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 (g_str_equal (key, "key-mgmt")) {
1906 parse_keymgmt (attr->value, caps);
1910 /* string must be valid UTF8 */
1911 if (!g_utf8_validate (attr->value, -1, NULL))
1914 if (!g_str_has_prefix (key, "x-"))
1915 tofree = key = g_strdup_printf ("a-%s", key);
1919 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
1920 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
1926 static const gchar *
1927 rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
1936 if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
1939 if (sscanf (attr, "%d ", &val) != 1)
1949 * Mapping of caps to and from SDP fields:
1951 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1952 * a=fmtp:<payload> <param>[=<value>];...
1955 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
1958 const gchar *rtpmap;
1962 gchar *params = NULL;
1968 /* get and parse rtpmap */
1969 rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
1972 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
1974 g_warning ("error parsing rtpmap, ignoring");
1978 /* dynamic payloads need rtpmap or we fail */
1979 if (rtpmap == NULL && pt >= 96)
1982 /* check if we have a rate, if not, we need to look up the rate from the
1983 * default rates based on the payload types. */
1985 const GstRTPPayloadInfo *info;
1987 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
1988 /* dynamic types, use media and encoding_name */
1989 tmp = g_ascii_strdown (media->media, -1);
1990 info = gst_rtp_payload_info_for_name (tmp, name);
1993 /* static types, use payload type */
1994 info = gst_rtp_payload_info_for_pt (pt);
1998 if ((rate = info->clock_rate) == 0)
2001 /* we fail if we cannot find one */
2006 tmp = g_ascii_strdown (media->media, -1);
2007 caps = gst_caps_new_simple ("application/x-unknown",
2008 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
2010 s = gst_caps_get_structure (caps, 0);
2012 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
2014 /* encoding name must be upper case */
2016 tmp = g_ascii_strup (name, -1);
2017 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
2021 /* params must be lower case */
2022 if (params != NULL) {
2023 tmp = g_ascii_strdown (params, -1);
2024 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
2028 /* parse optional fmtp: field */
2029 if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
2035 /* p is now of the format <payload> <param>[=<value>];... */
2036 PARSE_INT (p, " ", payload);
2037 if (payload != -1 && payload == pt) {
2041 /* <param>[=<value>] are separated with ';' */
2042 pairs = g_strsplit (p, ";", 0);
2043 for (i = 0; pairs[i]; i++) {
2045 const gchar *val, *key;
2047 /* the key may not have a '=', the value can have other '='s */
2048 valpos = strstr (pairs[i], "=");
2050 /* we have a '=' and thus a value, remove the '=' with \0 */
2052 /* value is everything between '=' and ';'. We split the pairs at ;
2053 * boundaries so we can take the remainder of the value. Some servers
2054 * put spaces around the value which we strip off here. Alternatively
2055 * we could strip those spaces in the depayloaders should these spaces
2056 * actually carry any meaning in the future. */
2057 val = g_strstrip (valpos + 1);
2059 /* simple <param>;.. is translated into <param>=1;... */
2062 /* strip the key of spaces, convert key to lowercase but not the value. */
2063 key = g_strstrip (pairs[i]);
2064 if (strlen (key) > 1) {
2065 tmp = g_ascii_strdown (key, -1);
2066 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
2078 g_warning ("rtpmap type not given for dynamic payload %d", pt);
2083 g_warning ("rate unknown for payload type %d", pt);
2089 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
2090 gint * rtpport, gint * rtcpport)
2093 GstStateChangeReturn ret;
2094 GstElement *udpsrc0, *udpsrc1;
2095 gint tmp_rtp, tmp_rtcp;
2099 src = stream->parent;
2105 /* Start at next port */
2106 tmp_rtp = src->next_port_num;
2108 if (stream->is_ipv6)
2109 host = "udp://[::0]";
2111 host = "udp://0.0.0.0";
2113 /* try to allocate 2 UDP ports, the RTP port should be an even
2114 * number and the RTCP port should be the next (uneven) port */
2117 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
2118 tmp_rtp >= src->client_port_range.max)
2121 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2122 if (udpsrc0 == NULL)
2123 goto no_udp_protocol;
2124 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
2126 if (src->udp_buffer_size != 0)
2127 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
2130 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
2131 if (ret == GST_STATE_CHANGE_FAILURE) {
2133 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
2136 if (++count > src->retry)
2139 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2140 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2141 gst_object_unref (udpsrc0);
2144 GST_DEBUG_OBJECT (src, "retry %d", count);
2147 goto no_udp_protocol;
2150 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
2151 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
2153 /* check if port is even */
2154 if ((tmp_rtp & 0x01) != 0) {
2155 /* port not even, close and allocate another */
2156 if (++count > src->retry)
2159 GST_DEBUG_OBJECT (src, "RTP port not even");
2161 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2162 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2163 gst_object_unref (udpsrc0);
2166 GST_DEBUG_OBJECT (src, "retry %d", count);
2171 /* allocate port+1 for RTCP now */
2172 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2173 if (udpsrc1 == NULL)
2174 goto no_udp_rtcp_protocol;
2177 tmp_rtcp = tmp_rtp + 1;
2178 if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
2181 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
2183 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
2184 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
2185 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
2186 if (ret == GST_STATE_CHANGE_FAILURE) {
2187 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
2189 if (++count > src->retry)
2192 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2193 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2194 gst_object_unref (udpsrc0);
2197 GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
2198 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2199 gst_object_unref (udpsrc1);
2203 GST_DEBUG_OBJECT (src, "retry %d", count);
2207 /* all fine, do port check */
2208 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
2209 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
2211 /* this should not happen... */
2212 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
2215 /* we keep these elements, we configure all in configure_transport when the
2216 * server told us to really use the UDP ports. */
2217 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
2218 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
2219 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
2220 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
2222 /* keep track of next available port number when we have a range
2224 if (src->next_port_num != 0)
2225 src->next_port_num = tmp_rtcp + 1;
2232 GST_DEBUG_OBJECT (src, "could not get UDP source");
2237 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
2241 no_udp_rtcp_protocol:
2243 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
2248 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
2249 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
2255 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2256 gst_object_unref (udpsrc0);
2259 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2260 gst_object_unref (udpsrc1);
2267 gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
2272 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
2274 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2275 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2278 for (i = 0; i < 2; i++) {
2279 if (stream->udpsrc[i])
2280 gst_element_set_state (stream->udpsrc[i], state);
2286 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
2293 event = gst_event_new_flush_start ();
2294 GST_DEBUG_OBJECT (src, "start flush");
2296 state = GST_STATE_PAUSED;
2298 event = gst_event_new_flush_stop (FALSE);
2299 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
2302 state = GST_STATE_PLAYING;
2304 state = GST_STATE_PAUSED;
2306 gst_rtspsrc_push_event (src, event);
2307 gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
2308 gst_rtspsrc_set_state (src, state);
2311 static GstRTSPResult
2312 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
2313 GstRTSPMessage * message, GTimeVal * timeout)
2318 ret = gst_rtsp_connection_send (conn, message, timeout);
2320 ret = GST_RTSP_ERROR;
2325 static GstRTSPResult
2326 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
2327 GstRTSPMessage * message, GTimeVal * timeout)
2332 ret = gst_rtsp_connection_receive (conn, message, timeout);
2334 ret = GST_RTSP_ERROR;
2340 gst_rtspsrc_get_position (GstRTSPSrc * src)
2345 query = gst_query_new_position (GST_FORMAT_TIME);
2346 /* should be known somewhere down the stream (e.g. jitterbuffer) */
2347 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2348 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2352 if (stream->srcpad) {
2353 if (gst_pad_query (stream->srcpad, query)) {
2354 gst_query_parse_position (query, &fmt, &pos);
2355 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
2356 GST_TIME_ARGS (pos));
2357 src->last_pos = pos;
2367 gst_query_unref (query);
2371 gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
2373 src->state = GST_RTSP_STATE_SEEKING;
2374 /* PLAY will add the range header now. */
2375 src->need_range = TRUE;
2381 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
2386 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
2388 gboolean flush, skip;
2391 GstSegment seeksegment = { 0, };
2395 GST_DEBUG_OBJECT (src, "doing seek with event");
2397 gst_event_parse_seek (event, &rate, &format, &flags,
2398 &cur_type, &cur, &stop_type, &stop);
2400 /* no negative rates yet */
2404 /* we need TIME format */
2405 if (format != src->segment.format)
2408 GST_DEBUG_OBJECT (src, "doing seek without event");
2410 cur_type = GST_SEEK_TYPE_SET;
2411 stop_type = GST_SEEK_TYPE_SET;
2414 /* get flush flag */
2415 flush = flags & GST_SEEK_FLAG_FLUSH;
2416 skip = flags & GST_SEEK_FLAG_SKIP;
2418 /* now we need to make sure the streaming thread is stopped. We do this by
2419 * either sending a FLUSH_START event downstream which will cause the
2420 * streaming thread to stop with a WRONG_STATE.
2421 * For a non-flushing seek we simply pause the task, which will happen as soon
2422 * as it completes one iteration (and thus might block when the sink is
2423 * blocking in preroll). */
2425 GST_DEBUG_OBJECT (src, "starting flush");
2426 gst_rtspsrc_flush (src, TRUE, FALSE);
2429 gst_task_pause (src->task);
2433 /* we should now be able to grab the streaming thread because we stopped it
2434 * with the above flush/pause code */
2435 GST_RTSP_STREAM_LOCK (src);
2437 GST_DEBUG_OBJECT (src, "stopped streaming");
2439 /* stop flushing the rtsp connection so we can send PAUSE/PLAY below */
2440 gst_rtspsrc_connection_flush (src, FALSE);
2442 /* copy segment, we need this because we still need the old
2443 * segment when we close the current segment. */
2444 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
2446 /* configure the seek parameters in the seeksegment. We will then have the
2447 * right values in the segment to perform the seek */
2449 GST_DEBUG_OBJECT (src, "configuring seek");
2450 gst_segment_do_seek (&seeksegment, rate, format, flags,
2451 cur_type, cur, stop_type, stop, &update);
2454 /* figure out the last position we need to play. If it's configured (stop !=
2455 * -1), use that, else we play until the total duration of the file */
2456 if ((stop = seeksegment.stop) == -1)
2457 stop = seeksegment.duration;
2459 playing = (src->state == GST_RTSP_STATE_PLAYING);
2461 /* if we were playing, pause first */
2463 /* obtain current position in case seek fails */
2464 gst_rtspsrc_get_position (src);
2465 gst_rtspsrc_pause (src, FALSE);
2469 gst_rtspsrc_do_seek (src, &seeksegment);
2471 /* and continue playing */
2473 gst_rtspsrc_play (src, &seeksegment, FALSE);
2475 /* prepare for streaming again */
2477 /* if we started flush, we stop now */
2478 GST_DEBUG_OBJECT (src, "stopping flush");
2479 gst_rtspsrc_flush (src, FALSE, playing);
2482 /* now we did the seek and can activate the new segment values */
2483 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
2485 /* if we're doing a segment seek, post a SEGMENT_START message */
2486 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2487 gst_element_post_message (GST_ELEMENT_CAST (src),
2488 gst_message_new_segment_start (GST_OBJECT_CAST (src),
2489 src->segment.format, src->segment.position));
2492 /* now create the newsegment */
2493 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
2494 " to %" G_GINT64_FORMAT, src->segment.position, stop);
2497 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
2498 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2499 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2500 stream->discont = TRUE;
2503 GST_RTSP_STREAM_UNLOCK (src);
2510 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
2515 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
2521 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
2525 gboolean res = TRUE;
2528 src = GST_RTSPSRC_CAST (parent);
2530 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
2531 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
2533 switch (GST_EVENT_TYPE (event)) {
2534 case GST_EVENT_SEEK:
2535 res = gst_rtspsrc_perform_seek (src, event);
2539 case GST_EVENT_NAVIGATION:
2540 case GST_EVENT_LATENCY:
2548 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
2549 res = gst_pad_send_event (target, event);
2550 gst_object_unref (target);
2552 gst_event_unref (event);
2555 gst_event_unref (event);
2561 /* this is the final event function we receive on the internal source pad when
2562 * we deal with TCP connections */
2564 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
2569 GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
2571 switch (GST_EVENT_TYPE (event)) {
2572 case GST_EVENT_SEEK:
2574 case GST_EVENT_NAVIGATION:
2575 case GST_EVENT_LATENCY:
2577 gst_event_unref (event);
2584 /* this is the final query function we receive on the internal source pad when
2585 * we deal with TCP connections */
2587 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
2591 gboolean res = TRUE;
2593 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
2595 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2596 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2598 switch (GST_QUERY_TYPE (query)) {
2599 case GST_QUERY_POSITION:
2604 case GST_QUERY_DURATION:
2608 gst_query_parse_duration (query, &format, NULL);
2611 case GST_FORMAT_TIME:
2612 gst_query_set_duration (query, format, src->segment.duration);
2620 case GST_QUERY_LATENCY:
2622 /* we are live with a min latency of 0 and unlimited max latency, this
2623 * result will be updated by the session manager if there is any. */
2624 gst_query_set_latency (query, TRUE, 0, -1);
2634 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2636 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2640 gboolean res = FALSE;
2642 src = GST_RTSPSRC_CAST (parent);
2644 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2645 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2647 switch (GST_QUERY_TYPE (query)) {
2648 case GST_QUERY_DURATION:
2652 gst_query_parse_duration (query, &format, NULL);
2655 case GST_FORMAT_TIME:
2656 gst_query_set_duration (query, format, src->segment.duration);
2664 case GST_QUERY_SEEKING:
2668 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2669 if (format == GST_FORMAT_TIME) {
2671 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2673 /* seeking without duration is unlikely */
2674 seekable = seekable && src->seekable && src->segment.duration &&
2675 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2677 /* FIXME ?? should we have 0 and segment.duration here; see demuxers */
2678 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
2679 src->segment.start, src->segment.stop);
2688 uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
2690 gst_query_set_uri (query, uri);
2698 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2700 /* forward the query to the proxy target pad */
2702 res = gst_pad_query (target, query);
2703 gst_object_unref (target);
2712 /* callback for RTCP messages to be sent to the server when operating in TCP
2714 static GstFlowReturn
2715 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2718 GstRTSPStream *stream;
2719 GstFlowReturn res = GST_FLOW_OK;
2724 GstRTSPMessage message = { 0 };
2725 GstRTSPConnection *conn;
2727 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2728 src = stream->parent;
2730 gst_buffer_map (buffer, &map, GST_MAP_READ);
2734 gst_rtsp_message_init_data (&message, stream->channel[1]);
2736 /* lend the body data to the message */
2737 gst_rtsp_message_take_body (&message, data, size);
2739 if (stream->conninfo.connection)
2740 conn = stream->conninfo.connection;
2742 conn = src->conninfo.connection;
2744 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2745 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2746 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2748 /* and steal it away again because we will free it when unreffing the
2750 gst_rtsp_message_steal_body (&message, &data, &size);
2751 gst_rtsp_message_unset (&message);
2753 gst_buffer_unmap (buffer, &map);
2754 gst_buffer_unref (buffer);
2759 static GstPadProbeReturn
2760 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2762 GstRTSPSrc *src = user_data;
2764 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2765 GST_DEBUG_PAD_NAME (pad));
2767 /* activate the streams */
2768 GST_OBJECT_LOCK (src);
2769 if (!src->need_activate)
2772 src->need_activate = FALSE;
2773 GST_OBJECT_UNLOCK (src);
2775 gst_rtspsrc_activate_streams (src);
2777 return GST_PAD_PROBE_OK;
2781 GST_OBJECT_UNLOCK (src);
2782 return GST_PAD_PROBE_OK;
2787 copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
2789 GstPad *gpad = GST_PAD_CAST (user_data);
2791 GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
2792 gst_pad_store_sticky_event (gpad, *event);
2797 /* this callback is called when the session manager generated a new src pad with
2798 * payloaded RTP packets. We simply ghost the pad here. */
2800 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2803 GstPadTemplate *template;
2806 GstRTSPStream *stream;
2809 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2811 GST_RTSP_STATE_LOCK (src);
2813 name = gst_object_get_name (GST_OBJECT_CAST (pad));
2814 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
2815 goto unknown_stream;
2817 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);
2819 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
2821 goto unknown_stream;
2824 stream->ssrc = ssrc;
2826 /* we'll add it later see below */
2827 stream->added = TRUE;
2829 /* check if we added all streams */
2831 for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
2832 GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;
2834 GST_DEBUG_OBJECT (src, "stream %p, container %d, added %d, setup %d",
2835 ostream, ostream->container, ostream->added, ostream->setup);
2837 /* if we find a stream for which we did a setup that is not added, we
2838 * need to wait some more */
2839 if (ostream->setup && !ostream->added) {
2844 GST_RTSP_STATE_UNLOCK (src);
2846 /* create a new pad we will use to stream to */
2847 template = gst_static_pad_template_get (&rtptemplate);
2848 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
2849 gst_object_unref (template);
2852 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
2853 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
2854 gst_pad_set_active (stream->srcpad, TRUE);
2855 gst_pad_sticky_events_foreach (pad, copy_sticky_events, stream->srcpad);
2856 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
2859 GST_DEBUG_OBJECT (src, "We added all streams");
2860 /* when we get here, all stream are added and we can fire the no-more-pads
2862 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
2870 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
2871 GST_RTSP_STATE_UNLOCK (src);
2878 stream_get_caps_for_pt (GstRTSPStream * stream, guint pt)
2882 len = stream->ptmap->len;
2883 for (i = 0; i < len; i++) {
2884 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
2892 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
2894 GstRTSPStream *stream;
2897 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
2899 GST_RTSP_STATE_LOCK (src);
2900 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2902 goto unknown_stream;
2904 if ((caps = stream_get_caps_for_pt (stream, pt)))
2905 gst_caps_ref (caps);
2906 GST_RTSP_STATE_UNLOCK (src);
2912 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
2913 GST_RTSP_STATE_UNLOCK (src);
2919 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
2921 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
2927 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
2933 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
2939 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2941 GstRTSPSrc *src = stream->parent;
2944 g_object_get (source, "ssrc", &ssrc, NULL);
2946 GST_DEBUG_OBJECT (src, "source %08x, stream %08x, session %u received BYE",
2947 ssrc, stream->ssrc, stream->id);
2949 if (ssrc == stream->ssrc)
2950 gst_rtspsrc_do_stream_eos (src, stream);
2954 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2956 GstRTSPSrc *src = stream->parent;
2959 g_object_get (source, "ssrc", &ssrc, NULL);
2961 GST_WARNING_OBJECT (src, "source %08x, stream %08x in session %u timed out",
2962 ssrc, stream->ssrc, stream->id);
2964 if (ssrc == stream->ssrc)
2965 gst_rtspsrc_do_stream_eos (src, stream);
2969 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
2971 GstRTSPStream *stream;
2973 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
2975 /* get stream for session */
2976 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2978 gst_rtspsrc_do_stream_eos (src, stream);
2983 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
2985 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
2990 set_manager_buffer_mode (GstRTSPSrc * src)
2992 GObjectClass *klass;
2994 if (src->manager == NULL)
2997 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
2999 if (!g_object_class_find_property (klass, "buffer-mode"))
3002 if (src->buffer_mode != BUFFER_MODE_AUTO) {
3003 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
3008 GST_DEBUG_OBJECT (src,
3009 "auto buffering mode, have clock %" GST_PTR_FORMAT, src->provided_clock);
3011 if (src->provided_clock) {
3012 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
3014 if (clock == src->provided_clock) {
3015 GST_DEBUG_OBJECT (src, "selected synced");
3016 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SYNCED, NULL);
3019 gst_object_unref (clock);
3024 /* Otherwise fall-through and use another buffer mode */
3026 gst_object_unref (clock);
3029 GST_DEBUG_OBJECT (src, "auto buffering mode");
3030 if (src->use_buffering) {
3031 GST_DEBUG_OBJECT (src, "selected buffer");
3032 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER, NULL);
3034 GST_DEBUG_OBJECT (src, "selected slave");
3035 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
3040 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
3042 GST_DEBUG ("request key %u", ssrc);
3043 return gst_caps_ref (stream_get_caps_for_pt (stream, stream->default_pt));
3047 request_rtp_decoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
3049 GST_DEBUG ("decoder session %u, stream %p, %d", session, stream, stream->id);
3050 if (stream->id != session)
3053 if (stream->profile != GST_RTSP_PROFILE_SAVP &&
3054 stream->profile != GST_RTSP_PROFILE_SAVPF)
3057 if (stream->srtpdec == NULL) {
3060 name = g_strdup_printf ("srtpdec_%u", session);
3061 stream->srtpdec = gst_element_factory_make ("srtpdec", name);
3064 g_signal_connect (stream->srtpdec, "request-key",
3065 (GCallback) request_key, stream);
3067 return gst_object_ref (stream->srtpdec);
3071 request_rtcp_encoder (GstElement * rtpbin, guint session,
3072 GstRTSPStream * stream)
3077 GST_DEBUG ("decoder session %u, stream %p, %d", session, stream, stream->id);
3078 if (stream->id != session)
3081 if (stream->profile != GST_RTSP_PROFILE_SAVP &&
3082 stream->profile != GST_RTSP_PROFILE_SAVPF)
3085 if (stream->srtpenc == NULL) {
3088 name = g_strdup_printf ("srtpenc_%u", session);
3089 stream->srtpenc = gst_element_factory_make ("srtpenc", name);
3092 /* get RTCP crypto parameters from caps */
3093 s = gst_caps_get_structure (stream->srtcpparams, 0);
3097 GType ciphertype, authtype;
3098 GValue rtcp_cipher = G_VALUE_INIT, rtcp_auth = G_VALUE_INIT;
3100 ciphertype = g_type_from_name ("GstSrtpCipherType");
3101 authtype = g_type_from_name ("GstSrtpAuthType");
3102 g_value_init (&rtcp_cipher, ciphertype);
3103 g_value_init (&rtcp_auth, authtype);
3105 str = gst_structure_get_string (s, "srtcp-cipher");
3106 gst_value_deserialize (&rtcp_cipher, str);
3107 str = gst_structure_get_string (s, "srtcp-auth");
3108 gst_value_deserialize (&rtcp_auth, str);
3109 gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL);
3111 g_object_set_property (G_OBJECT (stream->srtpenc), "rtcp-cipher",
3113 g_object_set_property (G_OBJECT (stream->srtpenc), "rtcp-auth",
3115 g_object_set (stream->srtpenc, "key", buf, NULL);
3117 g_value_unset (&rtcp_cipher);
3118 g_value_unset (&rtcp_auth);
3119 gst_buffer_unref (buf);
3122 name = g_strdup_printf ("rtcp_sink_%d", session);
3123 pad = gst_element_get_request_pad (stream->srtpenc, name);
3125 gst_object_unref (pad);
3127 return gst_object_ref (stream->srtpenc);
3131 request_aux_receiver (GstElement * rtpbin, guint sessid, GstRTSPSrc * src)
3133 GstElement *rtx, *bin;
3136 GstRTSPStream *stream;
3138 stream = find_stream (src, &sessid, (gpointer) find_stream_by_id);
3140 GST_WARNING_OBJECT (src, "Stream %u not found", sessid);
3144 GST_INFO_OBJECT (src, "creating retransmision receiver for session %u "
3145 "with map %" GST_PTR_FORMAT, sessid, stream->rtx_pt_map);
3146 bin = gst_bin_new (NULL);
3147 rtx = gst_element_factory_make ("rtprtxreceive", NULL);
3148 g_object_set (rtx, "payload-type-map", stream->rtx_pt_map, NULL);
3149 gst_bin_add (GST_BIN (bin), rtx);
3151 pad = gst_element_get_static_pad (rtx, "src");
3152 name = g_strdup_printf ("src_%u", sessid);
3153 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
3155 gst_object_unref (pad);
3157 pad = gst_element_get_static_pad (rtx, "sink");
3158 name = g_strdup_printf ("sink_%u", sessid);
3159 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
3161 gst_object_unref (pad);
3167 add_retransmission (GstRTSPSrc * src, GstRTSPTransport * transport)
3172 if (transport->trans != GST_RTSP_TRANS_RTP)
3175 signal_id = g_signal_lookup ("request-aux-receiver",
3176 G_OBJECT_TYPE (src->manager));
3177 /* there's already something connected */
3178 if (g_signal_handler_find (src->manager, G_SIGNAL_MATCH_ID, signal_id, 0,
3179 NULL, NULL, NULL) != 0) {
3180 GST_DEBUG_OBJECT (src, "Not adding RTX AUX element as "
3181 "\"request-aux-receiver\" signal is "
3182 "already used by the application");
3186 /* build the retransmission payload type map */
3187 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3188 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3191 if (stream->rtx_pt_map)
3192 gst_structure_free (stream->rtx_pt_map);
3193 stream->rtx_pt_map = gst_structure_new_empty ("application/x-rtp-pt-map");
3195 for (i = 0; i < stream->ptmap->len; i++) {
3196 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
3197 GstStructure *s = gst_caps_get_structure (item->caps, 0);
3198 const gchar *encoding;
3200 /* we only care about RTX streams */
3201 if ((encoding = gst_structure_get_string (s, "encoding-name"))
3202 && g_strcmp0 (encoding, "RTX") == 0) {
3203 const gchar *stream_pt_s;
3206 if (gst_structure_get_int (s, "payload", &rtx_pt)
3207 && (stream_pt_s = gst_structure_get_string (s, "apt"))) {
3210 gst_structure_set (stream->rtx_pt_map, stream_pt_s, G_TYPE_UINT,
3217 GST_DEBUG_OBJECT (src, "built retransmission payload map for stream "
3218 "id %i: %" GST_PTR_FORMAT, stream->id, stream->rtx_pt_map);
3221 g_object_set (src->manager, "do-retransmission", TRUE, NULL);
3223 /* enable RFC4588 retransmission handling by setting rtprtxreceive
3224 * as the "aux" element of rtpbin */
3225 g_signal_connect (src->manager, "request-aux-receiver",
3226 (GCallback) request_aux_receiver, src);
3229 /* try to get and configure a manager */
3231 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
3232 GstRTSPTransport * transport)
3234 const gchar *manager;
3236 GstStateChangeReturn ret;
3238 /* find a manager */
3239 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
3243 GST_DEBUG_OBJECT (src, "using manager %s", manager);
3245 /* configure the manager */
3246 if (src->manager == NULL) {
3247 GObjectClass *klass;
3249 if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
3251 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
3255 goto use_no_manager;
3257 if (!(src->manager = gst_element_factory_make (manager, "manager")))
3258 goto manager_failed;
3261 /* we manage this element */
3262 gst_element_set_locked_state (src->manager, TRUE);
3263 gst_bin_add (GST_BIN_CAST (src), src->manager);
3265 ret = gst_element_set_state (src->manager, GST_STATE_PAUSED);
3266 if (ret == GST_STATE_CHANGE_FAILURE)
3267 goto start_manager_failure;
3269 g_object_set (src->manager, "latency", src->latency, NULL);
3271 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
3273 if (g_object_class_find_property (klass, "ntp-sync")) {
3274 g_object_set (src->manager, "ntp-sync", src->ntp_sync, NULL);
3277 if (g_object_class_find_property (klass, "use-pipeline-clock")) {
3278 g_object_set (src->manager, "use-pipeline-clock",
3279 src->use_pipeline_clock, NULL);
3282 if (src->sdes && g_object_class_find_property (klass, "sdes")) {
3283 g_object_set (src->manager, "sdes", src->sdes, NULL);
3286 if (g_object_class_find_property (klass, "drop-on-latency")) {
3287 g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
3291 /* buffer mode pauses are handled by adding offsets to buffer times,
3292 * but some depayloaders may have a hard time syncing output times
3293 * with such input times, e.g. container ones, most notably ASF */
3294 /* TODO alternatives are having an event that indicates these shifts,
3295 * or having rtsp extensions provide suggestion on buffer mode */
3296 /* valid duration implies not likely live pipeline,
3297 * so slaving in jitterbuffer does not make much sense
3298 * (and might mess things up due to bursts) */
3299 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
3300 src->segment.duration && stream->container) {
3301 src->use_buffering = TRUE;
3303 src->use_buffering = FALSE;
3306 set_manager_buffer_mode (src);
3308 /* connect to signals */
3309 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
3311 src->manager_sig_id =
3312 g_signal_connect (src->manager, "pad-added",
3313 (GCallback) new_manager_pad, src);
3314 src->manager_ptmap_id =
3315 g_signal_connect (src->manager, "request-pt-map",
3316 (GCallback) request_pt_map, src);
3318 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
3321 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_NEW_MANAGER], 0,
3324 if (src->do_retransmission)
3325 add_retransmission (src, transport);
3327 g_signal_connect (src->manager, "request-rtp-decoder",
3328 (GCallback) request_rtp_decoder, stream);
3329 g_signal_connect (src->manager, "request-rtcp-decoder",
3330 (GCallback) request_rtp_decoder, stream);
3331 g_signal_connect (src->manager, "request-rtcp-encoder",
3332 (GCallback) request_rtcp_encoder, stream);
3334 /* we stream directly to the manager, get some pads. Each RTSP stream goes
3335 * into a separate RTP session. */
3336 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
3337 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
3339 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
3340 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
3343 /* now configure the bandwidth in the manager */
3344 if (g_signal_lookup ("get-internal-session",
3345 G_OBJECT_TYPE (src->manager)) != 0) {
3346 GObject *rtpsession;
3348 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
3351 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
3353 stream->session = rtpsession;
3355 if (stream->as_bandwidth != -1) {
3356 GST_INFO_OBJECT (src, "setting AS: %f",
3357 (gdouble) (stream->as_bandwidth * 1000));
3358 g_object_set (rtpsession, "bandwidth",
3359 (gdouble) (stream->as_bandwidth * 1000), NULL);
3361 if (stream->rr_bandwidth != -1) {
3362 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
3363 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
3366 if (stream->rs_bandwidth != -1) {
3367 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
3368 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
3372 g_object_set (rtpsession, "probation", src->probation, NULL);
3374 g_object_set (rtpsession, "internal-ssrc", stream->send_ssrc, NULL);
3376 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
3378 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
3380 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
3382 g_signal_connect (rtpsession, "on-ssrc-active",
3383 (GCallback) on_ssrc_active, stream);
3394 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3399 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
3402 start_manager_failure:
3404 GST_DEBUG_OBJECT (src, "could not start session manager");
3409 /* free the UDP sources allocated when negotiating a transport.
3410 * This function is called when the server negotiated to a transport where the
3411 * UDP sources are not needed anymore, such as TCP or multicast. */
3413 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
3417 for (i = 0; i < 2; i++) {
3418 if (stream->udpsrc[i]) {
3419 GST_DEBUG ("free UDP source %d for stream %p", i, stream);
3420 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
3421 gst_object_unref (stream->udpsrc[i]);
3422 stream->udpsrc[i] = NULL;
3427 /* for TCP, create pads to send and receive data to and from the manager and to
3428 * intercept various events and queries
3431 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
3432 GstRTSPTransport * transport, GstPad ** outpad)
3435 GstPadTemplate *template;
3436 GstPad *pad0, *pad1;
3438 /* configure for interleaved delivery, nothing needs to be done
3439 * here, the loop function will call the chain functions of the
3440 * session manager. */
3441 stream->channel[0] = transport->interleaved.min;
3442 stream->channel[1] = transport->interleaved.max;
3443 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
3444 stream->channel[0], stream->channel[1]);
3446 /* we can remove the allocated UDP ports now */
3447 gst_rtspsrc_stream_free_udp (stream);
3449 /* no session manager, send data to srcpad directly */
3450 if (!stream->channelpad[0]) {
3451 GST_DEBUG_OBJECT (src, "no manager, creating pad");
3453 /* create a new pad we will use to stream to */
3454 name = g_strdup_printf ("stream_%u", stream->id);
3455 template = gst_static_pad_template_get (&rtptemplate);
3456 stream->channelpad[0] = gst_pad_new_from_template (template, name);
3457 gst_object_unref (template);
3460 /* set caps and activate */
3461 gst_pad_use_fixed_caps (stream->channelpad[0]);
3462 gst_pad_set_active (stream->channelpad[0], TRUE);
3464 *outpad = gst_object_ref (stream->channelpad[0]);
3466 GST_DEBUG_OBJECT (src, "using manager source pad");
3468 template = gst_static_pad_template_get (&anysrctemplate);
3470 /* allocate pads for sending the channel data into the manager */
3471 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
3472 gst_pad_link_full (pad0, stream->channelpad[0], GST_PAD_LINK_CHECK_NOTHING);
3473 gst_object_unref (stream->channelpad[0]);
3474 stream->channelpad[0] = pad0;
3475 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
3476 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
3477 gst_pad_set_element_private (pad0, src);
3478 gst_pad_set_active (pad0, TRUE);
3480 if (stream->channelpad[1]) {
3481 /* if we have a sinkpad for the other channel, create a pad and link to the
3483 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
3484 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
3485 gst_pad_link_full (pad1, stream->channelpad[1],
3486 GST_PAD_LINK_CHECK_NOTHING);
3487 gst_object_unref (stream->channelpad[1]);
3488 stream->channelpad[1] = pad1;
3489 gst_pad_set_active (pad1, TRUE);
3491 gst_object_unref (template);
3493 /* setup RTCP transport back to the server if we have to. */
3494 if (src->manager && src->do_rtcp) {
3497 template = gst_static_pad_template_get (&anysinktemplate);
3499 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
3500 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
3501 gst_pad_set_element_private (stream->rtcppad, stream);
3502 gst_pad_set_active (stream->rtcppad, TRUE);
3504 /* get session RTCP pad */
3505 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3506 pad = gst_element_get_request_pad (src->manager, name);
3511 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3512 gst_object_unref (pad);
3515 gst_object_unref (template);
3521 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
3522 GstRTSPTransport * transport, const gchar ** destination, gint * min,
3523 gint * max, guint * ttl)
3525 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3527 if (!(*destination = transport->destination))
3528 *destination = stream->destination;
3531 /* transport first */
3532 *min = transport->port.min;
3533 *max = transport->port.max;
3534 if (*min == -1 && *max == -1) {
3535 /* then try from SDP */
3536 if (stream->port != 0) {
3537 *min = stream->port;
3538 *max = stream->port + 1;
3544 if (!(*ttl = transport->ttl))
3549 /* first take the source, then the endpoint to figure out where to send
3551 if (!(*destination = transport->source)) {
3552 if (src->conninfo.connection)
3553 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
3554 else if (stream->conninfo.connection)
3556 gst_rtsp_connection_get_ip (stream->conninfo.connection);
3560 /* for unicast we only expect the ports here */
3561 *min = transport->server_port.min;
3562 *max = transport->server_port.max;
3567 /* For multicast create UDP sources and join the multicast group. */
3569 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
3570 GstRTSPTransport * transport, GstPad ** outpad)
3573 const gchar *destination;
3576 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
3578 /* we can remove the allocated UDP ports now */
3579 gst_rtspsrc_stream_free_udp (stream);
3581 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
3584 /* we need a destination now */
3585 if (destination == NULL)
3586 goto no_destination;
3588 /* we really need ports now or we won't be able to receive anything at all */
3589 if (min == -1 && max == -1)
3592 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
3593 destination, min, max);
3595 /* creating UDP source for RTP */
3597 uri = g_strdup_printf ("udp://%s:%d", destination, min);
3599 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3601 if (stream->udpsrc[0] == NULL)
3604 /* take ownership */
3605 gst_object_ref_sink (stream->udpsrc[0]);
3607 if (src->udp_buffer_size != 0)
3608 g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
3609 src->udp_buffer_size, NULL);
3611 if (src->multi_iface != NULL)
3612 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3613 src->multi_iface, NULL);
3616 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3617 gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
3620 /* creating another UDP source for RTCP */
3624 uri = g_strdup_printf ("udp://%s:%d", destination, max);
3626 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3628 if (stream->udpsrc[1] == NULL)
3631 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3632 stream->profile == GST_RTSP_PROFILE_SAVPF)
3633 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3635 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3636 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3637 gst_caps_unref (caps);
3639 /* take ownership */
3640 gst_object_ref_sink (stream->udpsrc[1]);
3642 if (src->multi_iface != NULL)
3643 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3644 src->multi_iface, NULL);
3646 gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
3653 GST_DEBUG_OBJECT (src, "no UDP source element found");
3658 GST_DEBUG_OBJECT (src, "no destination found");
3663 GST_DEBUG_OBJECT (src, "no ports found");
3668 /* configure the remainder of the UDP ports */
3670 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
3671 GstRTSPTransport * transport, GstPad ** outpad)
3673 /* we manage the UDP elements now. For unicast, the UDP sources where
3674 * allocated in the stream when we suggested a transport. */
3675 if (stream->udpsrc[0]) {
3678 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3679 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
3681 GST_DEBUG_OBJECT (src, "setting up UDP source");
3683 /* configure a timeout on the UDP port. When the timeout message is
3684 * posted, we assume UDP transport is not possible. We reconnect using TCP
3686 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
3687 src->udp_timeout * 1000, NULL);
3689 if ((caps = stream_get_caps_for_pt (stream, stream->default_pt)))
3690 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
3692 /* get output pad of the UDP source. */
3693 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
3695 /* save it so we can unblock */
3696 stream->blockedpad = *outpad;
3698 /* configure pad block on the pad. As soon as there is dataflow on the
3699 * UDP source, we know that UDP is not blocked by a firewall and we can
3700 * configure all the streams to let the application autoplug decoders. */
3702 gst_pad_add_probe (stream->blockedpad,
3703 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3704 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocked, src, NULL);
3706 if (stream->channelpad[0]) {
3707 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
3708 /* configure for UDP delivery, we need to connect the UDP pads to
3709 * the session plugin. */
3710 gst_pad_link_full (*outpad, stream->channelpad[0],
3711 GST_PAD_LINK_CHECK_NOTHING);
3712 gst_object_unref (*outpad);
3714 /* we connected to pad-added signal to get pads from the manager */
3716 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
3721 if (stream->udpsrc[1]) {
3724 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
3725 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
3727 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3728 stream->profile == GST_RTSP_PROFILE_SAVPF)
3729 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3731 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3732 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3733 gst_caps_unref (caps);
3735 if (stream->channelpad[1]) {
3738 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
3740 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
3741 gst_pad_link_full (pad, stream->channelpad[1],
3742 GST_PAD_LINK_CHECK_NOTHING);
3743 gst_object_unref (pad);
3745 /* leave unlinked */
3751 /* configure the UDP sink back to the server for status reports */
3753 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
3754 GstRTSPStream * stream, GstRTSPTransport * transport)
3757 gint rtp_port, rtcp_port;
3758 gboolean do_rtp, do_rtcp;
3759 const gchar *destination;
3764 /* get transport info */
3765 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
3766 &rtp_port, &rtcp_port, &ttl);
3768 /* see what we need to do */
3769 do_rtp = (rtp_port != -1);
3770 /* it's possible that the server does not want us to send RTCP in which case
3772 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
3774 /* we need a destination when we have RTP or RTCP ports */
3775 if (destination == NULL && (do_rtp || do_rtcp))
3776 goto no_destination;
3778 /* try to construct the fakesrc to the RTP port of the server to open up any
3781 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
3784 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
3785 stream->udpsink[0] =
3786 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3788 if (stream->udpsink[0] == NULL)
3789 goto no_sink_element;
3791 /* don't join multicast group, we will have the source socket do that */
3792 /* no sync or async state changes needed */
3793 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
3794 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3796 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3798 if (stream->udpsrc[0]) {
3799 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
3800 * so that NAT firewalls will open a hole for us */
3801 g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
3802 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
3803 /* configure socket and make sure udpsink does not close it when shutting
3804 * down, it belongs to udpsrc after all. */
3805 g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
3806 "close-socket", FALSE, NULL);
3807 g_object_unref (socket);
3810 /* the source for the dummy packets to open up NAT */
3811 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
3812 if (stream->fakesrc == NULL)
3813 goto no_fakesrc_element;
3815 /* random data in 5 buffers, a size of 200 bytes should be fine */
3816 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
3817 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
3819 /* we don't want to consider this a sink */
3820 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
3822 /* keep everything locked */
3823 gst_element_set_locked_state (stream->udpsink[0], TRUE);
3824 gst_element_set_locked_state (stream->fakesrc, TRUE);
3826 gst_object_ref (stream->udpsink[0]);
3827 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
3828 gst_object_ref (stream->fakesrc);
3829 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
3831 gst_element_link_pads_full (stream->fakesrc, "src", stream->udpsink[0],
3832 "sink", GST_PAD_LINK_CHECK_NOTHING);
3835 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
3838 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
3839 stream->udpsink[1] =
3840 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3842 if (stream->udpsink[1] == NULL)
3843 goto no_sink_element;
3845 /* don't join multicast group, we will have the source socket do that */
3846 /* no sync or async state changes needed */
3847 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
3848 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3850 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3852 if (stream->udpsrc[1]) {
3853 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
3854 * because some servers check the port number of where it sends RTCP to identify
3855 * the RTCP packets it receives */
3856 g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
3857 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
3858 /* configure socket and make sure udpsink does not close it when shutting
3859 * down, it belongs to udpsrc after all. */
3860 g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
3861 "close-socket", FALSE, NULL);
3862 g_object_unref (socket);
3865 /* we don't want to consider this a sink */
3866 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
3868 /* we keep this playing always */
3869 gst_element_set_locked_state (stream->udpsink[1], TRUE);
3870 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
3872 gst_object_ref (stream->udpsink[1]);
3873 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
3875 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
3877 /* get session RTCP pad */
3878 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3879 pad = gst_element_get_request_pad (src->manager, name);
3884 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3885 gst_object_unref (pad);
3894 GST_DEBUG_OBJECT (src, "no destination address specified");
3899 GST_DEBUG_OBJECT (src, "no UDP sink element found");
3904 GST_DEBUG_OBJECT (src, "no fakesrc element found");
3909 /* sets up all elements needed for streaming over the specified transport.
3910 * Does not yet expose the element pads, this will be done when there is actuall
3911 * dataflow detected, which might never happen when UDP is blocked in a
3912 * firewall, for example.
3915 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
3916 GstRTSPTransport * transport)
3919 GstPad *outpad = NULL;
3920 GstPadTemplate *template;
3922 const gchar *media_type;
3925 src = stream->parent;
3927 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
3929 /* get the proper media type for this stream now */
3930 if (gst_rtsp_transport_get_media_type (transport, &media_type) < 0)
3931 goto unknown_transport;
3933 goto unknown_transport;
3935 /* configure the final media type */
3936 GST_DEBUG_OBJECT (src, "setting media type to %s", media_type);
3938 len = stream->ptmap->len;
3939 for (i = 0; i < len; i++) {
3941 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
3943 if (item->caps == NULL)
3946 s = gst_caps_get_structure (item->caps, 0);
3947 gst_structure_set_name (s, media_type);
3948 /* set ssrc if known */
3949 if (transport->ssrc)
3950 gst_structure_set (s, "ssrc", G_TYPE_UINT, transport->ssrc, NULL);
3953 /* try to get and configure a manager, channelpad[0-1] will be configured with
3954 * the pads for the manager, or NULL when no manager is needed. */
3955 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
3958 switch (transport->lower_transport) {
3959 case GST_RTSP_LOWER_TRANS_TCP:
3960 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
3961 goto transport_failed;
3963 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3964 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
3965 goto transport_failed;
3966 /* fallthrough, the rest is the same for UDP and MCAST */
3967 case GST_RTSP_LOWER_TRANS_UDP:
3968 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
3969 goto transport_failed;
3970 /* configure udpsinks back to the server for RTCP messages and for the
3971 * dummy RTP messages to open NAT. */
3972 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
3973 goto transport_failed;
3976 goto unknown_transport;
3980 GST_DEBUG_OBJECT (src, "creating ghostpad");
3982 gst_pad_use_fixed_caps (outpad);
3984 /* create ghostpad, don't add just yet, this will be done when we activate
3986 name = g_strdup_printf ("stream_%u", stream->id);
3987 template = gst_static_pad_template_get (&rtptemplate);
3988 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
3989 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
3990 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
3991 gst_object_unref (template);
3994 gst_object_unref (outpad);
3996 /* mark pad as ok */
3997 stream->last_ret = GST_FLOW_OK;
4004 GST_DEBUG_OBJECT (src, "failed to configure transport");
4009 GST_DEBUG_OBJECT (src, "unknown transport");
4014 GST_DEBUG_OBJECT (src, "cannot get a session manager");
4019 /* send a couple of dummy random packets on the receiver RTP port to the server,
4020 * this should make a firewall think we initiated the data transfer and
4021 * hopefully allow packets to go from the sender port to our RTP receiver port */
4023 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
4027 if (src->nat_method != GST_RTSP_NAT_DUMMY)
4030 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4031 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4033 if (stream->fakesrc && stream->udpsink[0]) {
4034 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
4035 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
4036 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
4037 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
4038 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
4044 /* Adds the source pads of all configured streams to the element.
4045 * This code is performed when we detected dataflow.
4047 * We detect dataflow from either the _loop function or with pad probes on the
4051 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
4055 GST_DEBUG_OBJECT (src, "activating streams");
4057 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4058 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4060 if (stream->udpsrc[0]) {
4061 /* remove timeout, we are streaming now and timeouts will be handled by
4062 * the session manager and jitter buffer */
4063 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
4065 if (stream->srcpad) {
4066 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
4067 gst_pad_set_active (stream->srcpad, TRUE);
4069 /* if we don't have a session manager, set the caps now. If we have a
4070 * session, we will get a notification of the pad and the caps. */
4071 if (!src->manager) {
4074 caps = stream_get_caps_for_pt (stream, stream->default_pt);
4075 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
4076 gst_pad_set_caps (stream->srcpad, caps);
4079 if (!stream->added) {
4080 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
4081 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
4082 stream->added = TRUE;
4087 /* unblock all pads */
4088 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4089 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4091 if (stream->blockid) {
4092 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
4093 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
4094 stream->blockid = 0;
4102 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
4103 gboolean reset_manager)
4106 guint64 start, stop;
4107 gdouble play_speed, play_scale;
4109 GST_DEBUG_OBJECT (src, "configuring stream caps");
4111 start = segment->position;
4112 stop = segment->duration;
4113 play_speed = segment->rate;
4114 play_scale = segment->applied_rate;
4116 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4117 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4123 len = stream->ptmap->len;
4124 for (j = 0; j < len; j++) {
4126 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, j);
4128 if (item->caps == NULL)
4131 caps = gst_caps_make_writable (item->caps);
4133 if (stream->timebase != -1)
4134 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
4135 (guint) stream->timebase, NULL);
4136 if (stream->seqbase != -1)
4137 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
4138 (guint) stream->seqbase, NULL);
4139 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
4141 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
4142 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
4143 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
4146 GST_DEBUG_OBJECT (src, "stream %p, pt %d, caps %" GST_PTR_FORMAT, stream,
4149 if (item->pt == stream->default_pt && stream->udpsrc[0]) {
4150 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
4154 if (reset_manager && src->manager) {
4155 GST_DEBUG_OBJECT (src, "clear session");
4156 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
4160 static GstFlowReturn
4161 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
4166 /* store the value */
4167 stream->last_ret = ret;
4169 /* if it's success we can return the value right away */
4170 if (ret == GST_FLOW_OK)
4173 /* any other error that is not-linked can be returned right
4175 if (ret != GST_FLOW_NOT_LINKED)
4178 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
4179 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4180 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4182 ret = ostream->last_ret;
4183 /* some other return value (must be SUCCESS but we can return
4184 * other values as well) */
4185 if (ret != GST_FLOW_NOT_LINKED)
4188 /* if we get here, all other pads were unlinked and we return
4189 * NOT_LINKED then */
4195 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
4198 gboolean res = TRUE;
4200 /* only streams that have a connection to the outside world */
4204 if (stream->udpsrc[0]) {
4205 gst_event_ref (event);
4206 res = gst_element_send_event (stream->udpsrc[0], event);
4207 } else if (stream->channelpad[0]) {
4208 gst_event_ref (event);
4209 if (GST_PAD_IS_SRC (stream->channelpad[0]))
4210 res = gst_pad_push_event (stream->channelpad[0], event);
4212 res = gst_pad_send_event (stream->channelpad[0], event);
4215 if (stream->udpsrc[1]) {
4216 gst_event_ref (event);
4217 res &= gst_element_send_event (stream->udpsrc[1], event);
4218 } else if (stream->channelpad[1]) {
4219 gst_event_ref (event);
4220 if (GST_PAD_IS_SRC (stream->channelpad[1]))
4221 res &= gst_pad_push_event (stream->channelpad[1], event);
4223 res &= gst_pad_send_event (stream->channelpad[1], event);
4227 gst_event_unref (event);
4233 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
4236 gboolean res = TRUE;
4238 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4239 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4241 gst_event_ref (event);
4242 res &= gst_rtspsrc_stream_push_event (src, ostream, event);
4244 gst_event_unref (event);
4249 static GstRTSPResult
4250 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
4255 if (info->connection == NULL) {
4256 if (info->url == NULL) {
4257 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
4258 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
4262 /* create connection */
4263 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
4264 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
4265 goto could_not_create;
4268 g_free (info->url_str);
4269 info->url_str = gst_rtsp_url_get_request_uri (info->url);
4271 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
4273 if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
4274 if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
4275 src->tls_validation_flags))
4276 GST_WARNING_OBJECT (src, "Unable to set TLS validation flags");
4278 if (src->tls_database)
4279 gst_rtsp_connection_set_tls_database (info->connection,
4283 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
4284 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
4286 if (src->proxy_host) {
4287 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
4289 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
4294 if (!info->connected) {
4297 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
4298 ("Connecting to %s", info->location));
4299 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
4301 gst_rtsp_connection_connect (info->connection,
4302 src->ptcp_timeout)) < 0)
4303 goto could_not_connect;
4305 info->connected = TRUE;
4312 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
4317 gchar *str = gst_rtsp_strresult (res);
4318 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
4324 gchar *str = gst_rtsp_strresult (res);
4325 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
4331 static GstRTSPResult
4332 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
4335 GST_RTSP_STATE_LOCK (src);
4336 if (info->connected) {
4337 GST_DEBUG_OBJECT (src, "closing connection...");
4338 gst_rtsp_connection_close (info->connection);
4339 info->connected = FALSE;
4341 if (free && info->connection) {
4342 /* free connection */
4343 GST_DEBUG_OBJECT (src, "freeing connection...");
4344 gst_rtsp_connection_free (info->connection);
4345 info->connection = NULL;
4347 GST_RTSP_STATE_UNLOCK (src);
4351 static GstRTSPResult
4352 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
4357 GST_DEBUG_OBJECT (src, "reconnecting connection...");
4358 gst_rtsp_conninfo_close (src, info, FALSE);
4359 res = gst_rtsp_conninfo_connect (src, info, async);
4365 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
4369 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
4370 GST_RTSP_STATE_LOCK (src);
4371 if (src->conninfo.connection && src->conninfo.flushing != flush) {
4372 GST_DEBUG_OBJECT (src, "connection flush");
4373 gst_rtsp_connection_flush (src->conninfo.connection, flush);
4374 src->conninfo.flushing = flush;
4376 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4377 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4378 if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
4379 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
4380 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
4381 stream->conninfo.flushing = flush;
4384 GST_RTSP_STATE_UNLOCK (src);
4387 /* FIXME, handle server request, reply with OK, for now */
4388 static GstRTSPResult
4389 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
4390 GstRTSPMessage * request)
4392 GstRTSPMessage response = { 0 };
4395 GST_DEBUG_OBJECT (src, "got server request message");
4398 gst_rtsp_message_dump (request);
4400 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
4402 if (res == GST_RTSP_ENOTIMPL) {
4403 /* default implementation, send OK */
4404 GST_DEBUG_OBJECT (src, "prepare OK reply");
4406 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
4411 /* let app parse and reply */
4412 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST],
4413 0, request, &response);
4416 gst_rtsp_message_dump (&response);
4418 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
4422 gst_rtsp_message_unset (&response);
4423 } else if (res == GST_RTSP_EEOF)
4431 gst_rtsp_message_unset (&response);
4436 /* send server keep-alive */
4437 static GstRTSPResult
4438 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
4440 GstRTSPMessage request = { 0 };
4442 GstRTSPMethod method;
4443 const gchar *control;
4445 if (src->do_rtsp_keep_alive == FALSE) {
4446 GST_DEBUG_OBJECT (src, "do-rtsp-keep-alive is FALSE, not sending.");
4447 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4451 GST_DEBUG_OBJECT (src, "creating server keep-alive");
4453 /* find a method to use for keep-alive */
4454 if (src->methods & GST_RTSP_GET_PARAMETER)
4455 method = GST_RTSP_GET_PARAMETER;
4457 method = GST_RTSP_OPTIONS;
4459 control = get_aggregate_control (src);
4460 if (control == NULL)
4463 res = gst_rtsp_message_init_request (&request, method, control);
4468 gst_rtsp_message_dump (&request);
4471 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
4476 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4477 gst_rtsp_message_unset (&request);
4484 GST_WARNING_OBJECT (src, "no control url to send keepalive");
4489 gchar *str = gst_rtsp_strresult (res);
4491 gst_rtsp_message_unset (&request);
4492 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
4493 ("Could not send keep-alive. (%s)", str));
4499 static GstFlowReturn
4500 gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
4502 GstFlowReturn ret = GST_FLOW_OK;
4504 GstRTSPStream *stream;
4505 GstPad *outpad = NULL;
4512 channel = message->type_data.data.channel;
4514 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
4516 goto unknown_stream;
4518 if (channel == stream->channel[0]) {
4519 outpad = stream->channelpad[0];
4521 } else if (channel == stream->channel[1]) {
4522 outpad = stream->channelpad[1];
4528 /* take a look at the body to figure out what we have */
4529 gst_rtsp_message_get_body (message, &data, &size);
4531 goto invalid_length;
4533 /* channels are not correct on some servers, do extra check */
4534 if (data[1] >= 200 && data[1] <= 204) {
4535 /* hmm RTCP message switch to the RTCP pad of the same stream. */
4536 outpad = stream->channelpad[1];
4540 /* we have no clue what this is, just ignore then. */
4542 goto unknown_stream;
4544 /* take the message body for further processing */
4545 gst_rtsp_message_steal_body (message, &data, &size);
4547 /* strip the trailing \0 */
4550 buf = gst_buffer_new ();
4551 gst_buffer_append_memory (buf,
4552 gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
4554 /* don't need message anymore */
4555 gst_rtsp_message_unset (message);
4557 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
4560 if (src->need_activate) {
4566 guint group_id = gst_util_group_id_next ();
4569 /* generate an SHA256 sum of the URI */
4570 cs = g_checksum_new (G_CHECKSUM_SHA256);
4571 uri = src->conninfo.location;
4572 g_checksum_update (cs, (const guchar *) uri, strlen (uri));
4574 gst_segment_init (&segment, GST_FORMAT_TIME);
4576 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4577 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4581 g_strdup_printf ("%s/%d", g_checksum_get_string (cs), ostream->id);
4582 event = gst_event_new_stream_start (stream_id);
4583 gst_event_set_group_id (event, group_id);
4586 gst_rtspsrc_stream_push_event (src, ostream, event);
4588 if ((caps = stream_get_caps_for_pt (ostream, ostream->default_pt))) {
4589 /* only streams that have a connection to the outside world */
4590 if (ostream->setup) {
4591 if (ostream->udpsrc[0]) {
4592 gst_element_send_event (ostream->udpsrc[0],
4593 gst_event_new_caps (caps));
4594 } else if (ostream->channelpad[0]) {
4595 if (GST_PAD_IS_SRC (ostream->channelpad[0]))
4596 gst_pad_push_event (ostream->channelpad[0],
4597 gst_event_new_caps (caps));
4599 gst_pad_send_event (ostream->channelpad[0],
4600 gst_event_new_caps (caps));
4603 gst_caps_unref (caps);
4604 caps = gst_caps_new_empty_simple ("application/x-rtcp");
4606 if (ostream->udpsrc[1]) {
4607 gst_element_send_event (ostream->udpsrc[1],
4608 gst_event_new_caps (caps));
4609 } else if (ostream->channelpad[1]) {
4610 if (GST_PAD_IS_SRC (ostream->channelpad[1]))
4611 gst_pad_push_event (ostream->channelpad[1],
4612 gst_event_new_caps (caps));
4614 gst_pad_send_event (ostream->channelpad[1],
4615 gst_event_new_caps (caps));
4619 gst_caps_unref (caps);
4622 /* Push a SEGMENT event if we don't have one pending, if we have one
4623 * pending we will just send that one a few lines below to all pads
4625 if (!src->start_segment)
4626 gst_rtspsrc_stream_push_event (src, ostream,
4627 gst_event_new_segment (&segment));
4629 g_checksum_free (cs);
4631 gst_rtspsrc_activate_streams (src);
4632 src->need_activate = FALSE;
4635 if ((event = src->start_segment) != NULL) {
4636 src->start_segment = NULL;
4637 gst_rtspsrc_push_event (src, event);
4640 if (src->base_time == -1) {
4641 /* Take current running_time. This timestamp will be put on
4642 * the first buffer of each stream because we are a live source and so we
4643 * timestamp with the running_time. When we are dealing with TCP, we also
4644 * only timestamp the first buffer (using the DISCONT flag) because a server
4645 * typically bursts data, for which we don't want to compensate by speeding
4646 * up the media. The other timestamps will be interpollated from this one
4647 * using the RTP timestamps. */
4648 GST_OBJECT_LOCK (src);
4649 if (GST_ELEMENT_CLOCK (src)) {
4651 GstClockTime base_time;
4653 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
4654 base_time = GST_ELEMENT_CAST (src)->base_time;
4656 src->base_time = now - base_time;
4658 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
4659 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
4661 GST_OBJECT_UNLOCK (src);
4664 if (stream->discont && !is_rtcp) {
4665 /* mark first RTP buffer as discont */
4666 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
4667 stream->discont = FALSE;
4668 /* first buffer gets the timestamp, other buffers are not timestamped and
4669 * their presentation time will be interpollated from the rtp timestamps. */
4670 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
4671 GST_TIME_ARGS (src->base_time));
4673 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
4676 /* chain to the peer pad */
4677 if (GST_PAD_IS_SINK (outpad))
4678 ret = gst_pad_chain (outpad, buf);
4680 ret = gst_pad_push (outpad, buf);
4683 /* combine all stream flows for the data transport */
4684 ret = gst_rtspsrc_combine_flows (src, stream, ret);
4691 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
4692 gst_rtsp_message_unset (message);
4697 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4698 ("Short message received, ignoring."));
4699 gst_rtsp_message_unset (message);
4704 static GstFlowReturn
4705 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
4707 GstRTSPMessage message = { 0 };
4709 GstFlowReturn ret = GST_FLOW_OK;
4710 GTimeVal tv_timeout;
4713 /* get the next timeout interval */
4714 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4716 /* see if the timeout period expired */
4717 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
4718 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
4719 /* send keep-alive, only act on interrupt, a warning will be posted for
4721 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4723 /* get new timeout */
4724 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4727 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
4728 tv_timeout.tv_sec, tv_timeout.tv_usec);
4730 /* protect the connection with the connection lock so that we can see when
4731 * we are finished doing server communication */
4733 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4734 &message, src->ptcp_timeout);
4738 GST_DEBUG_OBJECT (src, "we received a server message");
4740 case GST_RTSP_EINTR:
4741 /* we got interrupted this means we need to stop */
4743 case GST_RTSP_ETIMEOUT:
4744 /* no reply, send keep alive */
4745 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4746 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4750 /* go EOS when the server closed the connection */
4756 switch (message.type) {
4757 case GST_RTSP_MESSAGE_REQUEST:
4758 /* server sends us a request message, handle it */
4760 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4762 if (res == GST_RTSP_EEOF)
4765 goto handle_request_failed;
4767 case GST_RTSP_MESSAGE_RESPONSE:
4768 /* we ignore response messages */
4769 GST_DEBUG_OBJECT (src, "ignoring response message");
4771 gst_rtsp_message_dump (&message);
4773 case GST_RTSP_MESSAGE_DATA:
4774 GST_DEBUG_OBJECT (src, "got data message");
4775 ret = gst_rtspsrc_handle_data (src, &message);
4776 if (ret != GST_FLOW_OK)
4777 goto handle_data_failed;
4780 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4785 g_assert_not_reached ();
4790 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4791 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4792 ("The server closed the connection."));
4793 src->conninfo.connected = FALSE;
4794 gst_rtsp_message_unset (&message);
4795 return GST_FLOW_EOS;
4799 gst_rtsp_message_unset (&message);
4800 GST_DEBUG_OBJECT (src, "got interrupted");
4801 return GST_FLOW_FLUSHING;
4805 gchar *str = gst_rtsp_strresult (res);
4807 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4808 ("Could not receive message. (%s)", str));
4811 gst_rtsp_message_unset (&message);
4812 return GST_FLOW_ERROR;
4814 handle_request_failed:
4816 gchar *str = gst_rtsp_strresult (res);
4818 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4819 ("Could not handle server message. (%s)", str));
4821 gst_rtsp_message_unset (&message);
4822 return GST_FLOW_ERROR;
4826 GST_DEBUG_OBJECT (src, "could no handle data message");
4831 static GstFlowReturn
4832 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
4835 GstRTSPMessage message = { 0 };
4839 GTimeVal tv_timeout;
4841 /* get the next timeout interval */
4842 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4844 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
4845 (gint) tv_timeout.tv_sec);
4847 gst_rtsp_message_unset (&message);
4849 /* we should continue reading the TCP socket because the server might
4850 * send us requests. When the session timeout expires, we need to send a
4851 * keep-alive request to keep the session open. */
4852 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4853 &message, &tv_timeout);
4857 GST_DEBUG_OBJECT (src, "we received a server message");
4859 case GST_RTSP_EINTR:
4860 /* we got interrupted, see what we have to do */
4862 case GST_RTSP_ETIMEOUT:
4863 /* send keep-alive, ignore the result, a warning will be posted. */
4864 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4865 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4869 /* server closed the connection. not very fatal for UDP, reconnect and
4870 * see what happens. */
4871 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4872 ("The server closed the connection."));
4873 if (src->udp_reconnect) {
4875 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
4882 GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
4884 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4885 ("Unhandled return value %d.", res));
4889 switch (message.type) {
4890 case GST_RTSP_MESSAGE_REQUEST:
4891 /* server sends us a request message, handle it */
4893 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4895 if (res == GST_RTSP_EEOF)
4898 goto handle_request_failed;
4900 case GST_RTSP_MESSAGE_RESPONSE:
4901 /* we ignore response and data messages */
4902 GST_DEBUG_OBJECT (src, "ignoring response message");
4904 gst_rtsp_message_dump (&message);
4905 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
4906 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
4907 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
4908 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
4909 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4916 case GST_RTSP_MESSAGE_DATA:
4917 /* we ignore response and data messages */
4918 GST_DEBUG_OBJECT (src, "ignoring data message");
4921 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4926 g_assert_not_reached ();
4928 /* we get here when the connection got interrupted */
4931 gst_rtsp_message_unset (&message);
4932 GST_DEBUG_OBJECT (src, "got interrupted");
4933 return GST_FLOW_FLUSHING;
4937 gchar *str = gst_rtsp_strresult (res);
4940 src->conninfo.connected = FALSE;
4941 if (res != GST_RTSP_EINTR) {
4942 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
4943 ("Could not connect to server. (%s)", str));
4945 ret = GST_FLOW_ERROR;
4947 ret = GST_FLOW_FLUSHING;
4953 gchar *str = gst_rtsp_strresult (res);
4955 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4956 ("Could not receive message. (%s)", str));
4958 return GST_FLOW_ERROR;
4960 handle_request_failed:
4962 gchar *str = gst_rtsp_strresult (res);
4965 gst_rtsp_message_unset (&message);
4966 if (res != GST_RTSP_EINTR) {
4967 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4968 ("Could not handle server message. (%s)", str));
4970 ret = GST_FLOW_ERROR;
4972 ret = GST_FLOW_FLUSHING;
4978 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4979 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4980 ("The server closed the connection."));
4981 src->conninfo.connected = FALSE;
4982 gst_rtsp_message_unset (&message);
4983 return GST_FLOW_EOS;
4987 static GstRTSPResult
4988 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
4990 GstRTSPResult res = GST_RTSP_OK;
4993 GST_DEBUG_OBJECT (src, "doing reconnect");
4995 GST_OBJECT_LOCK (src);
4996 /* only restart when the pads were not yet activated, else we were
4997 * streaming over UDP */
4998 restart = src->need_activate;
4999 GST_OBJECT_UNLOCK (src);
5001 /* no need to restart, we're done */
5005 /* we can try only TCP now */
5006 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
5008 /* close and cleanup our state */
5009 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
5012 /* see if we have TCP left to try. Also don't try TCP when we were configured
5014 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
5017 /* We post a warning message now to inform the user
5018 * that nothing happened. It's most likely a firewall thing. */
5019 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5020 ("Could not receive any UDP packets for %.4f seconds, maybe your "
5021 "firewall is blocking it. Retrying using a TCP connection.",
5022 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
5024 /* open new connection using tcp */
5025 if (gst_rtspsrc_open (src, async) < 0)
5028 /* start playback */
5029 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
5038 src->cur_protocols = 0;
5039 /* no transport possible, post an error and stop */
5040 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5041 ("Could not receive any UDP packets for %.4f seconds, maybe your "
5042 "firewall is blocking it. No other protocols to try.",
5043 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
5044 return GST_RTSP_ERROR;
5048 GST_DEBUG_OBJECT (src, "open failed");
5053 GST_DEBUG_OBJECT (src, "play failed");
5059 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
5063 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
5066 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
5069 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
5072 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
5080 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
5084 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
5087 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
5090 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
5093 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
5101 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
5105 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
5108 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
5111 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
5114 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
5122 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
5126 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
5129 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
5132 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
5135 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
5143 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
5145 if (ret == GST_RTSP_OK)
5146 gst_rtspsrc_loop_complete_cmd (src, cmd);
5147 else if (ret == GST_RTSP_EINTR)
5148 gst_rtspsrc_loop_cancel_cmd (src, cmd);
5150 gst_rtspsrc_loop_error_cmd (src, cmd);
5154 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
5157 gboolean flushed = FALSE;
5159 /* start new request */
5160 gst_rtspsrc_loop_start_cmd (src, cmd);
5162 GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
5164 GST_OBJECT_LOCK (src);
5165 old = src->pending_cmd;
5166 if (old == CMD_RECONNECT) {
5167 GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
5168 cmd = CMD_RECONNECT;
5170 if (old != CMD_WAIT) {
5171 src->pending_cmd = CMD_WAIT;
5172 GST_OBJECT_UNLOCK (src);
5173 /* cancel previous request */
5174 GST_DEBUG_OBJECT (src, "cancel previous request %d", old);
5175 gst_rtspsrc_loop_cancel_cmd (src, old);
5176 GST_OBJECT_LOCK (src);
5178 src->pending_cmd = cmd;
5179 /* interrupt if allowed */
5180 if (src->busy_cmd & mask) {
5181 GST_DEBUG_OBJECT (src, "connection flush busy %d", src->busy_cmd);
5182 gst_rtspsrc_connection_flush (src, TRUE);
5185 GST_DEBUG_OBJECT (src, "not interrupting busy cmd %d", src->busy_cmd);
5188 gst_task_start (src->task);
5189 GST_OBJECT_UNLOCK (src);
5195 gst_rtspsrc_loop (GstRTSPSrc * src)
5199 if (!src->conninfo.connection || !src->conninfo.connected)
5202 if (src->interleaved)
5203 ret = gst_rtspsrc_loop_interleaved (src);
5205 ret = gst_rtspsrc_loop_udp (src);
5207 if (ret != GST_FLOW_OK)
5215 GST_WARNING_OBJECT (src, "we are not connected");
5216 ret = GST_FLOW_FLUSHING;
5221 const gchar *reason = gst_flow_get_name (ret);
5223 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
5224 src->running = FALSE;
5225 if (ret == GST_FLOW_EOS) {
5226 /* perform EOS logic */
5227 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
5228 gst_element_post_message (GST_ELEMENT_CAST (src),
5229 gst_message_new_segment_done (GST_OBJECT_CAST (src),
5230 src->segment.format, src->segment.position));
5231 gst_rtspsrc_push_event (src,
5232 gst_event_new_segment_done (src->segment.format,
5233 src->segment.position));
5235 gst_rtspsrc_push_event (src, gst_event_new_eos ());
5237 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
5238 /* for fatal errors we post an error message, post the error before the
5239 * EOS so the app knows about the error first. */
5240 GST_ELEMENT_ERROR (src, STREAM, FAILED,
5241 ("Internal data flow error."),
5242 ("streaming task paused, reason %s (%d)", reason, ret));
5243 gst_rtspsrc_push_event (src, gst_event_new_eos ());
5245 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_LOOP);
5250 #ifndef GST_DISABLE_GST_DEBUG
5251 static const gchar *
5252 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
5256 while (method != 0) {
5273 static const gchar *
5274 gst_rtspsrc_skip_lws (const gchar * s)
5276 while (g_ascii_isspace (*s))
5281 static const gchar *
5282 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
5284 while (s > start && g_ascii_isspace (*(s - 1)))
5289 static const gchar *
5290 gst_rtspsrc_skip_commas (const gchar * s)
5292 /* The grammar allows for multiple commas */
5293 while (g_ascii_isspace (*s) || *s == ',')
5298 static const gchar *
5299 gst_rtspsrc_skip_item (const gchar * s)
5301 gboolean quoted = FALSE;
5302 const gchar *start = s;
5304 /* A list item ends at the last non-whitespace character
5305 * before a comma which is not inside a quoted-string. Or at
5306 * the end of the string.
5312 if (*s == '\\' && *(s + 1))
5321 return gst_rtspsrc_unskip_lws (s, start);
5325 gst_rtsp_decode_quoted_string (gchar * quoted_string)
5329 src = quoted_string + 1;
5330 dst = quoted_string;
5331 while (*src && *src != '"') {
5332 if (*src == '\\' && *(src + 1))
5339 /* Extract the authentication tokens that the server provided for each method
5340 * into an array of structures and give those to the connection object.
5343 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
5344 const gchar * header, gboolean * stale)
5346 GSList *list = NULL, *iter;
5348 gchar *item, *eq, *name_end, *value;
5350 g_return_if_fail (stale != NULL);
5352 gst_rtsp_connection_clear_auth_params (conn);
5355 /* Parse a header whose content is described by RFC2616 as
5356 * "#something", where "something" does not itself contain commas,
5357 * except as part of quoted-strings, into a list of allocated strings.
5359 header = gst_rtspsrc_skip_commas (header);
5361 end = gst_rtspsrc_skip_item (header);
5362 list = g_slist_prepend (list, g_strndup (header, end - header));
5363 header = gst_rtspsrc_skip_commas (end);
5368 list = g_slist_reverse (list);
5369 for (iter = list; iter; iter = iter->next) {
5372 eq = strchr (item, '=');
5374 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
5375 if (name_end == item) {
5376 /* That's no good... */
5383 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
5385 gst_rtsp_decode_quoted_string (value);
5389 if (value && strcmp (item, "stale") == 0 && strcmp (value, "TRUE") == 0)
5391 gst_rtsp_connection_set_auth_param (conn, item, value);
5395 g_slist_free (list);
5398 /* Parse a WWW-Authenticate Response header and determine the
5399 * available authentication methods
5401 * This code should also cope with the fact that each WWW-Authenticate
5402 * header can contain multiple challenge methods + tokens
5404 * At the moment, for Basic auth, we just do a minimal check and don't
5405 * even parse out the realm */
5407 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
5408 GstRTSPConnection * conn, gboolean * stale)
5412 g_return_if_fail (hdr != NULL);
5413 g_return_if_fail (methods != NULL);
5414 g_return_if_fail (stale != NULL);
5416 /* Skip whitespace at the start of the string */
5417 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
5419 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
5420 *methods |= GST_RTSP_AUTH_BASIC;
5421 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
5422 *methods |= GST_RTSP_AUTH_DIGEST;
5423 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
5428 * gst_rtspsrc_setup_auth:
5429 * @src: the rtsp source
5431 * Configure a username and password and auth method on the
5432 * connection object based on a response we received from the
5435 * Currently, this requires that a username and password were supplied
5436 * in the uri. In the future, they may be requested on demand by sending
5437 * a message up the bus.
5439 * Returns: TRUE if authentication information could be set up correctly.
5442 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
5446 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
5447 GstRTSPAuthMethod method;
5448 GstRTSPResult auth_result;
5450 GstRTSPConnection *conn;
5452 gboolean stale = FALSE;
5454 conn = src->conninfo.connection;
5456 /* Identify the available auth methods and see if any are supported */
5457 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
5458 &hdr, 0) == GST_RTSP_OK) {
5459 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
5462 if (avail_methods == GST_RTSP_AUTH_NONE)
5463 goto no_auth_available;
5465 /* For digest auth, if the response indicates that the session
5466 * data are stale, we just update them in the connection object and
5467 * return TRUE to retry the request */
5469 src->tried_url_auth = FALSE;
5471 url = gst_rtsp_connection_get_url (conn);
5473 /* Do we have username and password available? */
5474 if (url != NULL && !src->tried_url_auth && url->user != NULL
5475 && url->passwd != NULL) {
5478 src->tried_url_auth = TRUE;
5479 GST_DEBUG_OBJECT (src,
5480 "Attempting authentication using credentials from the URL");
5482 user = src->user_id;
5483 pass = src->user_pw;
5484 GST_DEBUG_OBJECT (src,
5485 "Attempting authentication using credentials from the properties");
5488 /* FIXME: If the url didn't contain username and password or we tried them
5489 * already, request a username and passwd from the application via some kind
5490 * of credentials request message */
5492 /* If we don't have a username and passwd at this point, bail out. */
5493 if (user == NULL || pass == NULL)
5496 /* Try to configure for each available authentication method, strongest to
5498 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
5499 /* Check if this method is available on the server */
5500 if ((method & avail_methods) == 0)
5503 /* Pass the credentials to the connection to try on the next request */
5504 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
5505 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
5506 * ignore it and end up retrying later */
5507 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
5508 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
5509 gst_rtsp_auth_method_to_string (method));
5514 if (method == GST_RTSP_AUTH_NONE)
5515 goto no_auth_available;
5521 /* Output an error indicating that we couldn't connect because there were
5522 * no supported authentication protocols */
5523 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5524 ("No supported authentication protocol was found"));
5529 /* We don't fire an error message, we just return FALSE and let the
5530 * normal NOT_AUTHORIZED error be propagated */
5535 static GstRTSPResult
5536 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5537 GstRTSPMessage * request, GstRTSPMessage * response,
5538 GstRTSPStatusCode * code)
5541 GstRTSPStatusCode thecode;
5542 gchar *content_base = NULL;
5546 if (!src->short_header)
5547 gst_rtsp_ext_list_before_send (src->extensions, request);
5549 GST_DEBUG_OBJECT (src, "sending message");
5552 gst_rtsp_message_dump (request);
5554 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
5558 gst_rtsp_connection_reset_timeout (conn);
5561 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
5566 gst_rtsp_message_dump (response);
5568 switch (response->type) {
5569 case GST_RTSP_MESSAGE_REQUEST:
5570 res = gst_rtspsrc_handle_request (src, conn, response);
5571 if (res == GST_RTSP_EEOF)
5574 goto handle_request_failed;
5576 case GST_RTSP_MESSAGE_RESPONSE:
5577 /* ok, a response is good */
5578 GST_DEBUG_OBJECT (src, "received response message");
5580 case GST_RTSP_MESSAGE_DATA:
5581 /* get next response */
5582 GST_DEBUG_OBJECT (src, "handle data response message");
5583 gst_rtspsrc_handle_data (src, response);
5586 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
5591 thecode = response->type_data.response.code;
5593 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
5595 /* if the caller wanted the result code, we store it. */
5599 /* If the request didn't succeed, bail out before doing any more */
5600 if (thecode != GST_RTSP_STS_OK)
5603 /* store new content base if any */
5604 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
5607 g_free (src->content_base);
5608 src->content_base = g_strdup (content_base);
5610 gst_rtsp_ext_list_after_send (src->extensions, request, response);
5617 gchar *str = gst_rtsp_strresult (res);
5619 if (res != GST_RTSP_EINTR) {
5620 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5621 ("Could not send message. (%s)", str));
5623 GST_WARNING_OBJECT (src, "send interrupted");
5632 GST_WARNING_OBJECT (src, "server closed connection");
5633 if ((try == 0) && !src->interleaved && src->udp_reconnect) {
5635 /* if reconnect succeeds, try again */
5637 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
5641 /* only try once after reconnect, then fallthrough and error out */
5644 gchar *str = gst_rtsp_strresult (res);
5646 if (res != GST_RTSP_EINTR) {
5647 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5648 ("Could not receive message. (%s)", str));
5650 GST_WARNING_OBJECT (src, "receive interrupted");
5658 handle_request_failed:
5660 /* ERROR was posted */
5661 gst_rtsp_message_unset (response);
5666 GST_DEBUG_OBJECT (src, "we got an eof from the server");
5667 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5668 ("The server closed the connection."));
5669 gst_rtsp_message_unset (response);
5676 * @src: the rtsp source
5677 * @conn: the connection to send on
5678 * @request: must point to a valid request
5679 * @response: must point to an empty #GstRTSPMessage
5680 * @code: an optional code result
5682 * send @request and retrieve the response in @response. optionally @code can be
5683 * non-NULL in which case it will contain the status code of the response.
5685 * If This function returns #GST_RTSP_OK, @response will contain a valid response
5686 * message that should be cleaned with gst_rtsp_message_unset() after usage.
5688 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
5689 * @response message) if the response code was not 200 (OK).
5691 * If the attempt results in an authentication failure, then this will attempt
5692 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
5695 * Returns: #GST_RTSP_OK if the processing was successful.
5697 static GstRTSPResult
5698 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5699 GstRTSPMessage * request, GstRTSPMessage * response,
5700 GstRTSPStatusCode * code)
5702 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
5703 GstRTSPResult res = GST_RTSP_ERROR;
5706 GstRTSPMethod method = GST_RTSP_INVALID;
5712 /* make sure we don't loop forever */
5716 /* save method so we can disable it when the server complains */
5717 method = request->type_data.request.method;
5720 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
5724 case GST_RTSP_STS_UNAUTHORIZED:
5725 if (gst_rtspsrc_setup_auth (src, response)) {
5726 /* Try the request/response again after configuring the auth info
5734 } while (retry == TRUE);
5736 /* If the user requested the code, let them handle errors, otherwise
5737 * post an error below */
5740 else if (int_code != GST_RTSP_STS_OK)
5741 goto error_response;
5748 GST_DEBUG_OBJECT (src, "got error %d", res);
5753 res = GST_RTSP_ERROR;
5755 switch (response->type_data.response.code) {
5756 case GST_RTSP_STS_NOT_FOUND:
5757 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
5758 response->type_data.response.reason));
5760 case GST_RTSP_STS_UNAUTHORIZED:
5761 GST_ELEMENT_ERROR (src, RESOURCE, NOT_AUTHORIZED, (NULL), ("%s",
5762 response->type_data.response.reason));
5764 case GST_RTSP_STS_MOVED_PERMANENTLY:
5765 case GST_RTSP_STS_MOVE_TEMPORARILY:
5767 gchar *new_location;
5768 GstRTSPLowerTrans transports;
5770 GST_DEBUG_OBJECT (src, "got redirection");
5771 /* if we don't have a Location Header, we must error */
5772 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
5773 &new_location, 0) < 0)
5776 /* When we receive a redirect result, we go back to the INIT state after
5777 * parsing the new URI. The caller should do the needed steps to issue
5778 * a new setup when it detects this state change. */
5779 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
5781 /* save current transports */
5782 if (src->conninfo.url)
5783 transports = src->conninfo.url->transports;
5785 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
5787 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
5789 /* set old transports */
5790 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
5791 src->conninfo.url->transports = transports;
5793 src->need_redirect = TRUE;
5794 src->state = GST_RTSP_STATE_INIT;
5798 case GST_RTSP_STS_NOT_ACCEPTABLE:
5799 case GST_RTSP_STS_NOT_IMPLEMENTED:
5800 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
5801 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
5802 gst_rtsp_method_as_text (method));
5803 src->methods &= ~method;
5807 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5808 ("Got error response: %d (%s).", response->type_data.response.code,
5809 response->type_data.response.reason));
5812 /* if we return ERROR we should unset the response ourselves */
5813 if (res == GST_RTSP_ERROR)
5814 gst_rtsp_message_unset (response);
5820 static GstRTSPResult
5821 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
5822 GstRTSPMessage * response, GstRTSPSrc * src)
5824 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
5829 /* parse the response and collect all the supported methods. We need this
5830 * information so that we don't try to send an unsupported request to the
5834 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
5836 GstRTSPHeaderField field;
5840 /* reset supported methods */
5843 /* Try Allow Header first */
5844 field = GST_RTSP_HDR_ALLOW;
5847 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5848 if (indx == 0 && !respoptions) {
5849 /* if no Allow header was found then try the Public header... */
5850 field = GST_RTSP_HDR_PUBLIC;
5851 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5856 src->methods |= gst_rtsp_options_from_text (respoptions);
5861 if (src->methods == 0) {
5862 /* neither Allow nor Public are required, assume the server supports
5863 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
5865 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
5866 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
5868 /* always assume PLAY, FIXME, extensions should be able to override
5870 src->methods |= GST_RTSP_PLAY;
5871 /* also assume it will support Range */
5872 src->seekable = TRUE;
5874 /* we need describe and setup */
5875 if (!(src->methods & GST_RTSP_DESCRIBE))
5877 if (!(src->methods & GST_RTSP_SETUP))
5885 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5886 ("Server does not support DESCRIBE."));
5891 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5892 ("Server does not support SETUP."));
5897 /* masks to be kept in sync with the hardcoded protocol order of preference
5899 static const guint protocol_masks[] = {
5900 GST_RTSP_LOWER_TRANS_UDP,
5901 GST_RTSP_LOWER_TRANS_UDP_MCAST,
5902 GST_RTSP_LOWER_TRANS_TCP,
5906 static GstRTSPResult
5907 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
5908 GstRTSPLowerTrans protocols, GstRTSPProfile profile, gchar ** transports)
5912 gboolean add_udp_str;
5917 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
5922 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
5924 /* extension listed transports, use those */
5925 if (*transports != NULL)
5928 /* it's the default */
5929 add_udp_str = FALSE;
5931 /* the default RTSP transports */
5932 result = g_string_new ("RTP");
5935 case GST_RTSP_PROFILE_AVP:
5936 g_string_append (result, "/AVP");
5938 case GST_RTSP_PROFILE_SAVP:
5939 g_string_append (result, "/SAVP");
5941 case GST_RTSP_PROFILE_AVPF:
5942 g_string_append (result, "/AVPF");
5944 case GST_RTSP_PROFILE_SAVPF:
5945 g_string_append (result, "/SAVPF");
5951 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
5952 GST_DEBUG_OBJECT (src, "adding UDP unicast");
5954 g_string_append (result, "/UDP");
5955 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
5956 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
5957 GST_DEBUG_OBJECT (src, "adding UDP multicast");
5958 /* we don't have to allocate any UDP ports yet, if the selected transport
5959 * turns out to be multicast we can create them and join the multicast
5960 * group indicated in the transport reply */
5962 g_string_append (result, "/UDP");
5963 g_string_append (result, ";multicast");
5964 if (src->next_port_num != 0) {
5965 if (src->client_port_range.max > 0 &&
5966 src->next_port_num >= src->client_port_range.max)
5969 g_string_append_printf (result, ";client_port=%d-%d",
5970 src->next_port_num, src->next_port_num + 1);
5972 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
5973 GST_DEBUG_OBJECT (src, "adding TCP");
5975 g_string_append (result, "/TCP;unicast;interleaved=%%i1-%%i2");
5977 *transports = g_string_free (result, FALSE);
5979 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
5986 GST_ERROR ("extension gave error %d", res);
5991 GST_ERROR ("no more ports available");
5992 return GST_RTSP_ERROR;
5996 static GstRTSPResult
5997 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
5998 gint orig_rtpport, gint orig_rtcpport)
6001 gint nr_udp, nr_int;
6003 gint rtpport = 0, rtcpport = 0;
6006 src = stream->parent;
6008 /* find number of placeholders first */
6009 if (strstr (*transports, "%%i2"))
6011 else if (strstr (*transports, "%%i1"))
6016 if (strstr (*transports, "%%u2"))
6018 else if (strstr (*transports, "%%u1"))
6023 if (nr_udp == 0 && nr_int == 0)
6027 if (!orig_rtpport || !orig_rtcpport) {
6028 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
6031 rtpport = orig_rtpport;
6032 rtcpport = orig_rtcpport;
6036 str = g_string_new ("");
6038 while ((next = strstr (p, "%%"))) {
6039 g_string_append_len (str, p, next - p);
6040 if (next[2] == 'u') {
6042 g_string_append_printf (str, "%d", rtpport);
6043 else if (next[3] == '2')
6044 g_string_append_printf (str, "%d", rtcpport);
6046 if (next[2] == 'i') {
6048 g_string_append_printf (str, "%d", src->free_channel);
6049 else if (next[3] == '2')
6050 g_string_append_printf (str, "%d", src->free_channel + 1);
6055 /* append final part */
6056 g_string_append (str, p);
6058 g_free (*transports);
6059 *transports = g_string_free (str, FALSE);
6067 GST_ERROR ("failed to allocate udp ports");
6068 return GST_RTSP_ERROR;
6073 enc_key_length_from_cipher_name (const gchar * cipher)
6075 if (g_strcmp0 (cipher, "aes-128-icm") == 0)
6076 return AES_128_KEY_LEN;
6077 else if (g_strcmp0 (cipher, "aes-256-icm") == 0)
6078 return AES_256_KEY_LEN;
6080 GST_ERROR ("encryption algorithm '%s' not supported", cipher);
6086 auth_key_length_from_auth_name (const gchar * auth)
6088 if (g_strcmp0 (auth, "hmac-sha1-32") == 0)
6089 return HMAC_32_KEY_LEN;
6090 else if (g_strcmp0 (auth, "hmac-sha1-80") == 0)
6091 return HMAC_80_KEY_LEN;
6093 GST_ERROR ("authentication algorithm '%s' not supported", auth);
6099 signal_get_srtcp_params (GstRTSPSrc * src, GstRTSPStream * stream)
6101 GstCaps *caps = NULL;
6103 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_REQUEST_RTCP_KEY], 0,
6107 GST_DEBUG_OBJECT (src, "SRTP parameters received");
6113 default_srtcp_params (void)
6121 /* create a random key */
6122 key_data = g_malloc (KEY_SIZE);
6123 for (i = 0; i < KEY_SIZE; i += 4)
6124 GST_WRITE_UINT32_BE (key_data + i, g_random_int ());
6126 buf = gst_buffer_new_wrapped (key_data, KEY_SIZE);
6128 caps = gst_caps_new_simple ("application/x-srtp",
6129 "srtp-key", GST_TYPE_BUFFER, buf,
6130 "srtcp-cipher", G_TYPE_STRING, "aes-128-icm",
6131 "srtcp-auth", G_TYPE_STRING, "hmac-sha1-80", NULL);
6133 gst_buffer_unref (buf);
6139 gst_rtspsrc_stream_make_keymgmt (GstRTSPSrc * src, GstRTSPStream * stream)
6142 gchar *result, *base64;
6145 GstMIKEYMessage *msg;
6146 GstMIKEYPayload *payload, *pkd;
6152 const gchar *srtcpcipher, *srtcpauth;
6154 stream->srtcpparams = signal_get_srtcp_params (src, stream);
6155 if (stream->srtcpparams == NULL)
6156 stream->srtcpparams = default_srtcp_params ();
6158 s = gst_caps_get_structure (stream->srtcpparams, 0);
6160 srtcpcipher = gst_structure_get_string (s, "srtcp-cipher");
6161 srtcpauth = gst_structure_get_string (s, "srtcp-auth");
6162 val = gst_structure_get_value (s, "srtp-key");
6164 if (srtcpcipher == NULL || srtcpauth == NULL || val == NULL) {
6165 GST_ERROR_OBJECT (src, "could not find the right SRTP parameters in caps");
6169 srtpkey = gst_value_get_buffer (val);
6171 msg = gst_mikey_message_new ();
6172 /* unencrypted MIKEY message, we send this over TLS so this is allowed */
6173 gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT,
6174 FALSE, GST_MIKEY_PRF_MIKEY_1, g_random_int (), GST_MIKEY_MAP_TYPE_SRTP);
6175 /* add policy '0' for our SSRC */
6176 gst_mikey_message_add_cs_srtp (msg, 0, stream->send_ssrc, 0);
6177 /* timestamp is now */
6178 gst_mikey_message_add_t_now_ntp_utc (msg);
6179 /* add some random data */
6180 gst_mikey_message_add_rand_len (msg, 16);
6182 /* the policy '0' is SRTP */
6183 payload = gst_mikey_payload_new (GST_MIKEY_PT_SP);
6184 gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP);
6186 /* only AES-CM is supported */
6188 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1, &byte);
6189 /* encryption key length */
6190 byte = enc_key_length_from_cipher_name (srtcpcipher);
6191 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_KEY_LEN, 1,
6193 /* only HMAC-SHA1 */
6194 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1,
6196 /* authentication key length */
6197 byte = auth_key_length_from_auth_name (srtcpauth);
6198 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_KEY_LEN, 1,
6200 /* we enable encryption on RTP and RTCP */
6201 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1,
6203 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1,
6205 /* we enable authentication on RTP and RTCP */
6206 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1,
6208 gst_mikey_message_add_payload (msg, payload);
6210 /* make unencrypted KEMAC */
6211 payload = gst_mikey_payload_new (GST_MIKEY_PT_KEMAC);
6212 gst_mikey_payload_kemac_set (payload, GST_MIKEY_ENC_NULL, GST_MIKEY_MAC_NULL);
6213 /* add the key in KEMAC */
6214 pkd = gst_mikey_payload_new (GST_MIKEY_PT_KEY_DATA);
6215 gst_buffer_map (srtpkey, &info, GST_MAP_READ);
6216 gst_mikey_payload_key_data_set_key (pkd, GST_MIKEY_KD_TEK, info.size,
6218 gst_buffer_unmap (srtpkey, &info);
6219 gst_mikey_payload_kemac_add_sub (payload, pkd);
6220 gst_mikey_message_add_payload (msg, payload);
6222 /* now serialize this to bytes */
6223 bytes = gst_mikey_message_to_bytes (msg, NULL, NULL);
6224 gst_mikey_message_unref (msg);
6225 /* and make it into base64 */
6226 data = g_bytes_get_data (bytes, &size);
6227 base64 = g_base64_encode (data, size);
6228 g_bytes_unref (bytes);
6230 result = g_strdup_printf ("prot=mikey;uri=\"%s\";data=\"%s\"",
6231 stream->conninfo.location, base64);
6238 /* Perform the SETUP request for all the streams.
6240 * We ask the server for a specific transport, which initially includes all the
6241 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
6242 * two local UDP ports that we send to the server.
6244 * Once the server replied with a transport, we configure the other streams
6245 * with the same transport.
6247 * This function will also configure the stream for the selected transport,
6248 * which basically means creating the pipeline.
6250 static GstRTSPResult
6251 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
6254 GstRTSPResult res = GST_RTSP_ERROR;
6255 GstRTSPMessage request = { 0 };
6256 GstRTSPMessage response = { 0 };
6257 GstRTSPStream *stream = NULL;
6258 GstRTSPLowerTrans protocols;
6259 GstRTSPStatusCode code;
6260 gboolean unsupported_real = FALSE;
6261 gint rtpport, rtcpport;
6265 if (src->conninfo.connection) {
6266 url = gst_rtsp_connection_get_url (src->conninfo.connection);
6267 /* we initially allow all configured lower transports. based on the URL
6268 * transports and the replies from the server we narrow them down. */
6269 protocols = url->transports & src->cur_protocols;
6272 protocols = src->cur_protocols;
6278 /* reset some state */
6279 src->free_channel = 0;
6280 src->interleaved = FALSE;
6281 src->need_activate = FALSE;
6282 /* keep track of next port number, 0 is random */
6283 src->next_port_num = src->client_port_range.min;
6284 rtpport = rtcpport = 0;
6286 if (G_UNLIKELY (src->streams == NULL))
6289 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6290 GstRTSPConnection *conn;
6297 stream = (GstRTSPStream *) walk->data;
6299 caps = stream_get_caps_for_pt (stream, stream->default_pt);
6301 GST_DEBUG_OBJECT (src, "skipping stream %p, no caps", stream);
6305 if (stream->skipped) {
6306 GST_DEBUG_OBJECT (src, "skipping stream %p", stream);
6310 /* see if we need to configure this stream */
6311 if (!gst_rtsp_ext_list_configure_stream (src->extensions, caps)) {
6312 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
6317 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_SELECT_STREAM], 0,
6318 stream->id, caps, &selected);
6320 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by signal", stream);
6324 /* merge/overwrite global caps */
6329 s = gst_caps_get_structure (caps, 0);
6331 num = gst_structure_n_fields (src->props);
6332 for (j = 0; j < num; j++) {
6336 name = gst_structure_nth_field_name (src->props, j);
6337 val = gst_structure_get_value (src->props, name);
6338 gst_structure_set_value (s, name, val);
6340 GST_DEBUG_OBJECT (src, "copied %s", name);
6344 /* skip setup if we have no URL for it */
6345 if (stream->conninfo.location == NULL) {
6346 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
6350 if (src->conninfo.connection == NULL) {
6351 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
6352 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
6355 conn = stream->conninfo.connection;
6357 conn = src->conninfo.connection;
6359 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
6360 stream->conninfo.location);
6362 /* if we have a multicast connection, only suggest multicast from now on */
6363 if (stream->is_multicast)
6364 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
6367 /* first selectable protocol */
6368 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
6370 if (!protocol_masks[mask])
6374 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
6375 protocol_masks[mask]);
6376 /* create a string with first transport in line */
6378 res = gst_rtspsrc_create_transports_string (src,
6379 protocols & protocol_masks[mask], stream->profile, &transports);
6380 if (res < 0 || transports == NULL)
6381 goto setup_transport_failed;
6383 if (strlen (transports) == 0) {
6384 g_free (transports);
6385 GST_DEBUG_OBJECT (src, "no transports found");
6390 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
6392 /* replace placeholders with real values, this function will optionally
6393 * allocate UDP ports and other info needed to execute the setup request */
6394 res = gst_rtspsrc_prepare_transports (stream, &transports,
6395 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
6397 g_free (transports);
6398 goto setup_transport_failed;
6401 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
6403 /* create SETUP request */
6405 gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
6406 stream->conninfo.location);
6408 g_free (transports);
6409 goto create_request_failed;
6412 /* select transport */
6413 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
6416 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
6417 stream->profile == GST_RTSP_PROFILE_SAVPF) {
6418 hval = gst_rtspsrc_stream_make_keymgmt (src, stream);
6419 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_KEYMGMT, hval);
6422 /* if the user wants a non default RTP packet size we add the blocksize
6424 if (src->rtp_blocksize > 0) {
6425 hval = g_strdup_printf ("%d", src->rtp_blocksize);
6426 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
6430 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
6433 /* handle the code ourselves */
6434 res = gst_rtspsrc_send (src, conn, &request, &response, &code);
6439 case GST_RTSP_STS_OK:
6441 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
6442 gst_rtsp_message_unset (&request);
6443 gst_rtsp_message_unset (&response);
6444 /* cleanup of leftover transport */
6445 gst_rtspsrc_stream_free_udp (stream);
6446 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
6447 * we might be in this case */
6448 if (stream->container && rtpport && rtcpport && !retry) {
6449 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
6454 /* this transport did not go down well, but we may have others to try
6455 * that we did not send yet, try those and only give up then
6456 * but not without checking for lost cause/extension so we can
6457 * post a nicer/more useful error message later */
6458 if (!unsupported_real)
6459 unsupported_real = stream->is_real;
6460 /* select next available protocol, give up on this stream if none */
6462 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
6464 if (!protocol_masks[mask] || unsupported_real)
6469 /* cleanup of leftover transport and move to the next stream */
6470 gst_rtspsrc_stream_free_udp (stream);
6471 goto response_error;
6474 /* parse response transport */
6476 gchar *resptrans = NULL;
6477 GstRTSPTransport transport = { 0 };
6479 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
6482 gst_rtspsrc_stream_free_udp (stream);
6486 /* parse transport, go to next stream on parse error */
6487 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
6488 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
6492 /* update allowed transports for other streams. once the transport of
6493 * one stream has been determined, we make sure that all other streams
6494 * are configured in the same way */
6495 switch (transport.lower_transport) {
6496 case GST_RTSP_LOWER_TRANS_TCP:
6497 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
6498 protocols = GST_RTSP_LOWER_TRANS_TCP;
6499 src->interleaved = TRUE;
6500 /* update free channels */
6502 MAX (transport.interleaved.min, src->free_channel);
6504 MAX (transport.interleaved.max, src->free_channel);
6505 src->free_channel++;
6507 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
6508 /* only allow multicast for other streams */
6509 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
6510 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
6511 /* if the server selected our ports, increment our counters so that
6512 * we select a new port later */
6513 if (src->next_port_num == transport.port.min &&
6514 src->next_port_num + 1 == transport.port.max) {
6515 src->next_port_num += 2;
6518 case GST_RTSP_LOWER_TRANS_UDP:
6519 /* only allow unicast for other streams */
6520 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
6521 protocols = GST_RTSP_LOWER_TRANS_UDP;
6524 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
6525 transport.lower_transport);
6529 if (stream->container || (!src->interleaved && !retry)) {
6530 /* now configure the stream with the selected transport */
6531 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
6532 GST_DEBUG_OBJECT (src,
6533 "could not configure stream %p transport, skipping stream",
6536 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
6537 /* retain the first allocated UDP port pair */
6538 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
6539 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
6542 /* we need to activate at least one streams when we detect activity */
6543 src->need_activate = TRUE;
6545 /* stream is setup now */
6546 stream->setup = TRUE;
6551 GstRTSPStream *sskip;
6553 skip = g_list_next (skip);
6557 sskip = (GstRTSPStream *) skip->data;
6559 /* skip all streams with the same control url */
6560 if (g_str_equal (stream->conninfo.location, sskip->conninfo.location)) {
6561 GST_DEBUG_OBJECT (src, "found stream %p with same control %s",
6562 sskip, sskip->conninfo.location);
6563 sskip->skipped = TRUE;
6568 /* clean up our transport struct */
6569 gst_rtsp_transport_init (&transport);
6570 /* clean up used RTSP messages */
6571 gst_rtsp_message_unset (&request);
6572 gst_rtsp_message_unset (&response);
6576 /* store the transport protocol that was configured */
6577 src->cur_protocols = protocols;
6579 gst_rtsp_ext_list_stream_select (src->extensions, url);
6581 /* if there is nothing to activate, error out */
6582 if (!src->need_activate)
6583 goto nothing_to_activate;
6590 /* no transport possible, post an error and stop */
6591 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
6592 ("Could not connect to server, no protocols left"));
6593 return GST_RTSP_ERROR;
6597 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6598 ("SDP contains no streams"));
6599 return GST_RTSP_ERROR;
6601 create_request_failed:
6603 gchar *str = gst_rtsp_strresult (res);
6605 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6606 ("Could not create request. (%s)", str));
6610 setup_transport_failed:
6612 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6613 ("Could not setup transport."));
6614 res = GST_RTSP_ERROR;
6619 const gchar *str = gst_rtsp_status_as_text (code);
6621 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6622 ("Error (%d): %s", code, GST_STR_NULL (str)));
6623 res = GST_RTSP_ERROR;
6628 gchar *str = gst_rtsp_strresult (res);
6630 if (res != GST_RTSP_EINTR) {
6631 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6632 ("Could not send message. (%s)", str));
6634 GST_WARNING_OBJECT (src, "send interrupted");
6641 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6642 ("Server did not select transport."));
6643 res = GST_RTSP_ERROR;
6646 nothing_to_activate:
6648 /* none of the available error codes is really right .. */
6649 if (unsupported_real) {
6650 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6651 (_("No supported stream was found. You might need to install a "
6652 "GStreamer RTSP extension plugin for Real media streams.")),
6655 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6656 (_("No supported stream was found. You might need to allow "
6657 "more transport protocols or may otherwise be missing "
6658 "the right GStreamer RTSP extension plugin.")), (NULL));
6660 return GST_RTSP_ERROR;
6664 gst_rtsp_message_unset (&request);
6665 gst_rtsp_message_unset (&response);
6671 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
6672 GstSegment * segment)
6675 GstRTSPTimeRange *therange;
6678 gst_rtsp_range_free (src->range);
6680 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
6681 GST_DEBUG_OBJECT (src, "parsed range %s", range);
6682 src->range = therange;
6684 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
6686 gst_segment_init (segment, GST_FORMAT_TIME);
6690 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
6691 therange->min.type, therange->min.seconds, therange->max.type,
6692 therange->max.seconds);
6694 if (therange->min.type == GST_RTSP_TIME_NOW)
6696 else if (therange->min.type == GST_RTSP_TIME_END)
6699 seconds = therange->min.seconds * GST_SECOND;
6701 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
6702 GST_TIME_ARGS (seconds));
6704 /* we need to start playback without clipping from the position reported by
6706 segment->start = seconds;
6707 segment->position = seconds;
6709 if (therange->max.type == GST_RTSP_TIME_NOW)
6711 else if (therange->max.type == GST_RTSP_TIME_END)
6714 seconds = therange->max.seconds * GST_SECOND;
6716 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
6717 GST_TIME_ARGS (seconds));
6719 /* live (WMS) server might send overflowed large max as its idea of infinity,
6720 * compensate to prevent problems later on */
6721 if (seconds != -1 && seconds < 0) {
6723 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
6726 /* live (WMS) might send min == max, which is not worth recording */
6727 if (segment->duration == -1 && seconds == segment->start)
6730 /* don't change duration with unknown value, we might have a valid value
6731 * there that we want to keep. */
6733 segment->duration = seconds;
6738 /* Parse clock profived by the server with following syntax:
6740 * "GstNetTimeProvider <wrapped-clock> <server-IP:port> <clock-time>"
6743 gst_rtspsrc_parse_gst_clock (GstRTSPSrc * src, const gchar * gstclock)
6745 gboolean res = FALSE;
6747 if (g_str_has_prefix (gstclock, "GstNetTimeProvider ")) {
6748 gchar **fields = NULL, **parts = NULL;
6749 gchar *remote_ip, *str;
6751 GstClockTime base_time;
6754 fields = g_strsplit (gstclock, " ", 0);
6756 /* wrapped clock, not very interesting for now */
6757 if (fields[1] == NULL)
6760 /* remote IP address and port */
6761 if ((str = fields[2]) == NULL)
6764 parts = g_strsplit (str, ":", 0);
6766 if ((remote_ip = parts[0]) == NULL)
6769 if ((str = parts[1]) == NULL)
6777 if ((str = fields[3]) == NULL)
6780 base_time = g_ascii_strtoull (str, NULL, 10);
6783 gst_net_client_clock_new ((gchar *) "GstRTSPClock", remote_ip, port,
6786 if (src->provided_clock)
6787 gst_object_unref (src->provided_clock);
6788 src->provided_clock = netclock;
6790 gst_element_post_message (GST_ELEMENT_CAST (src),
6791 gst_message_new_clock_provide (GST_OBJECT_CAST (src),
6792 src->provided_clock, TRUE));
6796 g_strfreev (fields);
6802 /* must be called with the RTSP state lock */
6803 static GstRTSPResult
6804 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
6810 /* prepare global stream caps properties */
6812 gst_structure_remove_all_fields (src->props);
6814 src->props = gst_structure_new_empty ("RTSPProperties");
6817 gst_sdp_message_dump (sdp);
6819 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
6821 /* let the app inspect and change the SDP */
6822 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_ON_SDP], 0, sdp);
6824 gst_segment_init (&src->segment, GST_FORMAT_TIME);
6826 /* parse range for duration reporting. */
6831 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
6835 /* keep track of the range and configure it in the segment */
6836 if (gst_rtspsrc_parse_range (src, range, &src->segment))
6840 /* parse clock information. This is GStreamer specific, a server can tell the
6841 * client what clock it is using and wrap that in a network clock. The
6842 * advantage of that is that we can slave to it. */
6844 const gchar *gstclock;
6847 gstclock = gst_sdp_message_get_attribute_val_n (sdp, "x-gst-clock", i);
6848 if (gstclock == NULL)
6851 /* parse the clock and expose it in the provide_clock method */
6852 if (gst_rtspsrc_parse_gst_clock (src, gstclock))
6856 /* try to find a global control attribute. Note that a '*' means that we should
6857 * do aggregate control with the current url (so we don't do anything and
6858 * leave the current connection as is) */
6860 const gchar *control;
6863 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
6864 if (control == NULL)
6867 /* only take fully qualified urls */
6868 if (g_str_has_prefix (control, "rtsp://"))
6872 g_free (src->conninfo.location);
6873 src->conninfo.location = g_strdup (control);
6874 /* make a connection for this, if there was a connection already, nothing
6876 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
6877 GST_ERROR_OBJECT (src, "could not connect");
6880 /* we need to keep the control url separate from the connection url because
6881 * the rules for constructing the media control url need it */
6882 g_free (src->control);
6883 src->control = g_strdup (control);
6886 /* create streams */
6887 n_streams = gst_sdp_message_medias_len (sdp);
6888 for (i = 0; i < n_streams; i++) {
6889 gst_rtspsrc_create_stream (src, sdp, i);
6892 src->state = GST_RTSP_STATE_INIT;
6895 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
6898 /* reset our state */
6899 src->need_range = TRUE;
6902 src->state = GST_RTSP_STATE_READY;
6909 GST_ERROR_OBJECT (src, "setup failed");
6910 gst_rtspsrc_cleanup (src);
6915 static GstRTSPResult
6916 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
6920 GstRTSPMessage request = { 0 };
6921 GstRTSPMessage response = { 0 };
6924 gchar *respcont = NULL;
6927 src->need_redirect = FALSE;
6929 /* can't continue without a valid url */
6930 if (G_UNLIKELY (src->conninfo.url == NULL)) {
6931 res = GST_RTSP_EINVAL;
6934 src->tried_url_auth = FALSE;
6936 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
6937 goto connect_failed;
6939 /* create OPTIONS */
6940 GST_DEBUG_OBJECT (src, "create options...");
6942 gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
6943 src->conninfo.url_str);
6945 goto create_request_failed;
6948 GST_DEBUG_OBJECT (src, "send options...");
6951 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
6954 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6959 if (!gst_rtspsrc_parse_methods (src, &response))
6962 /* create DESCRIBE */
6963 GST_DEBUG_OBJECT (src, "create describe...");
6965 gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
6966 src->conninfo.url_str);
6968 goto create_request_failed;
6970 /* we only accept SDP for now */
6971 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
6975 GST_DEBUG_OBJECT (src, "send describe...");
6978 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
6981 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
6985 /* we only perform redirect for the describe, currently */
6986 if (src->need_redirect) {
6987 /* close connection, we don't have to send a TEARDOWN yet, ignore the
6989 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
6991 gst_rtsp_message_unset (&request);
6992 gst_rtsp_message_unset (&response);
6998 /* it could be that the DESCRIBE method was not implemented */
6999 if (!src->methods & GST_RTSP_DESCRIBE)
7002 /* check if reply is SDP */
7003 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
7005 /* could not be set but since the request returned OK, we assume it
7006 * was SDP, else check it. */
7008 if (g_ascii_strcasecmp (respcont, "application/sdp") != 0)
7009 goto wrong_content_type;
7012 /* get message body and parse as SDP */
7013 gst_rtsp_message_get_body (&response, &data, &size);
7014 if (data == NULL || size == 0)
7017 GST_DEBUG_OBJECT (src, "parse SDP...");
7018 gst_sdp_message_new (sdp);
7019 gst_sdp_message_parse_buffer (data, size, *sdp);
7021 /* clean up any messages */
7022 gst_rtsp_message_unset (&request);
7023 gst_rtsp_message_unset (&response);
7030 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
7031 ("No valid RTSP URL was provided"));
7036 gchar *str = gst_rtsp_strresult (res);
7038 if (res != GST_RTSP_EINTR) {
7039 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
7040 ("Failed to connect. (%s)", str));
7042 GST_WARNING_OBJECT (src, "connect interrupted");
7047 create_request_failed:
7049 gchar *str = gst_rtsp_strresult (res);
7051 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7052 ("Could not create request. (%s)", str));
7058 /* Don't post a message - the rtsp_send method will have
7059 * taken care of it because we passed NULL for the response code */
7064 /* error was posted */
7065 res = GST_RTSP_ERROR;
7070 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
7071 ("Server does not support SDP, got %s.", respcont));
7072 res = GST_RTSP_ERROR;
7077 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
7078 ("Server can not provide an SDP."));
7079 res = GST_RTSP_ERROR;
7084 if (src->conninfo.connection) {
7085 GST_DEBUG_OBJECT (src, "free connection");
7086 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
7088 gst_rtsp_message_unset (&request);
7089 gst_rtsp_message_unset (&response);
7094 static GstRTSPResult
7095 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
7100 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
7102 if (src->sdp == NULL) {
7103 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
7107 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
7112 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
7119 GST_WARNING_OBJECT (src, "can't get sdp");
7120 src->open_error = TRUE;
7125 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
7126 src->open_error = TRUE;
7131 static GstRTSPResult
7132 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
7134 GstRTSPMessage request = { 0 };
7135 GstRTSPMessage response = { 0 };
7136 GstRTSPResult res = GST_RTSP_OK;
7138 const gchar *control;
7140 GST_DEBUG_OBJECT (src, "TEARDOWN...");
7142 gst_rtspsrc_set_state (src, GST_STATE_READY);
7144 if (src->state < GST_RTSP_STATE_READY) {
7145 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
7152 /* construct a control url */
7153 control = get_aggregate_control (src);
7155 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
7158 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7159 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7160 const gchar *setup_url;
7161 GstRTSPConnInfo *info;
7163 /* try aggregate control first but do non-aggregate control otherwise */
7165 setup_url = control;
7166 else if ((setup_url = stream->conninfo.location) == NULL)
7169 if (src->conninfo.connection) {
7170 info = &src->conninfo;
7171 } else if (stream->conninfo.connection) {
7172 info = &stream->conninfo;
7176 if (!info->connected)
7181 gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
7183 goto create_request_failed;
7186 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
7189 gst_rtspsrc_send (src, info->connection, &request, &response,
7193 /* FIXME, parse result? */
7194 gst_rtsp_message_unset (&request);
7195 gst_rtsp_message_unset (&response);
7198 /* early exit when we did aggregate control */
7204 /* close connections */
7205 GST_DEBUG_OBJECT (src, "closing connection...");
7206 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
7207 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7208 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7209 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
7213 gst_rtspsrc_cleanup (src);
7215 src->state = GST_RTSP_STATE_INVALID;
7218 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
7223 create_request_failed:
7225 gchar *str = gst_rtsp_strresult (res);
7227 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7228 ("Could not create request. (%s)", str));
7234 gchar *str = gst_rtsp_strresult (res);
7236 gst_rtsp_message_unset (&request);
7237 if (res != GST_RTSP_EINTR) {
7238 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7239 ("Could not send message. (%s)", str));
7241 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
7248 GST_DEBUG_OBJECT (src,
7249 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
7254 /* RTP-Info is of the format:
7256 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
7258 * rtptime corresponds to the timestamp for the NPT time given in the header
7259 * seqbase corresponds to the next sequence number we received. This number
7260 * indicates the first seqnum after the seek and should be used to discard
7261 * packets that are from before the seek.
7264 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
7269 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
7271 infos = g_strsplit (rtpinfo, ",", 0);
7272 for (i = 0; infos[i]; i++) {
7274 GstRTSPStream *stream;
7278 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
7280 /* init values, types of seqbase and timebase are bigger than needed so we
7281 * can store -1 as uninitialized values */
7286 /* parse url, find stream for url.
7287 * parse seq and rtptime. The seq number should be configured in the rtp
7288 * depayloader or session manager to detect gaps. Same for the rtptime, it
7289 * should be used to create an initial time newsegment. */
7290 fields = g_strsplit (infos[i], ";", 0);
7291 for (j = 0; fields[j]; j++) {
7292 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
7293 /* remove leading whitespace */
7294 fields[j] = g_strchug (fields[j]);
7295 if (g_str_has_prefix (fields[j], "url=")) {
7296 /* get the url and the stream */
7298 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
7299 } else if (g_str_has_prefix (fields[j], "seq=")) {
7300 seqbase = atoi (fields[j] + 4);
7301 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
7302 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
7305 g_strfreev (fields);
7306 /* now we need to store the values for the caps of the stream */
7307 if (stream != NULL) {
7308 GST_DEBUG_OBJECT (src,
7309 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
7310 stream, seqbase, timebase);
7312 /* we have a stream, configure detected params */
7313 stream->seqbase = seqbase;
7314 stream->timebase = timebase;
7323 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
7328 interval = strtoul (rtcp, NULL, 10);
7329 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
7334 interval *= GST_MSECOND;
7336 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7337 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7339 /* already (optionally) retrieved this when configuring manager */
7340 if (stream->session) {
7341 GObject *rtpsession = stream->session;
7343 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
7345 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
7349 /* now it happens that (Xenon) server sending this may also provide bogus
7350 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
7351 * and just use RTP-Info to sync */
7353 GObjectClass *klass;
7355 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
7356 if (g_object_class_find_property (klass, "rtcp-sync")) {
7357 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
7358 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
7364 gst_rtspsrc_get_float (const gchar * dstr)
7366 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
7368 /* canonicalise floating point string so we can handle float strings
7369 * in the form "24.930" or "24,930" irrespective of the current locale */
7370 g_strlcpy (s, dstr, sizeof (s));
7371 g_strdelimit (s, ",", '.');
7372 return g_ascii_strtod (s, NULL);
7376 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
7378 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
7380 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
7381 g_strlcpy (val_str, "now", sizeof (val_str));
7383 if (segment->position == 0) {
7384 g_strlcpy (val_str, "0", sizeof (val_str));
7386 g_ascii_dtostr (val_str, sizeof (val_str),
7387 ((gdouble) segment->position) / GST_SECOND);
7390 return g_strdup_printf ("npt=%s-", val_str);
7394 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
7398 stream->timebase = -1;
7399 stream->seqbase = -1;
7401 len = stream->ptmap->len;
7402 for (i = 0; i < len; i++) {
7403 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
7406 if (item->caps == NULL)
7409 item->caps = gst_caps_make_writable (item->caps);
7410 s = gst_caps_get_structure (item->caps, 0);
7411 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
7415 static GstRTSPResult
7416 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
7418 GstRTSPResult res = GST_RTSP_OK;
7420 if (src->state < GST_RTSP_STATE_READY) {
7421 res = GST_RTSP_ERROR;
7422 if (src->open_error) {
7423 GST_DEBUG_OBJECT (src, "the stream was in error");
7427 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
7429 if ((res = gst_rtspsrc_open (src, async)) < 0) {
7430 GST_DEBUG_OBJECT (src, "failed to open stream");
7439 static GstRTSPResult
7440 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
7442 GstRTSPMessage request = { 0 };
7443 GstRTSPMessage response = { 0 };
7444 GstRTSPResult res = GST_RTSP_OK;
7448 const gchar *control;
7450 GST_DEBUG_OBJECT (src, "PLAY...");
7452 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
7455 if (!(src->methods & GST_RTSP_PLAY))
7458 if (src->state == GST_RTSP_STATE_PLAYING)
7461 if (!src->conninfo.connection || !src->conninfo.connected)
7464 /* send some dummy packets before we activate the receive in the
7466 gst_rtspsrc_send_dummy_packets (src);
7468 /* require new SR packets */
7470 g_signal_emit_by_name (src->manager, "reset-sync", NULL);
7472 /* construct a control url */
7473 control = get_aggregate_control (src);
7475 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7476 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7477 const gchar *setup_url;
7478 GstRTSPConnection *conn;
7480 /* try aggregate control first but do non-aggregate control otherwise */
7482 setup_url = control;
7483 else if ((setup_url = stream->conninfo.location) == NULL)
7486 if (src->conninfo.connection) {
7487 conn = src->conninfo.connection;
7488 } else if (stream->conninfo.connection) {
7489 conn = stream->conninfo.connection;
7495 res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
7497 goto create_request_failed;
7499 if (src->need_range) {
7500 hval = gen_range_header (src, segment);
7502 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
7504 /* store the newsegment event so it can be sent from the streaming thread. */
7505 if (src->start_segment)
7506 gst_event_unref (src->start_segment);
7507 src->start_segment = gst_event_new_segment (segment);
7510 if (segment->rate != 1.0) {
7511 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
7513 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
7515 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
7517 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
7521 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
7523 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7526 /* seek may have silently failed as it is not supported */
7527 if (!(src->methods & GST_RTSP_PLAY)) {
7528 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
7529 /* obviously it is supported as we made it here */
7530 src->methods |= GST_RTSP_PLAY;
7531 src->seekable = FALSE;
7532 /* but there is nothing to parse in the response,
7533 * so convey we have no idea and not to expect anything particular */
7534 clear_rtp_base (src, stream);
7538 /* need to do for all streams */
7539 for (run = src->streams; run; run = g_list_next (run))
7540 clear_rtp_base (src, (GstRTSPStream *) run->data);
7542 /* NOTE the above also disables npt based eos detection */
7543 /* and below forces position to 0,
7544 * which is visible feedback we lost the plot */
7545 segment->start = segment->position = src->last_pos;
7548 gst_rtsp_message_unset (&request);
7550 /* parse RTP npt field. This is the current position in the stream (Normal
7551 * Play Time) and should be put in the NEWSEGMENT position field. */
7552 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
7554 gst_rtspsrc_parse_range (src, hval, segment);
7556 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
7557 segment->rate = 1.0;
7559 /* parse Speed header. This is the intended playback rate of the stream
7560 * and should be put in the NEWSEGMENT rate field. */
7561 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
7562 0) == GST_RTSP_OK) {
7563 segment->rate = gst_rtspsrc_get_float (hval);
7564 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
7565 &hval, 0) == GST_RTSP_OK) {
7566 segment->rate = gst_rtspsrc_get_float (hval);
7569 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
7570 * for the RTP packets. If this is not present, we assume all starts from 0...
7571 * This is info for the RTP session manager that we pass to it in caps. */
7573 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
7574 &hval, hval_idx++) == GST_RTSP_OK)
7575 gst_rtspsrc_parse_rtpinfo (src, hval);
7577 /* some servers indicate RTCP parameters in PLAY response,
7578 * rather than properly in SDP */
7579 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
7580 &hval, 0) == GST_RTSP_OK)
7581 gst_rtspsrc_handle_rtcp_interval (src, hval);
7583 gst_rtsp_message_unset (&response);
7585 /* early exit when we did aggregate control */
7589 /* configure the caps of the streams after we parsed all headers. Only reset
7590 * the manager object when we set a new Range header (we did a seek) */
7591 gst_rtspsrc_configure_caps (src, segment, src->need_range);
7593 /* set to PLAYING after we have configured the caps, otherwise we
7594 * might end up calling request_key (with SRTP) while caps are still
7595 * being configured. */
7596 gst_rtspsrc_set_state (src, GST_STATE_PLAYING);
7598 /* set again when needed */
7599 src->need_range = FALSE;
7601 src->running = TRUE;
7602 src->base_time = -1;
7603 src->state = GST_RTSP_STATE_PLAYING;
7606 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
7607 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7608 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7609 stream->discont = TRUE;
7614 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
7621 GST_DEBUG_OBJECT (src, "failed to open stream");
7626 GST_DEBUG_OBJECT (src, "PLAY is not supported");
7631 GST_DEBUG_OBJECT (src, "we were already PLAYING");
7634 create_request_failed:
7636 gchar *str = gst_rtsp_strresult (res);
7638 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7639 ("Could not create request. (%s)", str));
7645 gchar *str = gst_rtsp_strresult (res);
7647 gst_rtsp_message_unset (&request);
7648 if (res != GST_RTSP_EINTR) {
7649 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7650 ("Could not send message. (%s)", str));
7652 GST_WARNING_OBJECT (src, "PLAY interrupted");
7659 static GstRTSPResult
7660 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
7662 GstRTSPResult res = GST_RTSP_OK;
7663 GstRTSPMessage request = { 0 };
7664 GstRTSPMessage response = { 0 };
7666 const gchar *control;
7668 GST_DEBUG_OBJECT (src, "PAUSE...");
7670 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
7673 if (!(src->methods & GST_RTSP_PAUSE))
7676 if (src->state == GST_RTSP_STATE_READY)
7679 if (!src->conninfo.connection || !src->conninfo.connected)
7682 /* construct a control url */
7683 control = get_aggregate_control (src);
7685 /* loop over the streams. We might exit the loop early when we could do an
7686 * aggregate control */
7687 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7688 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7689 GstRTSPConnection *conn;
7690 const gchar *setup_url;
7692 /* try aggregate control first but do non-aggregate control otherwise */
7694 setup_url = control;
7695 else if ((setup_url = stream->conninfo.location) == NULL)
7698 if (src->conninfo.connection) {
7699 conn = src->conninfo.connection;
7700 } else if (stream->conninfo.connection) {
7701 conn = stream->conninfo.connection;
7707 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
7708 ("Sending PAUSE request"));
7711 gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
7713 goto create_request_failed;
7715 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7718 gst_rtsp_message_unset (&request);
7719 gst_rtsp_message_unset (&response);
7721 /* exit early when we did agregate control */
7726 /* change element states now */
7727 gst_rtspsrc_set_state (src, GST_STATE_PAUSED);
7730 src->state = GST_RTSP_STATE_READY;
7734 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
7741 GST_DEBUG_OBJECT (src, "failed to open stream");
7746 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
7751 GST_DEBUG_OBJECT (src, "we were already PAUSED");
7754 create_request_failed:
7756 gchar *str = gst_rtsp_strresult (res);
7758 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7759 ("Could not create request. (%s)", str));
7765 gchar *str = gst_rtsp_strresult (res);
7767 gst_rtsp_message_unset (&request);
7768 if (res != GST_RTSP_EINTR) {
7769 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7770 ("Could not send message. (%s)", str));
7772 GST_WARNING_OBJECT (src, "PAUSE interrupted");
7780 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
7782 GstRTSPSrc *rtspsrc;
7784 rtspsrc = GST_RTSPSRC (bin);
7786 switch (GST_MESSAGE_TYPE (message)) {
7787 case GST_MESSAGE_EOS:
7788 gst_message_unref (message);
7790 case GST_MESSAGE_ELEMENT:
7792 const GstStructure *s = gst_message_get_structure (message);
7794 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
7795 gboolean ignore_timeout;
7797 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
7799 GST_OBJECT_LOCK (rtspsrc);
7800 ignore_timeout = rtspsrc->ignore_timeout;
7801 rtspsrc->ignore_timeout = TRUE;
7802 GST_OBJECT_UNLOCK (rtspsrc);
7804 /* we only act on the first udp timeout message, others are irrelevant
7805 * and can be ignored. */
7806 if (!ignore_timeout)
7807 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
7809 gst_message_unref (message);
7812 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7815 case GST_MESSAGE_ERROR:
7818 GstRTSPStream *stream;
7821 udpsrc = GST_MESSAGE_SRC (message);
7823 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
7824 GST_ELEMENT_NAME (udpsrc));
7826 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
7830 /* we ignore the RTCP udpsrc */
7831 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
7834 /* if we get error messages from the udp sources, that's not a problem as
7835 * long as not all of them error out. We also don't really know what the
7836 * problem is, the message does not give enough detail... */
7837 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
7838 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
7839 if (ret != GST_FLOW_OK)
7843 gst_message_unref (message);
7847 /* fatal but not our message, forward */
7848 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7853 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7859 /* the thread where everything happens */
7861 gst_rtspsrc_thread (GstRTSPSrc * src)
7865 GST_OBJECT_LOCK (src);
7866 cmd = src->pending_cmd;
7867 if (cmd == CMD_RECONNECT || cmd == CMD_PLAY || cmd == CMD_PAUSE
7868 || cmd == CMD_LOOP || cmd == CMD_OPEN)
7869 src->pending_cmd = CMD_LOOP;
7871 src->pending_cmd = CMD_WAIT;
7872 GST_DEBUG_OBJECT (src, "got command %d", cmd);
7874 /* we got the message command, so ensure communication is possible again */
7875 gst_rtspsrc_connection_flush (src, FALSE);
7877 src->busy_cmd = cmd;
7878 GST_OBJECT_UNLOCK (src);
7882 gst_rtspsrc_open (src, TRUE);
7885 gst_rtspsrc_play (src, &src->segment, TRUE);
7888 gst_rtspsrc_pause (src, TRUE);
7891 gst_rtspsrc_close (src, TRUE, FALSE);
7894 gst_rtspsrc_loop (src);
7897 gst_rtspsrc_reconnect (src, FALSE);
7903 GST_OBJECT_LOCK (src);
7904 /* and go back to sleep */
7905 if (src->pending_cmd == CMD_WAIT) {
7907 gst_task_pause (src->task);
7910 src->busy_cmd = CMD_WAIT;
7911 GST_OBJECT_UNLOCK (src);
7915 gst_rtspsrc_start (GstRTSPSrc * src)
7917 GST_DEBUG_OBJECT (src, "starting");
7919 GST_OBJECT_LOCK (src);
7921 src->pending_cmd = CMD_WAIT;
7923 if (src->task == NULL) {
7924 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src, NULL);
7925 if (src->task == NULL)
7928 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
7930 GST_OBJECT_UNLOCK (src);
7937 GST_OBJECT_UNLOCK (src);
7938 GST_ERROR_OBJECT (src, "failed to create task");
7944 gst_rtspsrc_stop (GstRTSPSrc * src)
7948 GST_DEBUG_OBJECT (src, "stopping");
7950 /* also cancels pending task */
7951 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_ALL);
7953 GST_OBJECT_LOCK (src);
7954 if ((task = src->task)) {
7956 GST_OBJECT_UNLOCK (src);
7958 gst_task_stop (task);
7960 /* make sure it is not running */
7961 GST_RTSP_STREAM_LOCK (src);
7962 GST_RTSP_STREAM_UNLOCK (src);
7964 /* now wait for the task to finish */
7965 gst_task_join (task);
7967 /* and free the task */
7968 gst_object_unref (GST_OBJECT (task));
7970 GST_OBJECT_LOCK (src);
7972 GST_OBJECT_UNLOCK (src);
7974 /* ensure synchronously all is closed and clean */
7975 gst_rtspsrc_close (src, FALSE, TRUE);
7980 static GstStateChangeReturn
7981 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
7983 GstRTSPSrc *rtspsrc;
7984 GstStateChangeReturn ret;
7986 rtspsrc = GST_RTSPSRC (element);
7988 switch (transition) {
7989 case GST_STATE_CHANGE_NULL_TO_READY:
7990 if (!gst_rtspsrc_start (rtspsrc))
7993 case GST_STATE_CHANGE_READY_TO_PAUSED:
7994 /* init some state */
7995 rtspsrc->cur_protocols = rtspsrc->protocols;
7996 /* first attempt, don't ignore timeouts */
7997 rtspsrc->ignore_timeout = FALSE;
7998 rtspsrc->open_error = FALSE;
7999 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
8001 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
8002 set_manager_buffer_mode (rtspsrc);
8004 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
8005 /* unblock the tcp tasks and make the loop waiting */
8006 if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
8007 /* make sure it is waiting before we send PAUSE or PLAY below */
8008 GST_RTSP_STREAM_LOCK (rtspsrc);
8009 GST_RTSP_STREAM_UNLOCK (rtspsrc);
8012 case GST_STATE_CHANGE_PAUSED_TO_READY:
8018 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
8019 if (ret == GST_STATE_CHANGE_FAILURE)
8022 switch (transition) {
8023 case GST_STATE_CHANGE_NULL_TO_READY:
8024 ret = GST_STATE_CHANGE_SUCCESS;
8026 case GST_STATE_CHANGE_READY_TO_PAUSED:
8027 ret = GST_STATE_CHANGE_NO_PREROLL;
8029 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
8030 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
8031 ret = GST_STATE_CHANGE_SUCCESS;
8033 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
8034 /* send pause request and keep the idle task around */
8035 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
8036 ret = GST_STATE_CHANGE_NO_PREROLL;
8038 case GST_STATE_CHANGE_PAUSED_TO_READY:
8039 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
8040 ret = GST_STATE_CHANGE_SUCCESS;
8042 case GST_STATE_CHANGE_READY_TO_NULL:
8043 gst_rtspsrc_stop (rtspsrc);
8044 ret = GST_STATE_CHANGE_SUCCESS;
8055 GST_DEBUG_OBJECT (rtspsrc, "start failed");
8056 return GST_STATE_CHANGE_FAILURE;
8061 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
8064 GstRTSPSrc *rtspsrc;
8066 rtspsrc = GST_RTSPSRC (element);
8068 if (GST_EVENT_IS_DOWNSTREAM (event)) {
8069 res = gst_rtspsrc_push_event (rtspsrc, event);
8071 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
8078 /*** GSTURIHANDLER INTERFACE *************************************************/
8081 gst_rtspsrc_uri_get_type (GType type)
8086 static const gchar *const *
8087 gst_rtspsrc_uri_get_protocols (GType type)
8089 static const gchar *protocols[] =
8090 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
8091 "rtsps", "rtspsu", "rtspst", "rtspsh", NULL
8098 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
8100 GstRTSPSrc *src = GST_RTSPSRC (handler);
8102 /* FIXME: make thread-safe */
8103 return g_strdup (src->conninfo.location);
8107 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
8113 GstRTSPUrl *newurl = NULL;
8114 GstSDPMessage *sdp = NULL;
8116 src = GST_RTSPSRC (handler);
8118 /* same URI, we're fine */
8119 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
8122 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
8123 sres = gst_sdp_message_new (&sdp);
8127 GST_DEBUG_OBJECT (src, "parsing SDP message");
8128 sres = gst_sdp_message_parse_uri (uri, sdp);
8133 GST_DEBUG_OBJECT (src, "parsing URI");
8134 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
8138 /* if worked, free previous and store new url object along with the original
8140 GST_DEBUG_OBJECT (src, "configuring URI");
8141 g_free (src->conninfo.location);
8142 src->conninfo.location = g_strdup (uri);
8143 gst_rtsp_url_free (src->conninfo.url);
8144 src->conninfo.url = newurl;
8145 g_free (src->conninfo.url_str);
8147 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
8149 src->conninfo.url_str = NULL;
8152 gst_sdp_message_free (src->sdp);
8154 src->from_sdp = sdp != NULL;
8156 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
8157 GST_DEBUG_OBJECT (src, "request uri is: %s",
8158 GST_STR_NULL (src->conninfo.url_str));
8165 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
8170 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", sres);
8171 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
8172 "Could not create SDP");
8177 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", sres,
8178 GST_STR_NULL (uri));
8179 gst_sdp_message_free (sdp);
8180 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
8186 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
8187 GST_STR_NULL (uri), res);
8188 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
8189 "Invalid RTSP URI");
8195 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
8197 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
8199 iface->get_type = gst_rtspsrc_uri_get_type;
8200 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
8201 iface->get_uri = gst_rtspsrc_uri_get_uri;
8202 iface->set_uri = gst_rtspsrc_uri_set_uri;