = g_strdup_printf ("%d", src->rtp_blocksize);
6440 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
6444 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
6447 /* handle the code ourselves */
6448 res = gst_rtspsrc_send (src, conn, &request, &response, &code);
6453 case GST_RTSP_STS_OK:
6455 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
6456 gst_rtsp_message_unset (&request);
6457 gst_rtsp_message_unset (&response);
6458 /* cleanup of leftover transport */
6459 gst_rtspsrc_stream_free_udp (stream);
6460 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
6461 * we might be in this case */
6462 if (stream->container && rtpport && rtcpport && !retry) {
6463 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
6468 /* this transport did not go down well, but we may have others to try
6469 * that we did not send yet, try those and only give up then
6470 * but not without checking for lost cause/extension so we can
6471 * post a nicer/more useful error message later */
6472 if (!unsupported_real)
6473 unsupported_real = stream->is_real;
6474 /* select next available protocol, give up on this stream if none */
6476 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
6478 if (!protocol_masks[mask] || unsupported_real)
6483 /* cleanup of leftover transport and move to the next stream */
6484 gst_rtspsrc_stream_free_udp (stream);
6485 goto response_error;
6488 /* parse response transport */
6490 gchar *resptrans = NULL;
6491 GstRTSPTransport transport = { 0 };
6493 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
6496 gst_rtspsrc_stream_free_udp (stream);
6500 /* parse transport, go to next stream on parse error */
6501 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
6502 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
6506 /* update allowed transports for other streams. once the transport of
6507 * one stream has been determined, we make sure that all other streams
6508 * are configured in the same way */
6509 switch (transport.lower_transport) {
6510 case GST_RTSP_LOWER_TRANS_TCP:
6511 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
6512 protocols = GST_RTSP_LOWER_TRANS_TCP;
6513 src->interleaved = TRUE;
6514 /* update free channels */
6516 MAX (transport.interleaved.min, src->free_channel);
6518 MAX (transport.interleaved.max, src->free_channel);
6519 src->free_channel++;
6521 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
6522 /* only allow multicast for other streams */
6523 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
6524 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
6525 /* if the server selected our ports, increment our counters so that
6526 * we select a new port later */
6527 if (src->next_port_num == transport.port.min &&
6528 src->next_port_num + 1 == transport.port.max) {
6529 src->next_port_num += 2;
6532 case GST_RTSP_LOWER_TRANS_UDP:
6533 /* only allow unicast for other streams */
6534 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
6535 protocols = GST_RTSP_LOWER_TRANS_UDP;
6538 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
6539 transport.lower_transport);
6543 if (!src->interleaved || !retry) {
6544 /* now configure the stream with the selected transport */
6545 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
6546 GST_DEBUG_OBJECT (src,
6547 "could not configure stream %p transport, skipping stream",
6550 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
6551 /* retain the first allocated UDP port pair */
6552 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
6553 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
6556 /* we need to activate at least one streams when we detect activity */
6557 src->need_activate = TRUE;
6559 /* stream is setup now */
6560 stream->setup = TRUE;
6565 GstRTSPStream *sskip;
6567 skip = g_list_next (skip);
6571 sskip = (GstRTSPStream *) skip->data;
6573 /* skip all streams with the same control url */
6574 if (g_str_equal (stream->conninfo.location, sskip->conninfo.location)) {
6575 GST_DEBUG_OBJECT (src, "found stream %p with same control %s",
6576 sskip, sskip->conninfo.location);
6577 sskip->skipped = TRUE;
6582 /* clean up our transport struct */
6583 gst_rtsp_transport_init (&transport);
6584 /* clean up used RTSP messages */
6585 gst_rtsp_message_unset (&request);
6586 gst_rtsp_message_unset (&response);
6590 /* store the transport protocol that was configured */
6591 src->cur_protocols = protocols;
6593 gst_rtsp_ext_list_stream_select (src->extensions, url);
6595 /* if there is nothing to activate, error out */
6596 if (!src->need_activate)
6597 goto nothing_to_activate;
6604 /* no transport possible, post an error and stop */
6605 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
6606 ("Could not connect to server, no protocols left"));
6607 return GST_RTSP_ERROR;
6611 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6612 ("SDP contains no streams"));
6613 return GST_RTSP_ERROR;
6615 create_request_failed:
6617 gchar *str = gst_rtsp_strresult (res);
6619 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6620 ("Could not create request. (%s)", str));
6624 setup_transport_failed:
6626 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6627 ("Could not setup transport."));
6628 res = GST_RTSP_ERROR;
6633 const gchar *str = gst_rtsp_status_as_text (code);
6635 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6636 ("Error (%d): %s", code, GST_STR_NULL (str)));
6637 res = GST_RTSP_ERROR;
6642 gchar *str = gst_rtsp_strresult (res);
6644 if (res != GST_RTSP_EINTR) {
6645 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6646 ("Could not send message. (%s)", str));
6648 GST_WARNING_OBJECT (src, "send interrupted");
6655 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6656 ("Server did not select transport."));
6657 res = GST_RTSP_ERROR;
6660 nothing_to_activate:
6662 /* none of the available error codes is really right .. */
6663 if (unsupported_real) {
6664 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6665 (_("No supported stream was found. You might need to install a "
6666 "GStreamer RTSP extension plugin for Real media streams.")),
6669 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6670 (_("No supported stream was found. You might need to allow "
6671 "more transport protocols or may otherwise be missing "
6672 "the right GStreamer RTSP extension plugin.")), (NULL));
6674 return GST_RTSP_ERROR;
6678 gst_rtsp_message_unset (&request);
6679 gst_rtsp_message_unset (&response);
6685 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
6686 GstSegment * segment)
6689 GstRTSPTimeRange *therange;
6692 gst_rtsp_range_free (src->range);
6694 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
6695 GST_DEBUG_OBJECT (src, "parsed range %s", range);
6696 src->range = therange;
6698 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
6700 gst_segment_init (segment, GST_FORMAT_TIME);
6704 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
6705 therange->min.type, therange->min.seconds, therange->max.type,
6706 therange->max.seconds);
6708 if (therange->min.type == GST_RTSP_TIME_NOW)
6710 else if (therange->min.type == GST_RTSP_TIME_END)
6713 seconds = therange->min.seconds * GST_SECOND;
6715 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
6716 GST_TIME_ARGS (seconds));
6718 /* we need to start playback without clipping from the position reported by
6720 segment->start = seconds;
6721 segment->position = seconds;
6723 if (therange->max.type == GST_RTSP_TIME_NOW)
6725 else if (therange->max.type == GST_RTSP_TIME_END)
6728 seconds = therange->max.seconds * GST_SECOND;
6730 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
6731 GST_TIME_ARGS (seconds));
6733 /* live (WMS) server might send overflowed large max as its idea of infinity,
6734 * compensate to prevent problems later on */
6735 if (seconds != -1 && seconds < 0) {
6737 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
6740 /* live (WMS) might send min == max, which is not worth recording */
6741 if (segment->duration == -1 && seconds == segment->start)
6744 /* don't change duration with unknown value, we might have a valid value
6745 * there that we want to keep. */
6747 segment->duration = seconds;
6752 /* Parse clock profived by the server with following syntax:
6754 * "GstNetTimeProvider <wrapped-clock> <server-IP:port> <clock-time>"
6757 gst_rtspsrc_parse_gst_clock (GstRTSPSrc * src, const gchar * gstclock)
6759 gboolean res = FALSE;
6761 if (g_str_has_prefix (gstclock, "GstNetTimeProvider ")) {
6762 gchar **fields = NULL, **parts = NULL;
6763 gchar *remote_ip, *str;
6765 GstClockTime base_time;
6768 fields = g_strsplit (gstclock, " ", 0);
6770 /* wrapped clock, not very interesting for now */
6771 if (fields[1] == NULL)
6774 /* remote IP address and port */
6775 if ((str = fields[2]) == NULL)
6778 parts = g_strsplit (str, ":", 0);
6780 if ((remote_ip = parts[0]) == NULL)
6783 if ((str = parts[1]) == NULL)
6791 if ((str = fields[3]) == NULL)
6794 base_time = g_ascii_strtoull (str, NULL, 10);
6797 gst_net_client_clock_new ((gchar *) "GstRTSPClock", remote_ip, port,
6800 if (src->provided_clock)
6801 gst_object_unref (src->provided_clock);
6802 src->provided_clock = netclock;
6804 gst_element_post_message (GST_ELEMENT_CAST (src),
6805 gst_message_new_clock_provide (GST_OBJECT_CAST (src),
6806 src->provided_clock, TRUE));
6810 g_strfreev (fields);
6816 /* must be called with the RTSP state lock */
6817 static GstRTSPResult
6818 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
6824 /* prepare global stream caps properties */
6826 gst_structure_remove_all_fields (src->props);
6828 src->props = gst_structure_new_empty ("RTSPProperties");
6831 gst_sdp_message_dump (sdp);
6833 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
6835 /* let the app inspect and change the SDP */
6836 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_ON_SDP], 0, sdp);
6838 gst_segment_init (&src->segment, GST_FORMAT_TIME);
6840 /* parse range for duration reporting. */
6845 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
6849 /* keep track of the range and configure it in the segment */
6850 if (gst_rtspsrc_parse_range (src, range, &src->segment))
6854 /* parse clock information. This is GStreamer specific, a server can tell the
6855 * client what clock it is using and wrap that in a network clock. The
6856 * advantage of that is that we can slave to it. */
6858 const gchar *gstclock;
6861 gstclock = gst_sdp_message_get_attribute_val_n (sdp, "x-gst-clock", i);
6862 if (gstclock == NULL)
6865 /* parse the clock and expose it in the provide_clock method */
6866 if (gst_rtspsrc_parse_gst_clock (src, gstclock))
6870 /* try to find a global control attribute. Note that a '*' means that we should
6871 * do aggregate control with the current url (so we don't do anything and
6872 * leave the current connection as is) */
6874 const gchar *control;
6877 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
6878 if (control == NULL)
6881 /* only take fully qualified urls */
6882 if (g_str_has_prefix (control, "rtsp://"))
6886 g_free (src->conninfo.location);
6887 src->conninfo.location = g_strdup (control);
6888 /* make a connection for this, if there was a connection already, nothing
6890 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
6891 GST_ERROR_OBJECT (src, "could not connect");
6894 /* we need to keep the control url separate from the connection url because
6895 * the rules for constructing the media control url need it */
6896 g_free (src->control);
6897 src->control = g_strdup (control);
6900 /* create streams */
6901 n_streams = gst_sdp_message_medias_len (sdp);
6902 for (i = 0; i < n_streams; i++) {
6903 gst_rtspsrc_create_stream (src, sdp, i);
6906 src->state = GST_RTSP_STATE_INIT;
6909 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
6912 /* reset our state */
6913 src->need_range = TRUE;
6916 src->state = GST_RTSP_STATE_READY;
6923 GST_ERROR_OBJECT (src, "setup failed");
6924 gst_rtspsrc_cleanup (src);
6929 static GstRTSPResult
6930 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
6934 GstRTSPMessage request = { 0 };
6935 GstRTSPMessage response = { 0 };
6938 gchar *respcont = NULL;
6941 src->need_redirect = FALSE;
6943 /* can't continue without a valid url */
6944 if (G_UNLIKELY (src->conninfo.url == NULL)) {
6945 res = GST_RTSP_EINVAL;
6948 src->tried_url_auth = FALSE;
6950 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
6951 goto connect_failed;
6953 /* create OPTIONS */
6954 GST_DEBUG_OBJECT (src, "create options...");
6956 gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
6957 src->conninfo.url_str);
6959 goto create_request_failed;
6962 GST_DEBUG_OBJECT (src, "send options...");
6965 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
6968 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6973 if (!gst_rtspsrc_parse_methods (src, &response))
6976 /* create DESCRIBE */
6977 GST_DEBUG_OBJECT (src, "create describe...");
6979 gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
6980 src->conninfo.url_str);
6982 goto create_request_failed;
6984 /* we only accept SDP for now */
6985 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
6989 GST_DEBUG_OBJECT (src, "send describe...");
6992 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
6995 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6999 /* we only perform redirect for the describe, currently */
7000 if (src->need_redirect) {
7001 /* close connection, we don't have to send a TEARDOWN yet, ignore the
7003 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
7005 gst_rtsp_message_unset (&request);
7006 gst_rtsp_message_unset (&response);
7012 /* it could be that the DESCRIBE method was not implemented */
7013 if (!src->methods & GST_RTSP_DESCRIBE)
7016 /* check if reply is SDP */
7017 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
7019 /* could not be set but since the request returned OK, we assume it
7020 * was SDP, else check it. */
7022 if (g_ascii_strcasecmp (respcont, "application/sdp") != 0)
7023 goto wrong_content_type;
7026 /* get message body and parse as SDP */
7027 gst_rtsp_message_get_body (&response, &data, &size);
7028 if (data == NULL || size == 0)
7031 GST_DEBUG_OBJECT (src, "parse SDP...");
7032 gst_sdp_message_new (sdp);
7033 gst_sdp_message_parse_buffer (data, size, *sdp);
7035 /* clean up any messages */
7036 gst_rtsp_message_unset (&request);
7037 gst_rtsp_message_unset (&response);
7044 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
7045 ("No valid RTSP URL was provided"));
7050 gchar *str = gst_rtsp_strresult (res);
7052 if (res != GST_RTSP_EINTR) {
7053 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
7054 ("Failed to connect. (%s)", str));
7056 GST_WARNING_OBJECT (src, "connect interrupted");
7061 create_request_failed:
7063 gchar *str = gst_rtsp_strresult (res);
7065 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7066 ("Could not create request. (%s)", str));
7072 /* Don't post a message - the rtsp_send method will have
7073 * taken care of it because we passed NULL for the response code */
7078 /* error was posted */
7079 res = GST_RTSP_ERROR;
7084 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
7085 ("Server does not support SDP, got %s.", respcont));
7086 res = GST_RTSP_ERROR;
7091 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
7092 ("Server can not provide an SDP."));
7093 res = GST_RTSP_ERROR;
7098 if (src->conninfo.connection) {
7099 GST_DEBUG_OBJECT (src, "free connection");
7100 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
7102 gst_rtsp_message_unset (&request);
7103 gst_rtsp_message_unset (&response);
7108 static GstRTSPResult
7109 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
7114 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
7116 if (src->sdp == NULL) {
7117 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
7121 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
7126 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
7133 GST_WARNING_OBJECT (src, "can't get sdp");
7134 src->open_error = TRUE;
7139 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
7140 src->open_error = TRUE;
7145 static GstRTSPResult
7146 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
7148 GstRTSPMessage request = { 0 };
7149 GstRTSPMessage response = { 0 };
7150 GstRTSPResult res = GST_RTSP_OK;
7152 const gchar *control;
7154 GST_DEBUG_OBJECT (src, "TEARDOWN...");
7156 gst_rtspsrc_set_state (src, GST_STATE_READY);
7158 if (src->state < GST_RTSP_STATE_READY) {
7159 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
7166 /* construct a control url */
7167 control = get_aggregate_control (src);
7169 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
7172 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7173 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7174 const gchar *setup_url;
7175 GstRTSPConnInfo *info;
7177 /* try aggregate control first but do non-aggregate control otherwise */
7179 setup_url = control;
7180 else if ((setup_url = stream->conninfo.location) == NULL)
7183 if (src->conninfo.connection) {
7184 info = &src->conninfo;
7185 } else if (stream->conninfo.connection) {
7186 info = &stream->conninfo;
7190 if (!info->connected)
7195 gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
7197 goto create_request_failed;
7200 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
7203 gst_rtspsrc_send (src, info->connection, &request, &response,
7207 /* FIXME, parse result? */
7208 gst_rtsp_message_unset (&request);
7209 gst_rtsp_message_unset (&response);
7212 /* early exit when we did aggregate control */
7218 /* close connections */
7219 GST_DEBUG_OBJECT (src, "closing connection...");
7220 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
7221 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7222 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7223 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
7227 gst_rtspsrc_cleanup (src);
7229 src->state = GST_RTSP_STATE_INVALID;
7232 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
7237 create_request_failed:
7239 gchar *str = gst_rtsp_strresult (res);
7241 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7242 ("Could not create request. (%s)", str));
7248 gchar *str = gst_rtsp_strresult (res);
7250 gst_rtsp_message_unset (&request);
7251 if (res != GST_RTSP_EINTR) {
7252 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7253 ("Could not send message. (%s)", str));
7255 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
7262 GST_DEBUG_OBJECT (src,
7263 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
7268 /* RTP-Info is of the format:
7270 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
7272 * rtptime corresponds to the timestamp for the NPT time given in the header
7273 * seqbase corresponds to the next sequence number we received. This number
7274 * indicates the first seqnum after the seek and should be used to discard
7275 * packets that are from before the seek.
7278 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
7283 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
7285 infos = g_strsplit (rtpinfo, ",", 0);
7286 for (i = 0; infos[i]; i++) {
7288 GstRTSPStream *stream;
7292 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
7294 /* init values, types of seqbase and timebase are bigger than needed so we
7295 * can store -1 as uninitialized values */
7300 /* parse url, find stream for url.
7301 * parse seq and rtptime. The seq number should be configured in the rtp
7302 * depayloader or session manager to detect gaps. Same for the rtptime, it
7303 * should be used to create an initial time newsegment. */
7304 fields = g_strsplit (infos[i], ";", 0);
7305 for (j = 0; fields[j]; j++) {
7306 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
7307 /* remove leading whitespace */
7308 fields[j] = g_strchug (fields[j]);
7309 if (g_str_has_prefix (fields[j], "url=")) {
7310 /* get the url and the stream */
7312 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
7313 } else if (g_str_has_prefix (fields[j], "seq=")) {
7314 seqbase = atoi (fields[j] + 4);
7315 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
7316 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
7319 g_strfreev (fields);
7320 /* now we need to store the values for the caps of the stream */
7321 if (stream != NULL) {
7322 GST_DEBUG_OBJECT (src,
7323 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
7324 stream, seqbase, timebase);
7326 /* we have a stream, configure detected params */
7327 stream->seqbase = seqbase;
7328 stream->timebase = timebase;
7337 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
7342 interval = strtoul (rtcp, NULL, 10);
7343 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
7348 interval *= GST_MSECOND;
7350 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7351 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7353 /* already (optionally) retrieved this when configuring manager */
7354 if (stream->session) {
7355 GObject *rtpsession = stream->session;
7357 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
7359 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
7363 /* now it happens that (Xenon) server sending this may also provide bogus
7364 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
7365 * and just use RTP-Info to sync */
7367 GObjectClass *klass;
7369 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
7370 if (g_object_class_find_property (klass, "rtcp-sync")) {
7371 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
7372 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
7378 gst_rtspsrc_get_float (const gchar * dstr)
7380 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
7382 /* canonicalise floating point string so we can handle float strings
7383 * in the form "24.930" or "24,930" irrespective of the current locale */
7384 g_strlcpy (s, dstr, sizeof (s));
7385 g_strdelimit (s, ",", '.');
7386 return g_ascii_strtod (s, NULL);
7390 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
7392 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
7394 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
7395 g_strlcpy (val_str, "now", sizeof (val_str));
7397 if (segment->position == 0) {
7398 g_strlcpy (val_str, "0", sizeof (val_str));
7400 g_ascii_dtostr (val_str, sizeof (val_str),
7401 ((gdouble) segment->position) / GST_SECOND);
7404 return g_strdup_printf ("npt=%s-", val_str);
7408 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
7412 stream->timebase = -1;
7413 stream->seqbase = -1;
7415 len = stream->ptmap->len;
7416 for (i = 0; i < len; i++) {
7417 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
7420 if (item->caps == NULL)
7423 item->caps = gst_caps_make_writable (item->caps);
7424 s = gst_caps_get_structure (item->caps, 0);
7425 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
7429 static GstRTSPResult
7430 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
7432 GstRTSPResult res = GST_RTSP_OK;
7434 if (src->state < GST_RTSP_STATE_READY) {
7435 res = GST_RTSP_ERROR;
7436 if (src->open_error) {
7437 GST_DEBUG_OBJECT (src, "the stream was in error");
7441 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
7443 if ((res = gst_rtspsrc_open (src, async)) < 0) {
7444 GST_DEBUG_OBJECT (src, "failed to open stream");
7453 static GstRTSPResult
7454 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
7456 GstRTSPMessage request = { 0 };
7457 GstRTSPMessage response = { 0 };
7458 GstRTSPResult res = GST_RTSP_OK;
7462 const gchar *control;
7464 GST_DEBUG_OBJECT (src, "PLAY...");
7466 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
7469 if (!(src->methods & GST_RTSP_PLAY))
7472 if (src->state == GST_RTSP_STATE_PLAYING)
7475 if (!src->conninfo.connection || !src->conninfo.connected)
7478 /* send some dummy packets before we activate the receive in the
7480 gst_rtspsrc_send_dummy_packets (src);
7482 /* require new SR packets */
7484 g_signal_emit_by_name (src->manager, "reset-sync", NULL);
7486 /* construct a control url */
7487 control = get_aggregate_control (src);
7489 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7490 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7491 const gchar *setup_url;
7492 GstRTSPConnection *conn;
7494 /* try aggregate control first but do non-aggregate control otherwise */
7496 setup_url = control;
7497 else if ((setup_url = stream->conninfo.location) == NULL)
7500 if (src->conninfo.connection) {
7501 conn = src->conninfo.connection;
7502 } else if (stream->conninfo.connection) {
7503 conn = stream->conninfo.connection;
7509 res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
7511 goto create_request_failed;
7513 if (src->need_range) {
7514 hval = gen_range_header (src, segment);
7516 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
7518 /* store the newsegment event so it can be sent from the streaming thread. */
7519 if (src->start_segment)
7520 gst_event_unref (src->start_segment);
7521 src->start_segment = gst_event_new_segment (segment);
7524 if (segment->rate != 1.0) {
7525 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
7527 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
7529 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
7531 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
7535 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
7537 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7540 /* seek may have silently failed as it is not supported */
7541 if (!(src->methods & GST_RTSP_PLAY)) {
7542 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
7543 /* obviously it is supported as we made it here */
7544 src->methods |= GST_RTSP_PLAY;
7545 src->seekable = FALSE;
7546 /* but there is nothing to parse in the response,
7547 * so convey we have no idea and not to expect anything particular */
7548 clear_rtp_base (src, stream);
7552 /* need to do for all streams */
7553 for (run = src->streams; run; run = g_list_next (run))
7554 clear_rtp_base (src, (GstRTSPStream *) run->data);
7556 /* NOTE the above also disables npt based eos detection */
7557 /* and below forces position to 0,
7558 * which is visible feedback we lost the plot */
7559 segment->start = segment->position = src->last_pos;
7562 gst_rtsp_message_unset (&request);
7564 /* parse RTP npt field. This is the current position in the stream (Normal
7565 * Play Time) and should be put in the NEWSEGMENT position field. */
7566 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
7568 gst_rtspsrc_parse_range (src, hval, segment);
7570 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
7571 segment->rate = 1.0;
7573 /* parse Speed header. This is the intended playback rate of the stream
7574 * and should be put in the NEWSEGMENT rate field. */
7575 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
7576 0) == GST_RTSP_OK) {
7577 segment->rate = gst_rtspsrc_get_float (hval);
7578 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
7579 &hval, 0) == GST_RTSP_OK) {
7580 segment->rate = gst_rtspsrc_get_float (hval);
7583 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
7584 * for the RTP packets. If this is not present, we assume all starts from 0...
7585 * This is info for the RTP session manager that we pass to it in caps. */
7587 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
7588 &hval, hval_idx++) == GST_RTSP_OK)
7589 gst_rtspsrc_parse_rtpinfo (src, hval);
7591 /* some servers indicate RTCP parameters in PLAY response,
7592 * rather than properly in SDP */
7593 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
7594 &hval, 0) == GST_RTSP_OK)
7595 gst_rtspsrc_handle_rtcp_interval (src, hval);
7597 gst_rtsp_message_unset (&response);
7599 /* early exit when we did aggregate control */
7603 /* configure the caps of the streams after we parsed all headers. Only reset
7604 * the manager object when we set a new Range header (we did a seek) */
7605 gst_rtspsrc_configure_caps (src, segment, src->need_range);
7607 /* set to PLAYING after we have configured the caps, otherwise we
7608 * might end up calling request_key (with SRTP) while caps are still
7609 * being configured. */
7610 gst_rtspsrc_set_state (src, GST_STATE_PLAYING);
7612 /* set again when needed */
7613 src->need_range = FALSE;
7615 src->running = TRUE;
7616 src->base_time = -1;
7617 src->state = GST_RTSP_STATE_PLAYING;
7620 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
7621 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7622 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7623 stream->discont = TRUE;
7628 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
7635 GST_DEBUG_OBJECT (src, "failed to open stream");
7640 GST_DEBUG_OBJECT (src, "PLAY is not supported");
7645 GST_DEBUG_OBJECT (src, "we were already PLAYING");
7648 create_request_failed:
7650 gchar *str = gst_rtsp_strresult (res);
7652 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7653 ("Could not create request. (%s)", str));
7659 gchar *str = gst_rtsp_strresult (res);
7661 gst_rtsp_message_unset (&request);
7662 if (res != GST_RTSP_EINTR) {
7663 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7664 ("Could not send message. (%s)", str));
7666 GST_WARNING_OBJECT (src, "PLAY interrupted");
7673 static GstRTSPResult
7674 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
7676 GstRTSPResult res = GST_RTSP_OK;
7677 GstRTSPMessage request = { 0 };
7678 GstRTSPMessage response = { 0 };
7680 const gchar *control;
7682 GST_DEBUG_OBJECT (src, "PAUSE...");
7684 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
7687 if (!(src->methods & GST_RTSP_PAUSE))
7690 if (src->state == GST_RTSP_STATE_READY)
7693 if (!src->conninfo.connection || !src->conninfo.connected)
7696 /* construct a control url */
7697 control = get_aggregate_control (src);
7699 /* loop over the streams. We might exit the loop early when we could do an
7700 * aggregate control */
7701 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7702 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7703 GstRTSPConnection *conn;
7704 const gchar *setup_url;
7706 /* try aggregate control first but do non-aggregate control otherwise */
7708 setup_url = control;
7709 else if ((setup_url = stream->conninfo.location) == NULL)
7712 if (src->conninfo.connection) {
7713 conn = src->conninfo.connection;
7714 } else if (stream->conninfo.connection) {
7715 conn = stream->conninfo.connection;
7721 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
7722 ("Sending PAUSE request"));
7725 gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
7727 goto create_request_failed;
7729 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7732 gst_rtsp_message_unset (&request);
7733 gst_rtsp_message_unset (&response);
7735 /* exit early when we did agregate control */
7740 /* change element states now */
7741 gst_rtspsrc_set_state (src, GST_STATE_PAUSED);
7744 src->state = GST_RTSP_STATE_READY;
7748 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
7755 GST_DEBUG_OBJECT (src, "failed to open stream");
7760 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
7765 GST_DEBUG_OBJECT (src, "we were already PAUSED");
7768 create_request_failed:
7770 gchar *str = gst_rtsp_strresult (res);
7772 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7773 ("Could not create request. (%s)", str));
7779 gchar *str = gst_rtsp_strresult (res);
7781 gst_rtsp_message_unset (&request);
7782 if (res != GST_RTSP_EINTR) {
7783 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7784 ("Could not send message. (%s)", str));
7786 GST_WARNING_OBJECT (src, "PAUSE interrupted");
7794 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
7796 GstRTSPSrc *rtspsrc;
7798 rtspsrc = GST_RTSPSRC (bin);
7800 switch (GST_MESSAGE_TYPE (message)) {
7801 case GST_MESSAGE_EOS:
7802 gst_message_unref (message);
7804 case GST_MESSAGE_ELEMENT:
7806 const GstStructure *s = gst_message_get_structure (message);
7808 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
7809 gboolean ignore_timeout;
7811 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
7813 GST_OBJECT_LOCK (rtspsrc);
7814 ignore_timeout = rtspsrc->ignore_timeout;
7815 rtspsrc->ignore_timeout = TRUE;
7816 GST_OBJECT_UNLOCK (rtspsrc);
7818 /* we only act on the first udp timeout message, others are irrelevant
7819 * and can be ignored. */
7820 if (!ignore_timeout)
7821 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
7823 gst_message_unref (message);
7826 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7829 case GST_MESSAGE_ERROR:
7832 GstRTSPStream *stream;
7835 udpsrc = GST_MESSAGE_SRC (message);
7837 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
7838 GST_ELEMENT_NAME (udpsrc));
7840 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
7844 /* we ignore the RTCP udpsrc */
7845 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
7848 /* if we get error messages from the udp sources, that's not a problem as
7849 * long as not all of them error out. We also don't really know what the
7850 * problem is, the message does not give enough detail... */
7851 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
7852 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
7853 if (ret != GST_FLOW_OK)
7857 gst_message_unref (message);
7861 /* fatal but not our message, forward */
7862 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7867 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7873 /* the thread where everything happens */
7875 gst_rtspsrc_thread (GstRTSPSrc * src)
7879 GST_OBJECT_LOCK (src);
7880 cmd = src->pending_cmd;
7881 if (cmd == CMD_RECONNECT || cmd == CMD_PLAY || cmd == CMD_PAUSE
7882 || cmd == CMD_LOOP || cmd == CMD_OPEN)
7883 src->pending_cmd = CMD_LOOP;
7885 src->pending_cmd = CMD_WAIT;
7886 GST_DEBUG_OBJECT (src, "got command %d", cmd);
7888 /* we got the message command, so ensure communication is possible again */
7889 gst_rtspsrc_connection_flush (src, FALSE);
7891 src->busy_cmd = cmd;
7892 GST_OBJECT_UNLOCK (src);
7896 gst_rtspsrc_open (src, TRUE);
7899 gst_rtspsrc_play (src, &src->segment, TRUE);
7902 gst_rtspsrc_pause (src, TRUE);
7905 gst_rtspsrc_close (src, TRUE, FALSE);
7908 gst_rtspsrc_loop (src);
7911 gst_rtspsrc_reconnect (src, FALSE);
7917 GST_OBJECT_LOCK (src);
7918 /* and go back to sleep */
7919 if (src->pending_cmd == CMD_WAIT) {
7921 gst_task_pause (src->task);
7924 src->busy_cmd = CMD_WAIT;
7925 GST_OBJECT_UNLOCK (src);
7929 gst_rtspsrc_start (GstRTSPSrc * src)
7931 GST_DEBUG_OBJECT (src, "starting");
7933 GST_OBJECT_LOCK (src);
7935 src->pending_cmd = CMD_WAIT;
7937 if (src->task == NULL) {
7938 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src, NULL);
7939 if (src->task == NULL)
7942 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
7944 GST_OBJECT_UNLOCK (src);
7951 GST_OBJECT_UNLOCK (src);
7952 GST_ERROR_OBJECT (src, "failed to create task");
7958 gst_rtspsrc_stop (GstRTSPSrc * src)
7962 GST_DEBUG_OBJECT (src, "stopping");
7964 /* also cancels pending task */
7965 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_ALL);
7967 GST_OBJECT_LOCK (src);
7968 if ((task = src->task)) {
7970 GST_OBJECT_UNLOCK (src);
7972 gst_task_stop (task);
7974 /* make sure it is not running */
7975 GST_RTSP_STREAM_LOCK (src);
7976 GST_RTSP_STREAM_UNLOCK (src);
7978 /* now wait for the task to finish */
7979 gst_task_join (task);
7981 /* and free the task */
7982 gst_object_unref (GST_OBJECT (task));
7984 GST_OBJECT_LOCK (src);
7986 GST_OBJECT_UNLOCK (src);
7988 /* ensure synchronously all is closed and clean */
7989 gst_rtspsrc_close (src, FALSE, TRUE);
7994 static GstStateChangeReturn
7995 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
7997 GstRTSPSrc *rtspsrc;
7998 GstStateChangeReturn ret;
8000 rtspsrc = GST_RTSPSRC (element);
8002 switch (transition) {
8003 case GST_STATE_CHANGE_NULL_TO_READY:
8004 if (!gst_rtspsrc_start (rtspsrc))
8007 case GST_STATE_CHANGE_READY_TO_PAUSED:
8008 /* init some state */
8009 rtspsrc->cur_protocols = rtspsrc->protocols;
8010 /* first attempt, don't ignore timeouts */
8011 rtspsrc->ignore_timeout = FALSE;
8012 rtspsrc->open_error = FALSE;
8013 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
8015 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
8016 set_manager_buffer_mode (rtspsrc);
8018 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
8019 /* unblock the tcp tasks and make the loop waiting */
8020 if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
8021 /* make sure it is waiting before we send PAUSE or PLAY below */
8022 GST_RTSP_STREAM_LOCK (rtspsrc);
8023 GST_RTSP_STREAM_UNLOCK (rtspsrc);
8026 case GST_STATE_CHANGE_PAUSED_TO_READY:
8032 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
8033 if (ret == GST_STATE_CHANGE_FAILURE)
8036 switch (transition) {
8037 case GST_STATE_CHANGE_NULL_TO_READY:
8038 ret = GST_STATE_CHANGE_SUCCESS;
8040 case GST_STATE_CHANGE_READY_TO_PAUSED:
8041 ret = GST_STATE_CHANGE_NO_PREROLL;
8043 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
8044 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
8045 ret = GST_STATE_CHANGE_SUCCESS;
8047 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
8048 /* send pause request and keep the idle task around */
8049 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
8050 ret = GST_STATE_CHANGE_NO_PREROLL;
8052 case GST_STATE_CHANGE_PAUSED_TO_READY:
8053 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
8054 ret = GST_STATE_CHANGE_SUCCESS;
8056 case GST_STATE_CHANGE_READY_TO_NULL:
8057 gst_rtspsrc_stop (rtspsrc);
8058 ret = GST_STATE_CHANGE_SUCCESS;
8069 GST_DEBUG_OBJECT (rtspsrc, "start failed");
8070 return GST_STATE_CHANGE_FAILURE;
8075 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
8078 GstRTSPSrc *rtspsrc;
8080 rtspsrc = GST_RTSPSRC (element);
8082 if (GST_EVENT_IS_DOWNSTREAM (event)) {
8083 res = gst_rtspsrc_push_event (rtspsrc, event);
8085 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
8092 /*** GSTURIHANDLER INTERFACE *************************************************/
8095 gst_rtspsrc_uri_get_type (GType type)
8100 static const gchar *const *
8101 gst_rtspsrc_uri_get_protocols (GType type)
8103 static const gchar *protocols[] =
8104 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
8105 "rtsps", "rtspsu", "rtspst", "rtspsh", NULL
8112 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
8114 GstRTSPSrc *src = GST_RTSPSRC (handler);
8116 /* FIXME: make thread-safe */
8117 return g_strdup (src->conninfo.location);
8121 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
8127 GstRTSPUrl *newurl = NULL;
8128 GstSDPMessage *sdp = NULL;
8130 src = GST_RTSPSRC (handler);
8132 /* same URI, we're fine */
8133 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
8136 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
8137 sres = gst_sdp_message_new (&sdp);
8141 GST_DEBUG_OBJECT (src, "parsing SDP message");
8142 sres = gst_sdp_message_parse_uri (uri, sdp);
8147 GST_DEBUG_OBJECT (src, "parsing URI");
8148 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
8152 /* if worked, free previous and store new url object along with the original
8154 GST_DEBUG_OBJECT (src, "configuring URI");
8155 g_free (src->conninfo.location);
8156 src->conninfo.location = g_strdup (uri);
8157 gst_rtsp_url_free (src->conninfo.url);
8158 src->conninfo.url = newurl;
8159 g_free (src->conninfo.url_str);
8161 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
8163 src->conninfo.url_str = NULL;
8166 gst_sdp_message_free (src->sdp);
8168 src->from_sdp = sdp != NULL;
8170 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
8171 GST_DEBUG_OBJECT (src, "request uri is: %s",
8172 GST_STR_NULL (src->conninfo.url_str));
8179 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
8184 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", sres);
8185 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
8186 "Could not create SDP");
8191 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", sres,
8192 GST_STR_NULL (uri));
8193 gst_sdp_message_free (sdp);
8194 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
8200 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
8201 GST_STR_NULL (uri), res);
8202 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
8203 "Invalid RTSP URI");
8209 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
8211 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
8213 iface->get_type = gst_rtspsrc_uri_get_type;
8214 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
8215 iface->get_uri = gst_rtspsrc_uri_get_uri;
8216 iface->set_uri = gst_rtspsrc_uri_set_uri;